From 5d585e1e756838d91144c3173323b96f5aa12874 Mon Sep 17 00:00:00 2001 From: Rob Herring Date: Tue, 28 Aug 2018 10:44:28 -0500 Subject: ASoC: Convert to using %pOFn instead of device_node.name In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Signed-off-by: Rob Herring Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 +++---- 5 files changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..b76a5f4f1785 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -478,7 +478,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..d597eba61992 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..56a227e0bd71 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1124,16 +1124,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } -- cgit v1.2.3-58-ga151 From d78b1e43e2182640b33d1c39245965d9231f0130 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 28 Aug 2018 14:35:02 +0100 Subject: ASoC: dapm: Remove clock framework ifdefs The clock code now has stub functions defined in its header files so the ifdefs around clocking code should no longer be necessary. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..78ab6965af55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1320,14 +1320,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, soc_dapm_async_complete(w->dapm); -#ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); } else { clk_disable_unprepare(w->clk); return 0; } -#endif + return 0; } EXPORT_SYMBOL_GPL(dapm_clock_event); @@ -3498,7 +3497,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: -#ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); @@ -3508,9 +3506,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } -#else - return NULL; -#endif break; default: break; -- cgit v1.2.3-58-ga151 From a5cd7e9cf587f51a84b86c828b4e1c7b392f448e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 28 Aug 2018 14:35:03 +0100 Subject: ASoC: dapm: Don't fail creating new DAPM control on NULL pinctrl devm_pinctrl_get will only return NULL in the case that pinctrl is not built into the kernel and all the pinctrl functions used by the DAPM core are appropriately stubbed for that case. There is no need to error out of snd_soc_dapm_new_control_unlocked if pinctrl isn't built into the kernel, so change the IS_ERR_OR_NULL to just an IS_ERR. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 78ab6965af55..d7be3981f026 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3487,7 +3487,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_pinctrl: w->pinctrl = devm_pinctrl_get(dapm->dev); - if (IS_ERR_OR_NULL(w->pinctrl)) { + if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); if (ret == -EPROBE_DEFER) return ERR_PTR(ret); -- cgit v1.2.3-58-ga151 From e33ffbd9cd39da09831ce62c11025d830bf78d9e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 Aug 2018 14:26:47 +0100 Subject: ASoC: dpcm: Properly initialise hw->rate_max If the CPU DAI does not initialise rate_max, say if using using KNOT or CONTINUOUS, then the rate_max field will be initialised to 0. A value of zero in the rate_max field of the hardware runtime will cause the sound card to support no sample rates at all. Obviously this is not desired, just a different mechanism is being used to apply the constraints. As such update the setting of rate_max in dpcm_init_runtime_hw to be consistent with the non-DPCM cases and set rate_max to UINT_MAX if nothing is defined on the CPU DAI. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e8b98bfd4cf1..eb6f4f1b65a9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1680,7 +1680,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; - runtime->hw.rate_max = stream->rate_max; + runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX); runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) -- cgit v1.2.3-58-ga151 From ac16df938e5107a1e08cf0f52818fa6a5f7bba94 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 28 Aug 2018 14:17:21 +0200 Subject: ASoC: meson: imply clock and reset controllers Add audio clock controller and ARB reset controller module implication for the device using them Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 8af8bc358a90..2ccbadc387de 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms" config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO + imply RESET_MESON_AUDIO_ARB config SND_MESON_AXG_FRDDR tristate "Amlogic AXG Playback FIFO support" @@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR config SND_MESON_AXG_TDM_FORMATTER tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO config SND_MESON_AXG_TDM_INTERFACE tristate @@ -58,6 +61,7 @@ config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" select SND_PCM_IEC958 imply SND_SOC_SPDIF + imply COMMON_CLK_AXG_AUDIO help Select Y or M to add support for SPDIF output serializer embedded in the Amlogic AXG SoC family -- cgit v1.2.3-58-ga151 From dadfab7272b13ca441efdb9aa9117bc669680b05 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 27 Aug 2018 16:15:29 +0200 Subject: ASoC: meson: axg-fifo: report interrupt request failure Return value of request_irq() was irgnored. Fix this and report the failure if any Fixes: 6dc4fa179fb8 ("ASoC: meson: add axg fifo base driver") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 30262550e37b..0e4f65e654c4 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, dev_name(dev), ss); + if (ret) + return ret; /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); -- cgit v1.2.3-58-ga151 From 302df2694b97e6a56f7956c1105630809b141ea0 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 27 Aug 2018 16:21:07 +0200 Subject: ASoC: meson: axg-tdm: restrict formats depending on slot width Restrict the formats possible on the TDM interface depending on the width of the TDM slot and let dpcm merging do the rest. Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 50 +++++++++++++++++++++---------------- 1 file changed, 29 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 7b8baf46d968..585ce030b79b 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, struct axg_tdm_stream *rx = (struct axg_tdm_stream *) dai->capture_dma_data; unsigned int tx_slots, rx_slots; + unsigned int fmt = 0; tx_slots = axg_tdm_slots_total(tx_mask); rx_slots = axg_tdm_slots_total(rx_mask); @@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, return -EINVAL; } - /* - * Amend the dai driver channel number and let dpcm channel merge do - * its job - */ - if (tx) { - tx->mask = tx_mask; - dai->driver->playback.channels_max = tx_slots; - } - - if (rx) { - rx->mask = rx_mask; - dai->driver->capture.channels_max = rx_slots; - } - iface->slots = slots; switch (slot_width) { case 0: - /* defaults width to 32 if not provided */ - iface->slot_width = 32; - break; - case 8: - case 16: - case 24: + slot_width = 32; + /* Fall-through */ case 32: - iface->slot_width = slot_width; + fmt |= SNDRV_PCM_FMTBIT_S32_LE; + /* Fall-through */ + case 24: + fmt |= SNDRV_PCM_FMTBIT_S24_LE; + fmt |= SNDRV_PCM_FMTBIT_S20_LE; + /* Fall-through */ + case 16: + fmt |= SNDRV_PCM_FMTBIT_S16_LE; + /* Fall-through */ + case 8: + fmt |= SNDRV_PCM_FMTBIT_S8; break; default: dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); return -EINVAL; } + iface->slot_width = slot_width; + + /* Amend the dai driver and let dpcm merge do its job */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + dai->driver->playback.formats = fmt; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + dai->driver->capture.formats = fmt; + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); -- cgit v1.2.3-58-ga151 From 513792c2554bdece11d82568ea25501a555abd34 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Fri, 24 Aug 2018 10:51:51 +0800 Subject: ASoC: rt5682: Update calibration function New calibration sequence allows rt5682 do calibration without MCLK. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 640d400ca013..0ea2759e5885 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2454,27 +2454,15 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); - regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); - regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); - regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); - regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); - regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); - regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); - regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); - regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); - regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); - regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2490,7 +2478,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); mutex_unlock(&rt5682->calibrate_mutex); -- cgit v1.2.3-58-ga151 From 8dce1d026da4588382ed8c03e791c7c9b37b22e8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 22 Aug 2018 15:24:58 -0500 Subject: ASoC: Intel: common: add table for HDA-based platforms Expose a table containing machine driver information for HDAudio-based platforms handled by ASoC on Intel hardware. We only set constant values that are valid across multiple platforms. The firmware name used by the DSP will be set dynamically for each platform. The table is made of a single entry for now, if we need more complicated set-up where HDAudio is mixed with ACPI-enumerated devices (I2C, SoundWire) then we'd expect the differentiation to be handled through information provided by the BIOS (as done for KBL Chromebooks). Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 6 ++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 +++++++++++++++++++++++ 3 files changed, 48 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c (limited to 'sound/soc') diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index bb1d24b703fb..f48f59e5b7b0 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -25,4 +25,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[]; +/* + * generic table used for HDA codec-based platforms, possibly with + * additional ACPI-enumerated codecs + */ +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[]; + #endif diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 915a34cdc8ac..c1f50a079d34 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hsw-bdw-match.o \ soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ - soc-acpi-intel-cnl-match.o + soc-acpi-intel-cnl-match.o \ + soc-acpi-intel-hda-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c new file mode 100644 index 000000000000..533c1064f84b --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c @@ -0,0 +1,40 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Intel Corporation. + +/* + * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata hda_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { + { + /* .id is not used in this file */ + .drv_name = "skl_hda_dsp_generic", + + /* .fw_filename is dynamically set in skylake driver */ + + /* .sof_fw_filename is dynamically set in sof/intel driver */ + + .sof_tplg_filename = "intel/sof-hda-generic.tplg", + + /* + * .machine_quirk and .quirk_data are not used here but + * can be used if we need a more complicated machine driver + * combining HDA+other device (e.g. DMIC). + */ + .pdata = &hda_pdata, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); -- cgit v1.2.3-58-ga151 From 7c33b5f16915a7bc3d3b81a9a041bdc562f71dfb Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Wed, 22 Aug 2018 15:24:59 -0500 Subject: ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs Add machine driver for Intel platforms (SKL/KBL/BXT/APL) with HDA and iDisp codecs. This patch adds support for only iDisp (HDMI/DP) codec. In the following patches support for HDA codecs will be added. This should work for other Intel platforms as well e.g. GLK,CNL however this series is not tested on all the platforms. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 8 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 103 +++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 +++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 144 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 2 + 6 files changed, 297 insertions(+) create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index cccda87f4b34..0f0d57859555 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -279,6 +279,14 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 87ef8b4058e5..6e88373cbe35 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -20,6 +20,7 @@ snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o +snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -46,3 +47,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c new file mode 100644 index 000000000000..f9917e0f2ba8 --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -0,0 +1,103 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Common functions used in different Intel machine drivers + */ +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "skl_hda_dsp_common.h" + +#define NAME_SIZE 32 + +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hda_hdmi_pcm *pcm; + char dai_name[NAME_SIZE]; + + pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d", + ctx->dai_index); + pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name); + if (!pcm->codec_dai) + return -EINVAL; + + pcm->device = device; + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +/* skl_hda_digital audio interface glue - connects codec <--> CPU */ +struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { + /* Back End DAI links */ + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +int skl_hda_hdmi_jack_init(struct snd_soc_card *card) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + struct skl_hda_hdmi_pcm *pcm; + char jack_name[NAME_SIZE]; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &pcm->hdmi_jack, + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &pcm->hdmi_jack); + if (err < 0) + return err; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h new file mode 100644 index 000000000000..b6c79696bfba --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -0,0 +1,38 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2015-18 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with HDA Codecs. + */ + +#ifndef __SOUND_SOC_HDA_DSP_COMMON_H +#define __SOUND_SOC_HDA_DSP_COMMON_H +#include +#include +#include +#include + +#define HDA_DSP_MAX_BE_DAI_LINKS 3 + +struct skl_hda_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + struct snd_soc_jack hdmi_jack; + int device; +}; + +struct skl_hda_private { + struct list_head hdmi_pcm_list; + int pcm_count; + int dai_index; + const char *platform_name; +}; + +extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; +int skl_hda_hdmi_jack_init(struct snd_soc_card *card); +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device); + +#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */ diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c new file mode 100644 index 000000000000..920bc2ce22aa --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -0,0 +1,144 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs + */ + +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "skl_hda_dsp_common.h" + +static const struct snd_soc_dapm_route skl_hda_map[] = { + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, +}; + +static int skl_hda_card_late_probe(struct snd_soc_card *card) +{ + return skl_hda_hdmi_jack_init(card); +} + +static int +skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + int ret = 0; + + dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name); + link->platform_name = ctx->platform_name; + link->nonatomic = 1; + + if (strstr(link->name, "HDMI")) { + ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); + + if (ret < 0) + return ret; + + ctx->dai_index++; + } + + ctx->pcm_count++; + return ret; +} + +static struct snd_soc_card hda_soc_card = { + .name = "skl_hda_card", + .owner = THIS_MODULE, + .dai_link = skl_hda_be_dai_links, + .dapm_routes = skl_hda_map, + .add_dai_link = skl_hda_add_dai_link, + .fully_routed = true, + .late_probe = skl_hda_card_late_probe, +}; + +#define IDISP_DAI_COUNT 3 +/* there are two routes per iDisp output */ +#define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) +#define IDISP_CODEC_MASK 0x4 + +static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) +{ + struct snd_soc_card *card = &hda_soc_card; + u32 codec_count, codec_mask; + int i, num_links, num_route; + + codec_mask = pdata->codec_mask; + codec_count = hweight_long(codec_mask); + + if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) { + num_links = IDISP_DAI_COUNT; + num_route = IDISP_ROUTE_COUNT; + } else { + return -EINVAL; + } + + card->num_links = num_links; + card->num_dapm_routes = num_route; + + for (i = 0; i < num_links; i++) + skl_hda_be_dai_links[i].platform_name = pdata->platform; + + return 0; +} + +static int skl_hda_audio_probe(struct platform_device *pdev) +{ + struct skl_machine_pdata *pdata; + struct skl_hda_private *ctx; + int ret; + + dev_dbg(&pdev->dev, "%s: entry\n", __func__); + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + pdata = dev_get_drvdata(&pdev->dev); + if (!pdata) + return -EINVAL; + + ret = skl_hda_fill_card_info(pdata); + if (ret < 0) { + dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); + return ret; + } + + ctx->pcm_count = hda_soc_card.num_links; + ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ + ctx->platform_name = pdata->platform; + + hda_soc_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&hda_soc_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); +} + +static struct platform_driver skl_hda_audio = { + .probe = skl_hda_audio_probe, + .driver = { + .name = "skl_hda_dsp_generic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(skl_hda_audio) + +/* Module information */ +MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver"); +MODULE_AUTHOR("Rakesh Ughreja "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_hda_dsp_generic"); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 78aa8bdcb619..4105a9371b64 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -117,6 +117,8 @@ struct skl_dma_params { struct skl_machine_pdata { u32 dmic_num; bool use_tplg_pcm; /* use dais and dai links from topology */ + const char *platform; + u32 codec_mask; }; struct skl_dsp_ops { -- cgit v1.2.3-58-ga151 From 9cdae4352cba3f66d39a4ef78bb726940ae1e513 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Wed, 22 Aug 2018 15:25:00 -0500 Subject: ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails When no I2S based codec entries are found in the BIOS, check if there are any HDA codecs detected on the bus. Based on the number of codecs found take appropriate action in machine driver. If there are two HDA codecs i.e. iDisp + HDA found on the bus, register DAIs and DAI links for both. If only one codec i.e. iDisp is found then load only iDisp machine driver. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 38 +++++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..3b836125d1de 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -472,6 +472,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = { {.name = "ssp5_sclkfs"}, }; +static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl, + struct snd_soc_acpi_mach *machines) +{ + struct hdac_bus *bus = skl_to_bus(skl); + struct snd_soc_acpi_mach *mach; + + /* check if we have any codecs detected on bus */ + if (bus->codec_mask == 0) + return NULL; + + /* point to common table */ + mach = snd_soc_acpi_intel_hda_machines; + + /* all entries in the machine table use the same firmware */ + mach->fw_filename = machines->fw_filename; + + return mach; +} + static int skl_find_machine(struct skl *skl, void *driver_data) { struct hdac_bus *bus = skl_to_bus(skl); @@ -479,9 +498,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data) struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); - if (mach == NULL) { - dev_err(bus->dev, "No matching machine driver found\n"); - return -ENODEV; + if (!mach) { + dev_dbg(bus->dev, "No matching I2S machine driver found\n"); + mach = skl_find_hda_machine(skl, driver_data); + if (!mach) { + dev_err(bus->dev, "No matching machine driver found\n"); + return -ENODEV; + } } skl->mach = mach; @@ -498,8 +521,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data) static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; + struct hdac_bus *bus = skl_to_bus(skl); + struct skl_machine_pdata *pdata; struct platform_device *pdev; int ret; @@ -516,8 +540,12 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } - if (mach->pdata) + if (mach->pdata) { + pdata = (struct skl_machine_pdata *)mach->pdata; + pdata->platform = dev_name(bus->dev); + pdata->codec_mask = bus->codec_mask; dev_set_drvdata(&pdev->dev, mach->pdata); + } skl->i2s_dev = pdev; -- cgit v1.2.3-58-ga151 From 3d17871349d5cec0a37ce9407ba72fdbf8572cfd Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Wed, 22 Aug 2018 15:25:01 -0500 Subject: ASoC: Intel: Skylake: add HDA BE DAIs Add support for HDA BE DAIs in SKL platform driver. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 70 ++++++++++++++++++++++++++++++++------- 1 file changed, 58 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..00b7a91b18c9 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -32,6 +32,7 @@ #define HDA_MONO 1 #define HDA_STEREO 2 #define HDA_QUAD 4 +#define HDA_MAX 8 static const struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | @@ -569,7 +570,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, stream_tag = hdac_stream(link_dev)->stream_tag; /* set the stream tag in the codec dai dma params */ - snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + else + snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); p_params.s_fmt = snd_pcm_format_width(params_format(params)); p_params.ch = params_channels(params); @@ -995,21 +999,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { - .name = "HD-Codec Pin", + .name = "Analog CPU DAI", .ops = &skl_link_dai_ops, .playback = { - .stream_name = "HD-Codec Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { - .stream_name = "HD-Codec Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Alt Analog CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Alt Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Alt Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Digital CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Digital CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Digital CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, }, }; -- cgit v1.2.3-58-ga151 From 00deadb5d86a3c1e691aaa073a8852a198595099 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Wed, 22 Aug 2018 15:25:02 -0500 Subject: ASoC: Intel: Skylake: use hda_bus instead of hdac_bus Use hda_bus instead of hdac_bus in the SKL ASoC platform driver to enable reuse of legacy HDA codec drivers. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 11 +++++++++-- sound/soc/intel/skylake/skl.h | 10 +++++++--- 2 files changed, 16 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3b836125d1de..5f281d443a53 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -33,6 +33,7 @@ #include #include #include +#include #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -673,7 +674,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) mutex_unlock(&bus->cmd_mutex); if (res == -1) return -EIO; - dev_dbg(bus->dev, "codec #%d probed OK\n", addr); + dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) @@ -816,7 +817,7 @@ static int skl_create(struct pci_dev *pci, { struct skl *skl; struct hdac_bus *bus; - + struct hda_bus *hbus; int err; *rskl = NULL; @@ -831,6 +832,7 @@ static int skl_create(struct pci_dev *pci, return -ENOMEM; } + hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); bus->use_posbuf = 1; @@ -838,6 +840,11 @@ static int skl_create(struct pci_dev *pci, INIT_WORK(&skl->probe_work, skl_probe_work); bus->bdl_pos_adj = 0; + mutex_init(&hbus->prepare_mutex); + hbus->pci = pci; + hbus->mixer_assigned = -1; + hbus->modelname = "sklbus"; + *rskl = skl; return 0; diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 4105a9371b64..8d48cd7c56c8 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -23,6 +23,7 @@ #include #include +#include #include #include "skl-nhlt.h" #include "skl-ssp-clk.h" @@ -71,7 +72,7 @@ struct skl_fw_config { }; struct skl { - struct hdac_bus hbus; + struct hda_bus hbus; struct pci_dev *pci; unsigned int init_done:1; /* delayed init status */ @@ -105,8 +106,11 @@ struct skl { struct snd_soc_acpi_mach *mach; }; -#define skl_to_bus(s) (&(s)->hbus) -#define bus_to_skl(bus) container_of(bus, struct skl, hbus) +#define skl_to_bus(s) (&(s)->hbus.core) +#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core) + +#define skl_to_hbus(s) (&(s)->hbus) +#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus)) /* to pass dai dma data */ struct skl_dma_params { -- cgit v1.2.3-58-ga151 From 6bae5ea9498926440ffc883f3dbceb0adc65e492 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Wed, 22 Aug 2018 15:25:03 -0500 Subject: ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers This patch adds a kernel module which is used by the legacy HDA codec drivers as library. This implements hdac_ext_bus_ops to enable the reuse of legacy HDA codec drivers with ASoC platform drivers. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 484 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/skl_hda_dsp_common.c | 24 ++ sound/soc/intel/boards/skl_hda_dsp_common.h | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 38 +++ sound/soc/intel/skylake/skl.c | 47 ++- 10 files changed, 634 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h (limited to 'sound/soc') diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 2222b47d4ec4..9174f1b3a987 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev) hda_codec_patch_t patch; int err; + if (codec->bus->core.ext_ops) { + if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach)) + return -EINVAL; + return codec->bus->core.ext_ops->hdev_attach(&codec->core); + } + if (WARN_ON(!codec->preset)) return -EINVAL; @@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + if (codec->bus->core.ext_ops) { + if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach)) + return -EINVAL; + return codec->bus->core.ext_ops->hdev_detach(&codec->core); + } + if (codec->patch_ops.free) codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..bf0b949eb7e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI + select SND_SOC_HDAC_HDA select SND_SOC_ICS43432 select SND_SOC_INNO_RK3036 select SND_SOC_ISABELLE if I2C @@ -615,6 +616,10 @@ config SND_SOC_HDAC_HDMI select SND_PCM_ELD select HDMI +config SND_SOC_HDAC_HDA + tristate + select SND_HDA + config SND_SOC_ICS43432 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..3046b33ca9d3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o +snd-soc-hdac-hda-objs := hdac_hda.o snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o @@ -338,6 +339,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o +obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c new file mode 100644 index 000000000000..8c25a1332fa7 --- /dev/null +++ b/sound/soc/codecs/hdac_hda.c @@ -0,0 +1,484 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers + * with ASoC platform drivers. These APIs are called by the legacy HDA + * codec drivers using hdac_ext_bus_ops ops. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "hdac_hda.h" + +#define HDAC_ANALOG_DAI_ID 0 +#define HDAC_DIGITAL_DAI_ID 1 +#define HDAC_ALT_ANALOG_DAI_ID 2 + +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static int hdac_hda_dai_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static void hdac_hda_dai_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); +static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, + struct snd_soc_dai *dai); + +static struct snd_soc_dai_ops hdac_hda_dai_ops = { + .startup = hdac_hda_dai_open, + .shutdown = hdac_hda_dai_close, + .prepare = hdac_hda_dai_prepare, + .hw_free = hdac_hda_dai_hw_free, + .set_tdm_slot = hdac_hda_dai_set_tdm_slot, +}; + +static struct snd_soc_dai_driver hdac_hda_dais[] = { +{ + .id = HDAC_ANALOG_DAI_ID, + .name = "Analog Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Analog Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Analog Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +}, +{ + .id = HDAC_DIGITAL_DAI_ID, + .name = "Digital Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Digital Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Digital Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +}, +{ + .id = HDAC_ALT_ANALOG_DAI_ID, + .name = "Alt Analog Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Alt Analog Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Alt Analog Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +} + +}; + +static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hdac_hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = &hda_pvt->pcm[dai->id]; + if (tx_mask) + pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; + else + pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; + + return 0; +} + +static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + hda_stream = &pcm->stream[substream->stream]; + snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream); + + return 0; +} + +static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hdac_device *hdev; + struct hda_pcm_stream *hda_stream; + unsigned int format_val; + struct hda_pcm *pcm; + unsigned int stream; + int ret = 0; + + hda_pvt = snd_soc_component_get_drvdata(component); + hdev = &hda_pvt->codec.core; + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + hda_stream = &pcm->stream[substream->stream]; + + format_val = snd_hdac_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hda_stream->maxbps, + 0); + if (!format_val) { + dev_err(&hdev->dev, + "invalid format_val, rate=%d, ch=%d, format=%d\n", + runtime->rate, runtime->channels, runtime->format); + return -EINVAL; + } + + stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + + ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, + stream, format_val, substream); + if (ret < 0) + dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + + return ret; +} + +static int hdac_hda_dai_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + int ret; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + snd_hda_codec_pcm_get(pcm); + + hda_stream = &pcm->stream[substream->stream]; + + ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream); + if (ret < 0) + snd_hda_codec_pcm_put(pcm); + + return ret; +} + +static void hdac_hda_dai_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return; + + hda_stream = &pcm->stream[substream->stream]; + + hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream); + + snd_hda_codec_pcm_put(pcm); +} + +static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, + struct snd_soc_dai *dai) +{ + struct hda_codec *hcodec = &hda_pvt->codec; + struct hda_pcm *cpcm; + const char *pcm_name; + + switch (dai->id) { + case HDAC_ANALOG_DAI_ID: + pcm_name = "Analog"; + break; + case HDAC_DIGITAL_DAI_ID: + pcm_name = "Digital"; + break; + case HDAC_ALT_ANALOG_DAI_ID: + pcm_name = "Alt Analog"; + break; + default: + dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + return NULL; + } + + list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) { + if (strpbrk(cpcm->name, pcm_name)) + return cpcm; + } + + dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + return NULL; +} + +static int hdac_hda_codec_probe(struct snd_soc_component *component) +{ + struct hdac_hda_priv *hda_pvt = + snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct hdac_device *hdev = &hda_pvt->codec.core; + struct hda_codec *hcodec = &hda_pvt->codec; + struct hdac_ext_link *hlink; + hda_codec_patch_t patch; + int ret; + + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + + snd_hdac_ext_bus_link_get(hdev->bus, hlink); + + ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, + hdev->addr, hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + goto error_no_pm; + } + + /* + * snd_hda_codec_device_new decrements the usage count so call get pm + * else the device will be powered off + */ + pm_runtime_get_noresume(&hdev->dev); + + hcodec->bus->card = dapm->card->snd_card; + + ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); + if (ret < 0) { + dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + goto error; + } + + ret = snd_hdac_regmap_init(&hcodec->core); + if (ret < 0) { + dev_err(&hdev->dev, "regmap init failed\n"); + goto error; + } + + patch = (hda_codec_patch_t)hcodec->preset->driver_data; + if (patch) { + ret = patch(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "patch failed %d\n", ret); + goto error; + } + } else { + dev_dbg(&hdev->dev, "no patch file found\n"); + } + + ret = snd_hda_codec_parse_pcms(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + goto error; + } + + ret = snd_hda_codec_build_controls(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to create controls %d\n", ret); + goto error; + } + + hcodec->core.lazy_cache = true; + + /* + * hdac_device core already sets the state to active and calls + * get_noresume. So enable runtime and set the device to suspend. + * pm_runtime_enable is also called during codec registeration + */ + pm_runtime_put(&hdev->dev); + pm_runtime_suspend(&hdev->dev); + + return 0; + +error: + pm_runtime_put(&hdev->dev); +error_no_pm: + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + return ret; +} + +static void hdac_hda_codec_remove(struct snd_soc_component *component) +{ + struct hdac_hda_priv *hda_pvt = + snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = &hda_pvt->codec.core; + struct hdac_ext_link *hlink = NULL; + + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return; + } + + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + pm_runtime_disable(&hdev->dev); +} + +static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = { + {"AIF1TX", NULL, "Codec Input Pin1"}, + {"AIF2TX", NULL, "Codec Input Pin2"}, + {"AIF3TX", NULL, "Codec Input Pin3"}, + + {"Codec Output Pin1", NULL, "AIF1RX"}, + {"Codec Output Pin2", NULL, "AIF2RX"}, + {"Codec Output Pin3", NULL, "AIF3RX"}, +}; + +static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + + /* Input Pins */ + SND_SOC_DAPM_INPUT("Codec Input Pin1"), + SND_SOC_DAPM_INPUT("Codec Input Pin2"), + SND_SOC_DAPM_INPUT("Codec Input Pin3"), + + /* Output Pins */ + SND_SOC_DAPM_OUTPUT("Codec Output Pin1"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin2"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin3"), +}; + +static const struct snd_soc_component_driver hdac_hda_codec = { + .probe = hdac_hda_codec_probe, + .remove = hdac_hda_codec_remove, + .idle_bias_on = false, + .dapm_widgets = hdac_hda_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdac_hda_dapm_widgets), + .dapm_routes = hdac_hda_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(hdac_hda_dapm_routes), +}; + +static int hdac_hda_dev_probe(struct hdac_device *hdev) +{ + struct hdac_ext_link *hlink; + struct hdac_hda_priv *hda_pvt; + int ret; + + /* hold the ref while we probe */ + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(hdev->bus, hlink); + + hda_pvt = hdac_to_hda_priv(hdev); + if (!hda_pvt) + return -ENOMEM; + + /* ASoC specific initialization */ + ret = snd_soc_register_component(&hdev->dev, + &hdac_hda_codec, hdac_hda_dais, + ARRAY_SIZE(hdac_hda_dais)); + if (ret < 0) { + dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + return ret; + } + + dev_set_drvdata(&hdev->dev, hda_pvt); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + + return ret; +} + +static int hdac_hda_dev_remove(struct hdac_device *hdev) +{ + snd_soc_unregister_component(&hdev->dev); + return 0; +} + +static struct hdac_ext_bus_ops hdac_ops = { + .hdev_attach = hdac_hda_dev_probe, + .hdev_detach = hdac_hda_dev_remove, +}; + +struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void) +{ + return &hdac_ops; +} +EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers"); +MODULE_AUTHOR("Rakesh Ughreja"); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h new file mode 100644 index 000000000000..e444ef593360 --- /dev/null +++ b/sound/soc/codecs/hdac_hda.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2015-18 Intel Corporation. + */ + +#ifndef __HDAC_HDA_H__ +#define __HDAC_HDA_H__ + +struct hdac_hda_pcm { + int stream_tag[2]; +}; + +struct hdac_hda_priv { + struct hda_codec codec; + struct hdac_hda_pcm pcm[2]; +}; + +#define hdac_to_hda_priv(_hdac) \ + container_of(_hdac, struct hdac_hda_priv, codec.core) +#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core) + +struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void); + +#endif /* __HDAC_HDA_H__ */ diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 0f0d57859555..88e4b4284738 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -283,6 +283,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" select SND_SOC_HDAC_HDMI + select SND_SOC_HDAC_HDA help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index f9917e0f2ba8..3fdbf239da74 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -69,6 +69,30 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .dpcm_playback = 1, .no_pcm = 1, }, + { + .name = "Analog Playback and Capture", + .id = 4, + .cpu_dai_name = "Analog CPU DAI", + .codec_name = "ehdaudio0D0", + .codec_dai_name = "Analog Codec DAI", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = NULL, + .no_pcm = 1, + }, + { + .name = "Digital Playback and Capture", + .id = 5, + .cpu_dai_name = "Digital CPU DAI", + .codec_name = "ehdaudio0D0", + .codec_dai_name = "Digital Codec DAI", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = NULL, + .no_pcm = 1, + }, }; int skl_hda_hdmi_jack_init(struct snd_soc_card *card) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index b6c79696bfba..87c50aff56cd 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -15,7 +15,7 @@ #include #include -#define HDA_DSP_MAX_BE_DAI_LINKS 3 +#define HDA_DSP_MAX_BE_DAI_LINKS 5 struct skl_hda_hdmi_pcm { struct list_head head; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 920bc2ce22aa..b213e9b47505 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -16,6 +16,15 @@ #include "../skylake/skl.h" #include "skl_hda_dsp_common.h" +static const struct snd_soc_dapm_widget skl_hda_widgets[] = { + SND_SOC_DAPM_HP("Analog Out", NULL), + SND_SOC_DAPM_MIC("Analog In", NULL), + SND_SOC_DAPM_HP("Alt Analog Out", NULL), + SND_SOC_DAPM_MIC("Alt Analog In", NULL), + SND_SOC_DAPM_SPK("Digital Out", NULL), + SND_SOC_DAPM_MIC("Digital In", NULL), +}; + static const struct snd_soc_dapm_route skl_hda_map[] = { { "hifi3", NULL, "iDisp3 Tx"}, { "iDisp3 Tx", NULL, "iDisp3_out"}, @@ -23,6 +32,29 @@ static const struct snd_soc_dapm_route skl_hda_map[] = { { "iDisp2 Tx", NULL, "iDisp2_out"}, { "hifi1", NULL, "iDisp1 Tx"}, { "iDisp1 Tx", NULL, "iDisp1_out"}, + + { "Analog Out", NULL, "Codec Output Pin1" }, + { "Digital Out", NULL, "Codec Output Pin2" }, + { "Alt Analog Out", NULL, "Codec Output Pin3" }, + + { "Codec Input Pin1", NULL, "Analog In" }, + { "Codec Input Pin2", NULL, "Digital In" }, + { "Codec Input Pin3", NULL, "Alt Analog In" }, + + /* CODEC BE connections */ + { "Analog Codec Playback", NULL, "Analog CPU Playback" }, + { "Analog CPU Playback", NULL, "codec0_out" }, + { "Digital Codec Playback", NULL, "Digital CPU Playback" }, + { "Digital CPU Playback", NULL, "codec1_out" }, + { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" }, + { "Alt Analog CPU Playback", NULL, "codec2_out" }, + + { "codec0_in", NULL, "Analog CPU Capture" }, + { "Analog CPU Capture", NULL, "Analog Codec Capture" }, + { "codec1_in", NULL, "Digital CPU Capture" }, + { "Digital CPU Capture", NULL, "Digital Codec Capture" }, + { "codec2_in", NULL, "Alt Analog CPU Capture" }, + { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, }; static int skl_hda_card_late_probe(struct snd_soc_card *card) @@ -57,6 +89,7 @@ static struct snd_soc_card hda_soc_card = { .name = "skl_hda_card", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, + .dapm_widgets = skl_hda_widgets, .dapm_routes = skl_hda_map, .add_dai_link = skl_hda_add_dai_link, .fully_routed = true, @@ -80,6 +113,11 @@ static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) { num_links = IDISP_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; + } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { + num_links = ARRAY_SIZE(skl_hda_be_dai_links); + num_route = ARRAY_SIZE(skl_hda_map), + card->dapm_widgets = skl_hda_widgets; + card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); } else { return -EINVAL; } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5f281d443a53..e7fd14daeb4f 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -37,6 +37,7 @@ #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +#include "../../../soc/codecs/hdac_hda.h" /* * initialize the PCI registers @@ -657,6 +658,24 @@ static void skl_clock_device_unregister(struct skl *skl) platform_device_unregister(skl->clk_dev); } +#define IDISP_INTEL_VENDOR_ID 0x80860000 + +/* + * load the legacy codec driver + */ +static void load_codec_module(struct hda_codec *codec) +{ +#ifdef MODULE + char modalias[MODULE_NAME_LEN]; + const char *mod = NULL; + + snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias)); + mod = modalias; + dev_dbg(&codec->core.dev, "loading %s codec module\n", mod); + request_module(mod); +#endif +} + /* * Probe the given codec address */ @@ -666,7 +685,9 @@ static int probe_codec(struct hdac_bus *bus, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; struct skl *skl = bus_to_skl(bus); + struct hdac_hda_priv *hda_codec; struct hdac_device *hdev; + int err; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -676,11 +697,24 @@ static int probe_codec(struct hdac_bus *bus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); - hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); - if (!hdev) + hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), + GFP_KERNEL); + if (!hda_codec) return -ENOMEM; - return snd_hdac_ext_bus_device_init(bus, addr, hdev); + hda_codec->codec.bus = skl_to_hbus(skl); + hdev = &hda_codec->codec.core; + + err = snd_hdac_ext_bus_device_init(bus, addr, hdev); + if (err < 0) + return err; + + /* use legacy bus only for HDA codecs, idisp uses ext bus */ + if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) { + hdev->type = HDA_DEV_LEGACY; + load_codec_module(&hda_codec->codec); + } + return 0; } /* Codec initialization */ @@ -815,6 +849,7 @@ static int skl_create(struct pci_dev *pci, const struct hdac_io_ops *io_ops, struct skl **rskl) { + struct hdac_ext_bus_ops *ext_ops = NULL; struct skl *skl; struct hdac_bus *bus; struct hda_bus *hbus; @@ -834,7 +869,11 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); - snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); + +#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) + ext_ops = snd_soc_hdac_hda_get_ops(); +#endif + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); -- cgit v1.2.3-58-ga151 From b0f2d651299f0743818bdadcbe6a67d7869e0da1 Mon Sep 17 00:00:00 2001 From: Danny Smith Date: Tue, 21 Aug 2018 13:07:49 +0200 Subject: ASoC: adau17x1: Implemented safeload support Safeload support has been implemented which is used when updating for instance filter parameters using alsa controls. Without safeload support audio can become distorted during update. Signed-off-by: Danny Smith Signed-off-by: Robert Rosengren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau17x1.c | 77 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..c0bc429249fa 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -21,11 +21,18 @@ #include #include #include +#include #include "sigmadsp.h" #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA 0x0001 +#define ADAU17X1_SAFELOAD_DATA_SIZE 20 +#define ADAU17X1_WORD_SIZE 4 + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -326,6 +333,17 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} + static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { @@ -957,6 +975,56 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[ADAU17X1_WORD_SIZE]; + uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE]; + unsigned int addr_offset; + unsigned int nbr_words; + int ret; + + /* write data to safeload addresses. Check if len is not a multiple of + * 4 bytes, if so we need to zero pad. + */ + nbr_words = len / ADAU17X1_WORD_SIZE; + if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) { + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, bytes, len); + } else { + nbr_words++; + memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE); + memcpy(data, bytes, len); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, data, + nbr_words * ADAU17X1_WORD_SIZE); + } + + if (ret < 0) + return ret; + + /* Write target address, target address is offset by 1 */ + addr_offset = addr - 1; + put_unaligned_be32(addr_offset, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + put_unaligned_be32(nbr_words, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + return 0; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1002,8 +1070,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); -- cgit v1.2.3-58-ga151 From 818838e6bfa4ddc6c76703237028dcffb80d6496 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 21 Aug 2018 13:43:35 +0200 Subject: ASoC: rt5670: Add quirk for Thinkpad 8 tablet The Thinkpad 8 needs a quirk for jack-detect and the internal mic to work correctly. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 732ef928b25d..455fe7cff700 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2875,6 +2875,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_DEV_GPIO | RT5670_JD_MODE1), }, + { + .callback = rt5670_quirk_cb, + .ident = "Lenovo Thinkpad Tablet 8", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_DEV_GPIO | + RT5670_JD_MODE1), + }, { .callback = rt5670_quirk_cb, .ident = "Lenovo Thinkpad Tablet 10", -- cgit v1.2.3-58-ga151 From 2ca426a24dd75e775ece1466ae45e019f0035b8d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 21 Aug 2018 13:43:36 +0200 Subject: ASoC: Intel: common: Add quirk for Thinkpad 8 tablet The Thinkpad 8 tablet uses 10EC5640 as ACPI HID, but it has a rt5670 codec add a quirk for this. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 4daa8a4f0c0c..097dc06377ba 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -30,6 +30,13 @@ static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) static const struct dmi_system_id byt_table[] = { + { + .callback = byt_thinkpad10_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"), + }, + }, { .callback = byt_thinkpad10_quirk_cb, .matches = { -- cgit v1.2.3-58-ga151 From f8fc397e13107f925186ee742e9e8dfbfe9a3d03 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 21 Aug 2018 13:43:37 +0200 Subject: ASoC: Intel: cht-bsw-rt5672: Add key-mappings for the headset buttons Having the headset buttons send BTN_0, BTN_1 and BTN_2 events is not really useful. Add mappings to PLAYPAUSE VOLUME_UP and VOLUME_DOWN like we do in other Intel machine drivers. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_rt5672.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index e5aa13058dd7..e054318185ea 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -16,6 +16,7 @@ * General Public License for more details. */ +#include #include #include #include @@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + rt5670_set_jack_detect(component, &ctx->headset); if (ctx->mclk) { /* -- cgit v1.2.3-58-ga151 From 6ee47d4a8dacfa484d526c0475730568d979de24 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Tue, 21 Aug 2018 18:52:46 +0200 Subject: ASoC: pcm3060: Add codec driver This commit adds support for TI PCM3060 CODEC. The technical documentation is available at [1]. [1] http://ti.com/product/pcm3060 Signed-off-by: Kirill Marinushkin Cc: Mark Brown Cc: Liam Girdwood Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: M R Swami Reddy Cc: Vishwas A Deshpande Cc: Kevin Cernekee Cc: Peter Ujfalusi Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/pcm3060.txt | 17 ++ MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 17 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3060-i2c.c | 61 +++++ sound/soc/codecs/pcm3060-spi.c | 60 +++++ sound/soc/codecs/pcm3060.c | 290 +++++++++++++++++++++ sound/soc/codecs/pcm3060.h | 88 +++++++ 8 files changed, 546 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm3060.txt create mode 100644 sound/soc/codecs/pcm3060-i2c.c create mode 100644 sound/soc/codecs/pcm3060-spi.c create mode 100644 sound/soc/codecs/pcm3060.c create mode 100644 sound/soc/codecs/pcm3060.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt new file mode 100644 index 000000000000..90fcb8523099 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -0,0 +1,17 @@ +PCM3060 audio CODEC + +This driver supports both I2C and SPI. + +Required properties: + +- compatible: "ti,pcm3060" + +- reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Examples: + + pcm3060: pcm3060@46 { + compatible = "ti,pcm3060"; + reg = <0x46>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index a5b256b25905..161b26e05732 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -14597,6 +14597,13 @@ L: netdev@vger.kernel.org S: Maintained F: drivers/net/ethernet/ti/netcp* +TI PCM3060 ASoC CODEC DRIVER +M: Kirill Marinushkin +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/pcm3060.txt +F: sound/soc/codecs/pcm3060* + TI TAS571X FAMILY ASoC CODEC DRIVER M: Kevin Cernekee L: alsa-devel@alsa-project.org (moderated for non-subscribers) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bf0b949eb7e8..adaf26e1989c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM186X_I2C if I2C select SND_SOC_PCM186X_SPI if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM3060_I2C if I2C + select SND_SOC_PCM3060_SPI if SPI_MASTER select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER select SND_SOC_PCM5102A @@ -737,6 +739,21 @@ config SND_SOC_PCM186X_SPI config SND_SOC_PCM3008 tristate +config SND_SOC_PCM3060 + tristate + +config SND_SOC_PCM3060_I2C + tristate "Texas Instruments PCM3060 CODEC - I2C" + depends on I2C + select SND_SOC_PCM3060 + select REGMAP_I2C + +config SND_SOC_PCM3060_SPI + tristate "Texas Instruments PCM3060 CODEC - SPI" + depends on SPI_MASTER + select SND_SOC_PCM3060 + select REGMAP_SPI + config SND_SOC_PCM3168A tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3046b33ca9d3..3d694c26192c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -120,6 +120,9 @@ snd-soc-pcm186x-objs := pcm186x.o snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o snd-soc-pcm186x-spi-objs := pcm186x-spi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm3060-objs := pcm3060.o +snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o +snd-soc-pcm3060-spi-objs := pcm3060-spi.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o @@ -381,6 +384,9 @@ obj-$(CONFIG_SND_SOC_PCM186X) += snd-soc-pcm186x.o obj-$(CONFIG_SND_SOC_PCM186X_I2C) += snd-soc-pcm186x-i2c.o obj-$(CONFIG_SND_SOC_PCM186X_SPI) += snd-soc-pcm186x-spi.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM3060) += snd-soc-pcm3060.o +obj-$(CONFIG_SND_SOC_PCM3060_I2C) += snd-soc-pcm3060-i2c.o +obj-$(CONFIG_SND_SOC_PCM3060_SPI) += snd-soc-pcm3060-spi.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c new file mode 100644 index 000000000000..03d2b4323626 --- /dev/null +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -0,0 +1,61 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * PCM3060 I2C driver + * + * Copyright (C) 2018 Kirill Marinushkin + */ + +#include +#include +#include + +#include "pcm3060.h" + +static int pcm3060_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct pcm3060_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + return pcm3060_probe(&i2c->dev); +} + +static const struct i2c_device_id pcm3060_i2c_id[] = { + { .name = "pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id pcm3060_of_match[] = { + { .compatible = "ti,pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(of, pcm3060_of_match); +#endif /* CONFIG_OF */ + +static struct i2c_driver pcm3060_i2c_driver = { + .driver = { + .name = "pcm3060", +#ifdef CONFIG_OF + .of_match_table = pcm3060_of_match, +#endif /* CONFIG_OF */ + }, + .id_table = pcm3060_i2c_id, + .probe = pcm3060_i2c_probe, +}; + +module_i2c_driver(pcm3060_i2c_driver); + +MODULE_DESCRIPTION("PCM3060 I2C driver"); +MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c new file mode 100644 index 000000000000..8961e095ae73 --- /dev/null +++ b/sound/soc/codecs/pcm3060-spi.c @@ -0,0 +1,60 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * PCM3060 SPI driver + * + * Copyright (C) 2018 Kirill Marinushkin + */ + +#include +#include +#include + +#include "pcm3060.h" + +static int pcm3060_spi_probe(struct spi_device *spi) +{ + struct pcm3060_priv *priv; + + priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + spi_set_drvdata(spi, priv); + + priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + return pcm3060_probe(&spi->dev); +} + +static const struct spi_device_id pcm3060_spi_id[] = { + { .name = "pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm3060_spi_id); + +#ifdef CONFIG_OF +static const struct of_device_id pcm3060_of_match[] = { + { .compatible = "ti,pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(of, pcm3060_of_match); +#endif /* CONFIG_OF */ + +static struct spi_driver pcm3060_spi_driver = { + .driver = { + .name = "pcm3060", +#ifdef CONFIG_OF + .of_match_table = pcm3060_of_match, +#endif /* CONFIG_OF */ + }, + .id_table = pcm3060_spi_id, + .probe = pcm3060_spi_probe, +}; + +module_spi_driver(pcm3060_spi_driver); + +MODULE_DESCRIPTION("PCM3060 SPI driver"); +MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c new file mode 100644 index 000000000000..ef7c627c9ac5 --- /dev/null +++ b/sound/soc/codecs/pcm3060.c @@ -0,0 +1,290 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * PCM3060 codec driver + * + * Copyright (C) 2018 Kirill Marinushkin + */ + +#include +#include +#include +#include + +#include "pcm3060.h" + +/* dai */ + +static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + + if (dir != SND_SOC_CLOCK_IN) { + dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir); + return -EINVAL; + } + + priv->dai[dai->id].sclk_freq = freq; + + return 0; +} + +static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + unsigned int reg; + unsigned int val; + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + priv->dai[dai->id].is_master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + priv->dai[dai->id].is_master = false; + break; + default: + dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val = PCM3060_REG_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = PCM3060_REG_FMT_RJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = PCM3060_REG_FMT_LJ; + break; + default: + dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt); + return -EINVAL; + } + + reg = (dai->id == PCM3060_DAI_ID_DAC ? PCM3060_REG67 : PCM3060_REG72); + + regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val); + + return 0; +} + +static int pcm3060_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + unsigned int rate; + unsigned int ratio; + unsigned int reg; + unsigned int val; + + if (!priv->dai[dai->id].is_master) { + val = PCM3060_REG_MS_S; + goto val_ready; + } + + rate = params_rate(params); + if (!rate) { + dev_err(comp->dev, "rate is not configured\n"); + return -EINVAL; + } + + ratio = priv->dai[dai->id].sclk_freq / rate; + + switch (ratio) { + case 768: + val = PCM3060_REG_MS_M768; + break; + case 512: + val = PCM3060_REG_MS_M512; + break; + case 384: + val = PCM3060_REG_MS_M384; + break; + case 256: + val = PCM3060_REG_MS_M256; + break; + case 192: + val = PCM3060_REG_MS_M192; + break; + case 128: + val = PCM3060_REG_MS_M128; + break; + default: + dev_err(comp->dev, "unsupported ratio: %d\n", ratio); + return -EINVAL; + } + +val_ready: + reg = (dai->id == PCM3060_DAI_ID_DAC ? PCM3060_REG67 : PCM3060_REG72); + + regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm3060_dai_ops = { + .set_sysclk = pcm3060_set_sysclk, + .set_fmt = pcm3060_set_fmt, + .hw_params = pcm3060_hw_params, +}; + +#define PCM3060_DAI_RATES_ADC (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PCM3060_DAI_RATES_DAC (PCM3060_DAI_RATES_ADC | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver pcm3060_dai[] = { + { + .name = "pcm3060-dac", + .id = PCM3060_DAI_ID_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM3060_DAI_RATES_DAC, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &pcm3060_dai_ops, + }, + { + .name = "pcm3060-adc", + .id = PCM3060_DAI_ID_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = PCM3060_DAI_RATES_ADC, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &pcm3060_dai_ops, + }, +}; + +/* dapm */ + +static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1); + +static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", + PCM3060_REG65, PCM3060_REG66, 0, + PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX, + 0, pcm3060_dapm_tlv), + SOC_DOUBLE("Master Playback Switch", PCM3060_REG68, + PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1), + + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", + PCM3060_REG70, PCM3060_REG71, 0, + PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX, + 0, pcm3060_dapm_tlv), + SOC_DOUBLE("Master Capture Switch", PCM3060_REG73, + PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1), +}; + +static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("OUTL+"), + SND_SOC_DAPM_OUTPUT("OUTR+"), + SND_SOC_DAPM_OUTPUT("OUTL-"), + SND_SOC_DAPM_OUTPUT("OUTR-"), + + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), +}; + +static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { + { "OUTL+", NULL, "Playback" }, + { "OUTR+", NULL, "Playback" }, + { "OUTL-", NULL, "Playback" }, + { "OUTR-", NULL, "Playback" }, + + { "Capture", NULL, "INL" }, + { "Capture", NULL, "INR" }, +}; + +/* soc component */ + +static const struct snd_soc_component_driver pcm3060_soc_comp_driver = { + .controls = pcm3060_dapm_controls, + .num_controls = ARRAY_SIZE(pcm3060_dapm_controls), + .dapm_widgets = pcm3060_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets), + .dapm_routes = pcm3060_dapm_map, + .num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map), +}; + +/* regmap */ + +static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg) +{ + return (reg >= PCM3060_REG64); +} + +static bool pcm3060_reg_readable(struct device *dev, unsigned int reg) +{ + return (reg >= PCM3060_REG64); +} + +static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg) +{ + /* PCM3060_REG64 is volatile */ + return (reg == PCM3060_REG64); +} + +static const struct reg_default pcm3060_reg_defaults[] = { + { PCM3060_REG64, 0xF0 }, + { PCM3060_REG65, 0xFF }, + { PCM3060_REG66, 0xFF }, + { PCM3060_REG67, 0x00 }, + { PCM3060_REG68, 0x00 }, + { PCM3060_REG69, 0x00 }, + { PCM3060_REG70, 0xD7 }, + { PCM3060_REG71, 0xD7 }, + { PCM3060_REG72, 0x00 }, + { PCM3060_REG73, 0x00 }, +}; + +const struct regmap_config pcm3060_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = pcm3060_reg_writeable, + .readable_reg = pcm3060_reg_readable, + .volatile_reg = pcm3060_reg_volatile, + .max_register = PCM3060_REG73, + .reg_defaults = pcm3060_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL(pcm3060_regmap); + +/* device */ + +int pcm3060_probe(struct device *dev) +{ + int rc; + + rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver, + pcm3060_dai, + ARRAY_SIZE(pcm3060_dai)); + if (rc) { + dev_err(dev, "failed to register component, rc=%d\n", rc); + return rc; + } + + return 0; +} +EXPORT_SYMBOL(pcm3060_probe); + +MODULE_DESCRIPTION("PCM3060 codec driver"); +MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h new file mode 100644 index 000000000000..fd89a68aa8a7 --- /dev/null +++ b/sound/soc/codecs/pcm3060.h @@ -0,0 +1,88 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * PCM3060 codec driver + * + * Copyright (C) 2018 Kirill Marinushkin + */ + +#ifndef _SND_SOC_PCM3060_H +#define _SND_SOC_PCM3060_H + +#include +#include + +extern const struct regmap_config pcm3060_regmap; + +#define PCM3060_DAI_ID_DAC 0 +#define PCM3060_DAI_ID_ADC 1 +#define PCM3060_DAI_IDS_NUM 2 + +struct pcm3060_priv_dai { + bool is_master; + unsigned int sclk_freq; +}; + +struct pcm3060_priv { + struct regmap *regmap; + struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM]; +}; + +int pcm3060_probe(struct device *dev); +int pcm3060_remove(struct device *dev); + +/* registers */ + +#define PCM3060_REG64 0x40 +#define PCM3060_REG_MRST 0x80 +#define PCM3060_REG_SRST 0x40 +#define PCM3060_REG_ADPSV 0x20 +#define PCM3060_REG_DAPSV 0x10 +#define PCM3060_REG_SE 0x01 + +#define PCM3060_REG65 0x41 +#define PCM3060_REG66 0x42 +#define PCM3060_REG_AT2_MIN 0x36 +#define PCM3060_REG_AT2_MAX 0xFF + +#define PCM3060_REG67 0x43 +#define PCM3060_REG72 0x48 +#define PCM3060_REG_CSEL 0x80 +#define PCM3060_REG_MASK_MS 0x70 +#define PCM3060_REG_MS_S 0x00 +#define PCM3060_REG_MS_M768 (0x01 << 4) +#define PCM3060_REG_MS_M512 (0x02 << 4) +#define PCM3060_REG_MS_M384 (0x03 << 4) +#define PCM3060_REG_MS_M256 (0x04 << 4) +#define PCM3060_REG_MS_M192 (0x05 << 4) +#define PCM3060_REG_MS_M128 (0x06 << 4) +#define PCM3060_REG_MASK_FMT 0x03 +#define PCM3060_REG_FMT_I2S 0x00 +#define PCM3060_REG_FMT_LJ 0x01 +#define PCM3060_REG_FMT_RJ 0x02 + +#define PCM3060_REG68 0x44 +#define PCM3060_REG_OVER 0x40 +#define PCM3060_REG_DREV2 0x04 +#define PCM3060_REG_SHIFT_MUT21 0x00 +#define PCM3060_REG_SHIFT_MUT22 0x01 + +#define PCM3060_REG69 0x45 +#define PCM3060_REG_FLT 0x80 +#define PCM3060_REG_MASK_DMF 0x60 +#define PCM3060_REG_DMC 0x10 +#define PCM3060_REG_ZREV 0x02 +#define PCM3060_REG_AZRO 0x01 + +#define PCM3060_REG70 0x46 +#define PCM3060_REG71 0x47 +#define PCM3060_REG_AT1_MIN 0x0E +#define PCM3060_REG_AT1_MAX 0xFF + +#define PCM3060_REG73 0x49 +#define PCM3060_REG_ZCDD 0x10 +#define PCM3060_REG_BYP 0x08 +#define PCM3060_REG_DREV1 0x04 +#define PCM3060_REG_SHIFT_MUT11 0x00 +#define PCM3060_REG_SHIFT_MUT12 0x01 + +#endif /* _SND_SOC_PCM3060_H */ -- cgit v1.2.3-58-ga151 From c736cbd3a668c3befd3ce08e0fbccac302fbecac Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Tue, 21 Aug 2018 12:25:05 +0530 Subject: ASoC: AMD: Set constraints for DMIC and MAX98357a codec We support dual channel, 48Khz. This constraint was set only for da7219. It is being extended to DMIC and MAX98357a. Signed-off-by: Akshu Agrawal Reviewed-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8e3275a96a82..95c5701e2a30 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -162,10 +162,21 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static int cz_max_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -177,20 +188,42 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) static int cz_dmic0_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); -- cgit v1.2.3-58-ga151 From a1b1e9880f0c2754a5ac416a546d9f295f72eabc Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Tue, 21 Aug 2018 12:29:43 +0530 Subject: ASoC: AMD: Change MCLK to 48Mhz 25Mhz MCLK which was earlier used was of spread type. Thus, we were not getting accurate rate. The 48Mhz system clk is of non-spread type and we are changing to it to get accurate rate. Signed-off-by: Akshu Agrawal Reviewed-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 95c5701e2a30..3879cccbd2c0 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -42,7 +42,7 @@ #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" -#define CZ_PLAT_CLK 25000000 +#define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -- cgit v1.2.3-58-ga151 From 1b3b7981524af2b6dcdf78cfa6dca89522c13ade Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 20 Aug 2018 12:14:09 +0200 Subject: ASoC: atmel: tse850: switch to SPDX license identifier Convert to // comments in the leading comment, drop the boilerplate license text and use the correct MODULE_LICENSE. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/atmel/tse850-pcm5142.c | 78 +++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 42 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 3a1393283156..214adcad5419 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -1,44 +1,38 @@ -/* - * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec - * - * Copyright (C) 2016 Axentia Technologies AB - * - * Author: Peter Rosin - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -/* - * loop1 relays - * IN1 +---o +------------+ o---+ OUT1 - * \ / - * + + - * | / | - * +--o +--. | - * | add | | - * | V | - * | .---. | - * DAC +----------->|Sum|---+ - * | '---' | - * | | - * + + - * - * IN2 +---o--+------------+--o---+ OUT2 - * loop2 relays - * - * The 'loop1' gpio pin controlls two relays, which are either in loop - * position, meaning that input and output are directly connected, or - * they are in mixer position, meaning that the signal is passed through - * the 'Sum' mixer. Similarly for 'loop2'. - * - * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the - * mixer (if 'add' is active) and feeding the mixer output to OUT1. The - * 'loop2' relays are active, short-cutting the TSE-850 from channel 2. - * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name - * of the (filtered) output from the PCM5142 codec. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec +// +// Copyright (C) 2016 Axentia Technologies AB +// +// Author: Peter Rosin +// +// loop1 relays +// IN1 +---o +------------+ o---+ OUT1 +// \ / +// + + +// | / | +// +--o +--. | +// | add | | +// | V | +// | .---. | +// DAC +----------->|Sum|---+ +// | '---' | +// | | +// + + +// +// IN2 +---o--+------------+--o---+ OUT2 +// loop2 relays +// +// The 'loop1' gpio pin controlls two relays, which are either in loop +// position, meaning that input and output are directly connected, or +// they are in mixer position, meaning that the signal is passed through +// the 'Sum' mixer. Similarly for 'loop2'. +// +// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the +// mixer (if 'add' is active) and feeding the mixer output to OUT1. The +// 'loop2' relays are active, short-cutting the TSE-850 from channel 2. +// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name +// of the (filtered) output from the PCM5142 codec. #include #include @@ -452,4 +446,4 @@ module_platform_driver(tse850_driver); /* Module information */ MODULE_AUTHOR("Peter Rosin "); MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-58-ga151 From aec785f6a0dcd68c3d2ad4a7d5b48d5fc94d75e8 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Tue, 28 Aug 2018 23:42:30 +0200 Subject: ASoC: pcm3060: Improve stylistics of file comments Modified the complete file comments in C++ style, to make them look more intentional Signed-off-by: Kirill Marinushkin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3060-i2c.c | 9 ++++----- sound/soc/codecs/pcm3060-spi.c | 9 ++++----- sound/soc/codecs/pcm3060.c | 9 ++++----- 3 files changed, 12 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c index 03d2b4323626..cdc8314882bc 100644 --- a/sound/soc/codecs/pcm3060-i2c.c +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -1,9 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * PCM3060 I2C driver - * - * Copyright (C) 2018 Kirill Marinushkin - */ +// +// PCM3060 I2C driver +// +// Copyright (C) 2018 Kirill Marinushkin #include #include diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c index 8961e095ae73..f6f19fa80932 100644 --- a/sound/soc/codecs/pcm3060-spi.c +++ b/sound/soc/codecs/pcm3060-spi.c @@ -1,9 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * PCM3060 SPI driver - * - * Copyright (C) 2018 Kirill Marinushkin - */ +// +// PCM3060 SPI driver +// +// Copyright (C) 2018 Kirill Marinushkin #include #include diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index ef7c627c9ac5..5b9718fa766d 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -1,9 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * PCM3060 codec driver - * - * Copyright (C) 2018 Kirill Marinushkin - */ +// +// PCM3060 codec driver +// +// Copyright (C) 2018 Kirill Marinushkin #include #include -- cgit v1.2.3-58-ga151 From 080aaf10892eec4b359126473e582f62ebb09496 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Tue, 28 Aug 2018 23:42:31 +0200 Subject: ASoC: pcm3060: Improve legibility of if-statements Modified some if-statements to make them more clear Signed-off-by: Kirill Marinushkin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3060.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 5b9718fa766d..494d9d662be8 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -68,7 +68,10 @@ static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - reg = (dai->id == PCM3060_DAI_ID_DAC ? PCM3060_REG67 : PCM3060_REG72); + if (dai->id == PCM3060_DAI_ID_DAC) + reg = PCM3060_REG67; + else + reg = PCM3060_REG72; regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val); @@ -124,7 +127,10 @@ static int pcm3060_hw_params(struct snd_pcm_substream *substream, } val_ready: - reg = (dai->id == PCM3060_DAI_ID_DAC ? PCM3060_REG67 : PCM3060_REG72); + if (dai->id == PCM3060_DAI_ID_DAC) + reg = PCM3060_REG67; + else + reg = PCM3060_REG72; regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val); -- cgit v1.2.3-58-ga151 From dba508b5ab1d138bd7543a3f503492f1a173aa32 Mon Sep 17 00:00:00 2001 From: Robert Rosengren Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Signed-off-by: Robert Rosengren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index c0bc429249fa..3959e6ad113d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -67,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -320,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -331,7 +334,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static bool adau17x1_has_safeload(struct adau *adau) { @@ -854,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -898,7 +900,6 @@ err: return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- cgit v1.2.3-58-ga151 From 26bcf1c368d9460987d597fb0476d60e51a1bf82 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 29 Aug 2018 17:00:48 +0200 Subject: ASoC: dmic: add Kconfig prompt for the generic dmic codec. Add Kconfig prompt for the generic digital mic to make it configurable through menuconfig Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index adaf26e1989c..9989d35e0fc6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -578,7 +578,11 @@ config SND_SOC_DA9055 tristate config SND_SOC_DMIC - tristate + tristate "Generic Digital Microphone CODEC" + depends on GPIOLIB + help + Enable support for the Generic Digital Microphone CODEC. + Select this if your sound card has DMICs. config SND_SOC_HDMI_CODEC tristate -- cgit v1.2.3-58-ga151 From cb06a037f8362e250a6e61872ffa01ab086ec9e2 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 29 Aug 2018 17:00:49 +0200 Subject: ASoC: dmic: add DT module alias Before this patch the only alias provided by the dmic module is: alias: platform:dmic-codec Device instantiated from DT will not probe automatically with this After this patch, here is the new alias list: alias: platform:dmic-codec alias: of:N*T*Cdmic-codecC* alias: of:N*T*Cdmic-codec Now the dmic codec probes automatically when instantiated from DT. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/dmic.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 8c4926df9286..71322e0410ee 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = { {.compatible = "dmic-codec"}, {} }; +MODULE_DEVICE_TABLE(of, dmic_dev_match); static struct platform_driver dmic_driver = { .driver = { -- cgit v1.2.3-58-ga151 From 2cfc123eea7477f26f59506fb45f25cb09ee1591 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 29 Aug 2018 17:00:51 +0200 Subject: ASoC: meson: add axg pdm input Add pdm input driver for the device found on the amlogic AXG SoC family Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-pdm.c | 654 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 665 insertions(+) create mode 100644 sound/soc/meson/axg-pdm.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 2ccbadc387de..8b8426ed2363 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -54,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD imply SND_MESON_AXG_TDMIN imply SND_MESON_AXG_TDMOUT imply SND_MESON_AXG_SPDIFOUT + imply SND_MESON_AXG_PDM help Select Y or M to add support for the AXG SoC sound card @@ -66,4 +67,12 @@ config SND_MESON_AXG_SPDIFOUT Select Y or M to add support for SPDIF output serializer embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_PDM + tristate "Amlogic AXG PDM Input Support" + imply SND_SOC_DMIC + imply COMMON_CLK_AXG_AUDIO + help + Select Y or M to add support for PDM input embedded + in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index c5e003b093db..4cd25104029d 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -9,6 +9,7 @@ snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o +snd-soc-meson-axg-pdm-objs := axg-pdm.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o +obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c new file mode 100644 index 000000000000..9d5684493ffc --- /dev/null +++ b/sound/soc/meson/axg-pdm.c @@ -0,0 +1,654 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include + +#define PDM_CTRL 0x00 +#define PDM_CTRL_EN BIT(31) +#define PDM_CTRL_OUT_MODE BIT(29) +#define PDM_CTRL_BYPASS_MODE BIT(28) +#define PDM_CTRL_RST_FIFO BIT(16) +#define PDM_CTRL_CHAN_RSTN_MASK GENMASK(15, 8) +#define PDM_CTRL_CHAN_RSTN(x) ((x) << 8) +#define PDM_CTRL_CHAN_EN_MASK GENMASK(7, 0) +#define PDM_CTRL_CHAN_EN(x) ((x) << 0) +#define PDM_HCIC_CTRL1 0x04 +#define PDM_FILTER_EN BIT(31) +#define PDM_HCIC_CTRL1_GAIN_SFT_MASK GENMASK(29, 24) +#define PDM_HCIC_CTRL1_GAIN_SFT(x) ((x) << 24) +#define PDM_HCIC_CTRL1_GAIN_MULT_MASK GENMASK(23, 16) +#define PDM_HCIC_CTRL1_GAIN_MULT(x) ((x) << 16) +#define PDM_HCIC_CTRL1_DSR_MASK GENMASK(8, 4) +#define PDM_HCIC_CTRL1_DSR(x) ((x) << 4) +#define PDM_HCIC_CTRL1_STAGE_NUM_MASK GENMASK(3, 0) +#define PDM_HCIC_CTRL1_STAGE_NUM(x) ((x) << 0) +#define PDM_HCIC_CTRL2 0x08 +#define PDM_F1_CTRL 0x0c +#define PDM_LPF_ROUND_MODE_MASK GENMASK(17, 16) +#define PDM_LPF_ROUND_MODE(x) ((x) << 16) +#define PDM_LPF_DSR_MASK GENMASK(15, 12) +#define PDM_LPF_DSR(x) ((x) << 12) +#define PDM_LPF_STAGE_NUM_MASK GENMASK(8, 0) +#define PDM_LPF_STAGE_NUM(x) ((x) << 0) +#define PDM_LPF_MAX_STAGE 336 +#define PDM_LPF_NUM 3 +#define PDM_F2_CTRL 0x10 +#define PDM_F3_CTRL 0x14 +#define PDM_HPF_CTRL 0x18 +#define PDM_HPF_SFT_STEPS_MASK GENMASK(20, 16) +#define PDM_HPF_SFT_STEPS(x) ((x) << 16) +#define PDM_HPF_OUT_FACTOR_MASK GENMASK(15, 0) +#define PDM_HPF_OUT_FACTOR(x) ((x) << 0) +#define PDM_CHAN_CTRL 0x1c +#define PDM_CHAN_CTRL_POINTER_WIDTH 8 +#define PDM_CHAN_CTRL_POINTER_MAX ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1) +#define PDM_CHAN_CTRL_NUM 4 +#define PDM_CHAN_CTRL1 0x20 +#define PDM_COEFF_ADDR 0x24 +#define PDM_COEFF_DATA 0x28 +#define PDM_CLKG_CTRL 0x2c +#define PDM_STS 0x30 + +struct axg_pdm_lpf { + unsigned int ds; + unsigned int round_mode; + const unsigned int *tap; + unsigned int tap_num; +}; + +struct axg_pdm_hcic { + unsigned int shift; + unsigned int mult; + unsigned int steps; + unsigned int ds; +}; + +struct axg_pdm_hpf { + unsigned int out_factor; + unsigned int steps; +}; + +struct axg_pdm_filters { + struct axg_pdm_hcic hcic; + struct axg_pdm_hpf hpf; + struct axg_pdm_lpf lpf[PDM_LPF_NUM]; +}; + +struct axg_pdm_cfg { + const struct axg_pdm_filters *filters; + unsigned int sys_rate; +}; + +struct axg_pdm { + const struct axg_pdm_cfg *cfg; + struct regmap *map; + struct clk *dclk; + struct clk *sysclk; + struct clk *pclk; +}; + +static void axg_pdm_enable(struct regmap *map) +{ + /* Reset AFIFO */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO); + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0); + + /* Enable PDM */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN); +} + +static void axg_pdm_disable(struct regmap *map) +{ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0); +} + +static void axg_pdm_filters_enable(struct regmap *map, bool enable) +{ + unsigned int val = enable ? PDM_FILTER_EN : 0; + + regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val); +} + +static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_pdm_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_pdm_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static unsigned int axg_pdm_get_os(struct axg_pdm *priv) +{ + const struct axg_pdm_filters *filters = priv->cfg->filters; + unsigned int os = filters->hcic.ds; + int i; + + /* + * The global oversampling factor is defined by the down sampling + * factor applied by each filter (HCIC and LPFs) + */ + + for (i = 0; i < PDM_LPF_NUM; i++) + os *= filters->lpf[i].ds; + + return os; +} + +static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os, + unsigned int rate) +{ + unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX; + + /* + * Set the default system clock rate unless it is too fast for + * for the requested sample rate. In this case, the sample pointer + * counter could overflow so set a lower system clock rate + */ + if (sys_rate < priv->cfg->sys_rate) + return clk_set_rate(priv->sysclk, sys_rate); + + return clk_set_rate(priv->sysclk, priv->cfg->sys_rate); +} + +static int axg_pdm_set_sample_pointer(struct axg_pdm *priv) +{ + unsigned int spmax, sp, val; + int i; + + /* Max sample counter value per half period of dclk */ + spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk), + clk_get_rate(priv->dclk) * 2); + + /* Check if sysclk is not too fast - should not happen */ + if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX)) + return -EINVAL; + + /* Capture the data when we are at 75% of the half period */ + sp = spmax * 3 / 4; + + for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++) + val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i); + + regmap_write(priv->map, PDM_CHAN_CTRL, val); + regmap_write(priv->map, PDM_CHAN_CTRL1, val); + + return 0; +} + +static void axg_pdm_set_channel_mask(struct axg_pdm *priv, + unsigned int channels) +{ + unsigned int mask = GENMASK(channels - 1, 0); + + /* Put all channel in reset */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK, 0); + + /* Take the necessary channels out of reset and enable them */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK | + PDM_CTRL_CHAN_EN_MASK, + PDM_CTRL_CHAN_RSTN(mask) | + PDM_CTRL_CHAN_EN(mask)); +} + +static int axg_pdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + unsigned int os = axg_pdm_get_os(priv); + unsigned int rate = params_rate(params); + unsigned int val; + int ret; + + switch (params_width(params)) { + case 24: + val = PDM_CTRL_OUT_MODE; + break; + case 32: + val = 0; + break; + default: + dev_err(dai->dev, "unsupported sample width\n"); + return -EINVAL; + } + + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val); + + ret = axg_pdm_set_sysclk(priv, os, rate); + if (ret) { + dev_err(dai->dev, "failed to set system clock\n"); + return ret; + } + + ret = clk_set_rate(priv->dclk, rate * os); + if (ret) { + dev_err(dai->dev, "failed to set dclk\n"); + return ret; + } + + ret = axg_pdm_set_sample_pointer(priv); + if (ret) { + dev_err(dai->dev, "invalid clock setting\n"); + return ret; + } + + axg_pdm_set_channel_mask(priv, params_channels(params)); + + return 0; +} + +static int axg_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->dclk); + if (ret) { + dev_err(dai->dev, "enabling dclk failed\n"); + return ret; + } + + /* Enable the filters */ + axg_pdm_filters_enable(priv->map, true); + + return ret; +} + +static void axg_pdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + axg_pdm_filters_enable(priv->map, false); + clk_disable_unprepare(priv->dclk); +} + +static const struct snd_soc_dai_ops axg_pdm_dai_ops = { + .trigger = axg_pdm_trigger, + .hw_params = axg_pdm_hw_params, + .startup = axg_pdm_startup, + .shutdown = axg_pdm_shutdown, +}; + +static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic; + unsigned int val; + + val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps); + val |= PDM_HCIC_CTRL1_DSR(hcic->ds); + val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult); + val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift); + + regmap_update_bits(priv->map, PDM_HCIC_CTRL1, + PDM_HCIC_CTRL1_STAGE_NUM_MASK | + PDM_HCIC_CTRL1_DSR_MASK | + PDM_HCIC_CTRL1_GAIN_MULT_MASK | + PDM_HCIC_CTRL1_GAIN_SFT_MASK, + val); +} + +static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index) +{ + const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index]; + unsigned int offset = index * regmap_get_reg_stride(priv->map) + + PDM_F1_CTRL; + unsigned int val; + + val = PDM_LPF_STAGE_NUM(lpf->tap_num); + val |= PDM_LPF_DSR(lpf->ds); + val |= PDM_LPF_ROUND_MODE(lpf->round_mode); + + regmap_update_bits(priv->map, offset, + PDM_LPF_STAGE_NUM_MASK | + PDM_LPF_DSR_MASK | + PDM_LPF_ROUND_MODE_MASK, + val); +} + +static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf; + unsigned int val; + + val = PDM_HPF_OUT_FACTOR(hpf->out_factor); + val |= PDM_HPF_SFT_STEPS(hpf->steps); + + regmap_update_bits(priv->map, PDM_HPF_CTRL, + PDM_HPF_OUT_FACTOR_MASK | + PDM_HPF_SFT_STEPS_MASK, + val); +} + +static int axg_pdm_set_lpf_filters(struct axg_pdm *priv) +{ + const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf; + unsigned int count = 0; + int i, j; + + for (i = 0; i < PDM_LPF_NUM; i++) + count += lpf[i].tap_num; + + /* Make sure the coeffs fit in the memory */ + if (count >= PDM_LPF_MAX_STAGE) + return -EINVAL; + + /* Set the initial APB bus register address */ + regmap_write(priv->map, PDM_COEFF_ADDR, 0); + + /* Set the tap filter values of all 3 filters */ + for (i = 0; i < PDM_LPF_NUM; i++) { + axg_pdm_set_lpf_ctrl(priv, i); + + for (j = 0; j < lpf[i].tap_num; j++) + regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]); + } + + return 0; +} + +static int axg_pdm_dai_probe(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "enabling pclk failed\n"); + return ret; + } + + /* + * sysclk must be set and enabled as well to access the pdm registers + * Accessing the register w/o it will give a bus error. + */ + ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate); + if (ret) { + dev_err(dai->dev, "setting sysclk failed\n"); + goto err_pclk; + } + + ret = clk_prepare_enable(priv->sysclk); + if (ret) { + dev_err(dai->dev, "enabling sysclk failed\n"); + goto err_pclk; + } + + /* Make sure the device is initially disabled */ + axg_pdm_disable(priv->map); + + /* Make sure filter bypass is disabled */ + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0); + + /* Load filter settings */ + axg_pdm_set_hcic_ctrl(priv); + axg_pdm_set_hpf_ctrl(priv); + + ret = axg_pdm_set_lpf_filters(priv); + if (ret) { + dev_err(dai->dev, "invalid filter configuration\n"); + goto err_sysclk; + } + + return 0; + +err_sysclk: + clk_disable_unprepare(priv->sysclk); +err_pclk: + clk_disable_unprepare(priv->pclk); + return ret; +} + +static int axg_pdm_dai_remove(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->sysclk); + clk_disable_unprepare(priv->pclk); + + return 0; +} + +static struct snd_soc_dai_driver axg_pdm_dai_drv = { + .name = "PDM", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 48000, + .formats = (SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &axg_pdm_dai_ops, + .probe = axg_pdm_dai_probe, + .remove = axg_pdm_dai_remove, +}; + +static const struct snd_soc_component_driver axg_pdm_component_drv = {}; + +static const struct regmap_config axg_pdm_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = PDM_STS, +}; + +static const unsigned int lpf1_default_tap[] = { + 0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45, + 0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21, + 0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212, + 0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa, + 0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef, + 0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077, + 0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690, + 0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3, + 0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686, + 0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812, + 0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60, + 0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000, + 0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e, + 0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c, + 0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece, + 0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645, + 0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9, + 0xffffb2, 0x000014, +}; + +static const unsigned int lpf2_default_tap[] = { + 0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818, + 0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41, + 0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0, + 0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000, + 0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93, + 0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12, + 0x0002c1, 0xfff004, 0x00050a, +}; + +static const unsigned int lpf3_default_tap[] = { + 0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000, + 0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7, + 0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000, + 0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0, + 0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000, + 0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7, + 0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000, + 0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45, + 0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000, + 0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c, + 0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000, + 0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216, + 0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000, + 0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d, + 0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000, + 0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368, + 0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000, + 0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb, + 0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000, + 0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c, + 0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000, + 0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46, + 0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000, + 0x000081, 0x000000, +}; + +/* + * These values are sane defaults for the axg platform: + * - OS = 64 + * - Latency = 38700 (?) + * + * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible. + * the configuration may depend on the dmic used by the platform, the + * expected tradeoff between latency and quality, etc ... If/When other + * settings are required, we should add a fw interface to this driver to + * load new filter settings. + */ +static const struct axg_pdm_filters axg_default_filters = { + .hcic = { + .shift = 0x15, + .mult = 0x80, + .steps = 7, + .ds = 8, + }, + .hpf = { + .out_factor = 0x8000, + .steps = 13, + }, + .lpf = { + [0] = { + .ds = 2, + .round_mode = 1, + .tap = lpf1_default_tap, + .tap_num = ARRAY_SIZE(lpf1_default_tap), + }, + [1] = { + .ds = 2, + .round_mode = 0, + .tap = lpf2_default_tap, + .tap_num = ARRAY_SIZE(lpf2_default_tap), + }, + [2] = { + .ds = 2, + .round_mode = 1, + .tap = lpf3_default_tap, + .tap_num = ARRAY_SIZE(lpf3_default_tap) + }, + }, +}; + +static const struct axg_pdm_cfg axg_pdm_config = { + .filters = &axg_default_filters, + .sys_rate = 250000000, +}; + +static const struct of_device_id axg_pdm_of_match[] = { + { + .compatible = "amlogic,axg-pdm", + .data = &axg_pdm_config, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_pdm_of_match); + +static int axg_pdm_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_pdm *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->cfg = of_device_get_match_data(dev); + if (!priv->cfg) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->dclk = devm_clk_get(dev, "dclk"); + if (IS_ERR(priv->dclk)) { + ret = PTR_ERR(priv->dclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + priv->sysclk = devm_clk_get(dev, "sysclk"); + if (IS_ERR(priv->sysclk)) { + ret = PTR_ERR(priv->sysclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_pdm_component_drv, + &axg_pdm_dai_drv, 1); +} + +static struct platform_driver axg_pdm_pdrv = { + .probe = axg_pdm_probe, + .driver = { + .name = "axg-pdm", + .of_match_table = axg_pdm_of_match, + }, +}; +module_platform_driver(axg_pdm_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG PDM Input driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-58-ga151 From 5fcb457ac2fdc62c5ff6d3962c958b1301bc5ea1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 31 Aug 2018 11:24:56 +0300 Subject: ASoC: davinci-mcasp: Add support for FIFO usage caused delay reporting McASP have write and read FIFO, each 64 words deep. From the WFIFOS/RFIFOS registers we can read the amount of data currently in the FIFO which can be directly reported as delay. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f70db8412c7c..267aee776b2d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1041,6 +1041,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, return error_ppm; } +static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->txnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET); +} + +static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->rxnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET); +} + +static snd_pcm_sframes_t davinci_mcasp_delay( + struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + u32 fifo_use; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_use = davinci_mcasp_tx_delay(mcasp); + else + fifo_use = davinci_mcasp_rx_delay(mcasp); + + /* + * Divide the used locations with the channel count to get the + * FIFO usage in samples (don't care about partial samples in the + * buffer). + */ + return fifo_use / substream->runtime->channels; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -1365,6 +1401,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .shutdown = davinci_mcasp_shutdown, .trigger = davinci_mcasp_trigger, + .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, -- cgit v1.2.3-58-ga151 From ec94c177bf3700ce44c53c375a3fb4c347f2b08f Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg Date: Fri, 31 Aug 2018 09:47:13 -0500 Subject: ASoC: codecs: tas5720: add TAS5722 specific volume control The TAS5722 supports modifying volume in 0.25dB steps (as opposed to 0.5dB steps on the TAS5720). Introduce a custom mixer control that allows taking advantage of this finer output volume granularity. Also add custom getters/setters for access as the TAS5722 digital volume controls are split over two registers. Signed-off-by: Andreas Dannenberg Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tas5720.c | 88 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 80 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index ae3d032ac35a..3c469112477a 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -485,15 +485,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, ); /* - * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that - * setting the gain below -100 dB (register value <0x7) is effectively a MUTE - * as per device datasheet. + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps + * depending on the device. Note that setting the gain below -100 dB + * (register value <0x7) is effectively a MUTE as per device datasheet. + * + * Note that for the TAS5722 the digital volume controls are actually split + * over two registers, so we need custom getters/setters for access. */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0); + +static int tas5722_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int val; + + snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val); + ucontrol->value.integer.value[0] = val << 1; + + snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val); + ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB; + + return 0; +} + +static int tas5722_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int sel = ucontrol->value.integer.value[0]; + + snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1); + snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_VOL_CONTROL_LSB, sel); + + return 0; +} static const struct snd_kcontrol_new tas5720_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", - TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_kcontrol_new tas5722_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume", + 0, 0, 511, 0, + tas5722_volume_get, tas5722_volume_set, + tas5722_dac_tlv), SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), }; @@ -527,6 +568,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_component_dev_tas5722 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -613,9 +671,23 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, ARRAY_SIZE(tas5720_dai)); + switch (id->driver_data) { + case TAS5720: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5720, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + case TAS5722: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5722, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + default: + dev_err(dev, "unexpected private driver data\n"); + return -EINVAL; + } if (ret < 0) { dev_err(dev, "failed to register component: %d\n", ret); return ret; -- cgit v1.2.3-58-ga151 From db658f40cae33a9fddbd9ca5c35c6bbfbd593a82 Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg Date: Fri, 31 Aug 2018 09:47:14 -0500 Subject: ASoC: codecs: tas5720: add TAS5722 TDM slot width setting support Unlike the TAS5720, the TAS5722 can be configured to utilize 16-bit wide slots in TDM mode. This can help easing audio clocking/frequency requirements. Signed-off-by: Andreas Dannenberg Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tas5720.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 3c469112477a..6bd0e5d5347f 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component); unsigned int first_slot; int ret; @@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) goto error_snd_soc_component_update_bits; + /* Configure TDM slot width. This is only applicable to TAS5722. */ + switch (tas5720->devtype) { + case TAS5722: + ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_TDM_SLOT_16B, + slot_width == 16 ? + TAS5722_TDM_SLOT_16B : 0); + if (ret < 0) + goto error_snd_soc_component_update_bits; + break; + default: + break; + } + return 0; error_snd_soc_component_update_bits: -- cgit v1.2.3-58-ga151 From 6f18bcdaa24bae39c746b57b95af19ff3c41b17f Mon Sep 17 00:00:00 2001 From: Matt Flax Date: Thu, 30 Aug 2018 09:38:00 +1000 Subject: ASoC: cs4265: SOC_SINGLE register value error fix The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de669f..15b4ae04870f 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- cgit v1.2.3-58-ga151 From be47e75eb1419ffc1d9c26230963fd5fa3055097 Mon Sep 17 00:00:00 2001 From: Matt Flax Date: Thu, 30 Aug 2018 09:38:01 +1000 Subject: ASoC: cs4265: Add native 32bit I2S transport The cs4265 uses 32 bit transport on the I2S bus. This patch enables native 32 bit mode for machine drivers which use this sound card driver. Signed-off-by: Matt Flax Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 15b4ae04870f..17d7e6f0dcdb 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -495,7 +495,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component, SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) static const struct snd_soc_dai_ops cs4265_ops = { .hw_params = cs4265_pcm_hw_params, -- cgit v1.2.3-58-ga151 From f853d6b3ba345297974d877d8ed0f4a91eaca739 Mon Sep 17 00:00:00 2001 From: Matt Flax Date: Thu, 30 Aug 2018 09:38:02 +1000 Subject: ASoC: cs4265: Add a S/PDIF enable switch This patch adds a S/PDIF enable switch as a SOC_SINGLE. Signed-off-by: Matt Flax Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 17d7e6f0dcdb..d9eebf6af7a8 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -154,6 +154,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), + SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), -- cgit v1.2.3-58-ga151 From e664de680b10c13d65f982bcb9cfe56096e1de55 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:08:09 +0000 Subject: ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple_card_utils for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 27 ++++++++++++++-------- sound/soc/generic/simple-card-utils.c | 42 +++++++++++++++++++++++++++++++++-- 2 files changed, 58 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 8bc5e2d8b13c..3b5bd6e76f88 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -51,29 +51,35 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, #define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \ asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ - dai_link->cpu_dai_name) + dai_link->cpu_dai_name, NULL) #define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \ asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ - dai_link->codec_dai_name) + dai_link->codec_dai_name, dai_link->codecs) int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, - const char *name); + const char *dai_name, + struct snd_soc_dai_link_component *dlc); int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ - asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node, \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name, list_name, cells_name, is_single_link) #define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, &dai_link->codec_of_node, \ - &dai_link->codec_dai_name, list_name, cells_name, NULL) + asoc_simple_card_parse_dai(node, dai_link->codecs, \ + &dai_link->codec_of_node, \ + &dai_link->codec_dai_name, \ + list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, &dai_link->platform_of_node, \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, struct device_node **endpoint_np, const char **dai_name, const char *list_name, @@ -81,12 +87,15 @@ int asoc_simple_card_parse_dai(struct device_node *node, int *is_single_links); #define asoc_simple_card_parse_graph_cpu(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node, \ + asoc_simple_card_parse_graph_dai(ep, NULL, \ + &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name) #define asoc_simple_card_parse_graph_codec(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node, \ + asoc_simple_card_parse_graph_dai(ep, dai_link->codecs, \ + &dai_link->codec_of_node, \ &dai_link->codec_dai_name) int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, struct device_node **endpoint_np, const char **dai_name); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d3f3f0fec74c..73c0a904f32e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,11 +173,23 @@ int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, - const char *name) + const char *dai_name, + struct snd_soc_dai_link_component *dlc) { struct clk *clk; u32 val; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_of_node = dlc->of_node; + dai_name = dlc->dai_name; + } + /* * Parse dai->sysclk come from "clocks = <&xxx>" * (if system has common clock) @@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; - dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name, + dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name, simple_dai->sysclk, simple_dai->clk_direction); return 0; @@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev, EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name, const char *list_name, @@ -220,6 +233,17 @@ int asoc_simple_card_parse_dai(struct device_node *node, if (!node) return 0; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + /* * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() @@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) } int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name) { @@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, struct of_phandle_args args; int ret; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + if (!ep) return 0; if (!dai_name) @@ -374,6 +410,8 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) num_links++, dai_link++) { of_node_put(dai_link->cpu_of_node); of_node_put(dai_link->codec_of_node); + if (dai_link->codecs) + of_node_put(dai_link->codecs->of_node); } return 0; } -- cgit v1.2.3-58-ga151 From 710af9196ce614ee02185c2ec55e617a71843183 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:08:24 +0000 Subject: ASoC: simple-card: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-card for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 64bf3560c1d1..67a56f32f983 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -20,6 +20,7 @@ struct simple_card_data { struct simple_dai_props { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -234,7 +235,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; @@ -363,7 +364,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct snd_soc_card *card; - int num, ret; + int num, ret, i; /* Get the number of DAI links */ if (np && of_get_child_by_name(np, PREFIX "dai-link")) @@ -381,6 +382,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < num; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; @@ -403,6 +415,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } else { struct asoc_simple_card_info *cinfo; + struct snd_soc_dai_link_component *codecs; cinfo = dev->platform_data; if (!cinfo) { @@ -419,13 +432,15 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } + codecs = dai_link->codecs; + codecs->name = cinfo->codec; + codecs->dai_name = cinfo->codec_dai.name; + card->name = (cinfo->card) ? cinfo->card : cinfo->name; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; - dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; - dai_link->codec_dai_name = cinfo->codec_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, -- cgit v1.2.3-58-ga151 From 5ece10ab99205d11419e3d2e6c7ac0e382d952b5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:08:38 +0000 Subject: ASoC: simple-scu-card: use simple_dai_props simple-card and simple-scu-card are very similar driver, but using different feature. Thus we are keeping synchronization on these 2 drivers style, because it is easy to confirm / check. Current big difference between these 2 drivers are "dai_props" on simple_card_data (= priv). It will be difficult to keep synchronize if we will add new feature on simple-scu-card. Thus, this patch synchronize it. [simple] struct simple_card_data { ... struct simple_dai_props { ... } *dai_props; ... }; [simple scu] struct simple_card_data { ... struct asoc_simple_dai *dai_props; ... }; Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 16a83bc51e0e..09be02e7416f 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -22,7 +22,9 @@ struct simple_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct simple_dai_props { + struct asoc_simple_dai dai; + } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; }; @@ -40,20 +42,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - return asoc_simple_card_clk_enable(dai_props); + return asoc_simple_card_clk_enable(&dai_props->dai); } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(&dai_props->dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -66,7 +68,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct simple_dai_props *dai_props; int num = rtd->num; dai_link = simple_priv_to_link(priv, num); @@ -75,7 +77,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - return asoc_simple_card_init_dai(dai, dai_props); + return asoc_simple_card_init_dai(dai, &dai_props->dai); } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -95,7 +97,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct snd_soc_card *card = simple_priv_to_card(priv); int ret; @@ -116,7 +118,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -141,7 +143,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -157,7 +159,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, PREFIX "prefix"); } - ret = asoc_simple_card_of_parse_tdm(np, dai_props); + ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); if (ret) return ret; @@ -230,7 +232,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct simple_dai_props *dai_props; struct snd_soc_card *card; struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; -- cgit v1.2.3-58-ga151 From 2289cc1c78574653d30b80696327646ba340babf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:08:51 +0000 Subject: ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-scu-card for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 09be02e7416f..91efc8653746 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -24,6 +24,7 @@ struct simple_card_data { struct snd_soc_codec_conf codec_conf; struct simple_dai_props { struct asoc_simple_dai dai; + struct snd_soc_dai_link_component codecs; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -103,11 +104,13 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (is_fe) { int is_single_links = 0; + struct snd_soc_dai_link_component *codecs; /* BE is dummy */ - dai_link->codec_of_node = NULL; - dai_link->codec_dai_name = "snd-soc-dummy-dai"; - dai_link->codec_name = "snd-soc-dummy"; + codecs = dai_link->codecs; + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; /* FE settings */ dai_link->dynamic = 1; @@ -149,13 +152,13 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "be.%s", - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) return ret; snd_soc_of_parse_audio_prefix(card, &priv->codec_conf, - dai_link->codec_of_node, + dai_link->codecs->of_node, PREFIX "prefix"); } @@ -236,7 +239,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_card *card; struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; - int num, ret; + int num, ret, i; /* Allocate the private data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -250,6 +253,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < num; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; -- cgit v1.2.3-58-ga151 From 8e6746db2e66c30e59a49ac15eb0c54d51acfb4b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:09:05 +0000 Subject: ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-card for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2094d2c8919f..5b2ecf8c2652 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -25,6 +25,7 @@ struct graph_card_data { struct graph_dai_props { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -213,7 +214,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; @@ -299,7 +300,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_dai_props *dai_props; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret; + int num, ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -315,6 +316,17 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < num; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + } + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); if (IS_ERR(priv->pa_gpio)) { ret = PTR_ERR(priv->pa_gpio); -- cgit v1.2.3-58-ga151 From 1340739d4de4d9d99f1134180f95b42cc4eda438 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:09:20 +0000 Subject: ASoC: audio-graph-scu-card: use simple_dai_props audi-graph-card and audio-graph-scu-card are very similar driver, but using different feature. Thus we are keeping synchronization on these 2 drivers style, because it is easy to confirm / check. Current big difference between these 2 drivers are "dai_props" on graph_card_data (= priv). It will be difficult to keep synchronize if we will add new feature on audio-graph-scu-card. Thus, this patch synchronize it. [audio-graph] struct graph_card_data { ... struct graph_dai_props { ... } *dai_props; ... }; [audio-graph-scu] struct graph_card_data { ... struct asoc_simple_dai *dai_props; ... }; Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 92882e392d6c..043938fece42 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -25,7 +25,9 @@ struct graph_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct graph_dai_props { + struct asoc_simple_dai dai; + } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; }; @@ -39,18 +41,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - return asoc_simple_card_clk_enable(dai_props); + return asoc_simple_card_clk_enable(&dai_props->dai); } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(&dai_props->dai); } static const struct snd_soc_ops asoc_graph_card_ops = { @@ -63,7 +65,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct graph_dai_props *dai_props; int num = rtd->num; dai_link = graph_priv_to_link(priv, num); @@ -72,7 +74,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - return asoc_simple_card_init_dai(dai, dai_props); + return asoc_simple_card_init_dai(dai, &dai_props->dai); } static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -92,7 +94,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); struct snd_soc_card *card = graph_priv_to_card(priv); int ret; @@ -110,7 +112,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -137,7 +139,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -153,7 +155,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, "prefix"); } - ret = asoc_simple_card_of_parse_tdm(ep, dai_props); + ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai); if (ret) return ret; @@ -331,7 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct graph_dai_props *dai_props; struct device *dev = &pdev->dev; struct snd_soc_card *card; int num, ret; -- cgit v1.2.3-58-ga151 From 04f7267aa8d17773917580951c740496e8059cba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:09:33 +0000 Subject: ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-scu-card for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 043938fece42..eeb3c1975fe3 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -27,6 +27,7 @@ struct graph_card_data { struct snd_soc_codec_conf codec_conf; struct graph_dai_props { struct asoc_simple_dai dai; + struct snd_soc_dai_link_component codecs; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -99,10 +100,13 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, int ret; if (is_fe) { + struct snd_soc_dai_link_component *codecs; + /* BE is dummy */ - dai_link->codec_of_node = NULL; - dai_link->codec_dai_name = "snd-soc-dummy-dai"; - dai_link->codec_name = "snd-soc-dummy"; + codecs = dai_link->codecs; + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; /* FE settings */ dai_link->dynamic = 1; @@ -145,13 +149,13 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "be.%s", - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) return ret; snd_soc_of_parse_audio_prefix(card, &priv->codec_conf, - dai_link->codec_of_node, + dai_link->codecs->of_node, "prefix"); } @@ -336,7 +340,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_dai_props *dai_props; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret; + int num, ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -352,6 +356,17 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < num; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; -- cgit v1.2.3-58-ga151 From 2967e5ea19ec041104fd43da4cb07e8e76a39e55 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:09:47 +0000 Subject: ASoC: simple-card-util: remove dai_link compatible code for codec Now no simple/audio cards are using legacy dai_link style for codec. Let's remove compatible code. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 73c0a904f32e..e7057be957e1 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -409,9 +409,7 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) num_links < card->num_links; num_links++, dai_link++) { of_node_put(dai_link->cpu_of_node); - of_node_put(dai_link->codec_of_node); - if (dai_link->codecs) - of_node_put(dai_link->codecs->of_node); + of_node_put(dai_link->codecs->of_node); } return 0; } -- cgit v1.2.3-58-ga151 From daecf46ee0e5f0fb2349e20af53c4653e2afc440 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:10:08 +0000 Subject: ASoC: soc-core: use snd_soc_dai_link_component for platform Current struct snd_soc_dai_link is supporting multicodec, and it is supporting legacy style of codec_name codec_of_node code_dai_name This is handled as single entry of multicodec. We don't have multicpu support yet, but in the future we will. In such case, we can use snd_soc_dai_link_component for both cpu/codec. Then the code will be more simple and readble. As next step, we want to use it for platform, too. This patch adds snd_soc_dai_link_component style for platform. We might have multiplatform support in the future, but we don't know yet. To avoid un-known issue / complex code, this patch supports just single-platform as 1st step. If we could use snd_soc_dai_link_component for all CPU/Codec/Platform, we will switch to new style, and remove legacy code. This is prepare for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 48 +++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 43 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 41cec42fb456..96c19aabf21b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -915,6 +915,8 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; + struct snd_soc_dai_link_component *platform; + int id; /* optional ID for machine driver link identification */ const struct snd_soc_pcm_stream *params; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 473eefe8658e..2a73630d0680 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -892,8 +892,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->codec_dai = codec_dais[0]; /* if there's no platform we match on the empty platform */ - platform_name = dai_link->platform_name; - if (!platform_name && !dai_link->platform_of_node) + platform_name = dai_link->platform->name; + if (!platform_name && !dai_link->platform->of_node) platform_name = "snd-soc-dummy"; /* find one from the set of registered platforms */ @@ -902,8 +902,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, if (!platform_of_node && component->dev->parent->of_node) platform_of_node = component->dev->parent->of_node; - if (dai_link->platform_of_node) { - if (platform_of_node != dai_link->platform_of_node) + if (dai_link->platform->of_node) { + if (platform_of_node != dai_link->platform->of_node) continue; } else { if (strcmp(component->name, platform_name)) @@ -1015,6 +1015,31 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_platform(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* + * FIXME + * + * this function should be removed in the future + */ + /* convert Legacy platform link */ + if (dai_link->platform) + return 0; + + dai_link->platform = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->platform) + return -ENOMEM; + + dai_link->platform->name = dai_link->platform_name; + dai_link->platform->of_node = dai_link->platform_of_node; + dai_link->platform->dai_name = NULL; + + return 0; +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -1047,6 +1072,12 @@ static int soc_init_dai_link(struct snd_soc_card *card, { int i, ret; + ret = snd_soc_init_platform(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multiplatform\n"); + return ret; + } + ret = snd_soc_init_multicodec(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init multicodec\n"); @@ -1076,13 +1107,12 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform_name && link->platform_of_node) { + if (link->platform->name && link->platform->of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); return -EINVAL; } - /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1917,7 +1947,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) card->dai_link[i].name); /* override platform component */ - dai_link->platform_name = component->name; + if (snd_soc_init_platform(card, dai_link) < 0) { + dev_err(card->dev, "init platform error"); + continue; + } + dai_link->platform->name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- cgit v1.2.3-58-ga151 From 868cdb4690699b04ca4d09b1e0178dfc680dbd8e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:10:20 +0000 Subject: ASoC: simple-card-util: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-card-util for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/simple-card-utils.c | 11 ++++++++--- 2 files changed, 9 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 3b5bd6e76f88..fb0318f9b10f 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); &dai_link->codec_dai_name, \ list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, NULL, \ + asoc_simple_card_parse_dai(node, dai_link->platform, \ &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index e7057be957e1..644cd62ba027 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -376,10 +376,15 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform_of_node) - dai_link->platform_of_node = dai_link->cpu_of_node; - + if (dai_link->platform) { + if (!dai_link->platform->of_node) + dai_link->platform->of_node = dai_link->cpu_of_node; + } else { + if (!dai_link->platform_of_node) + dai_link->platform_of_node = dai_link->cpu_of_node; + } return 0; + } EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); -- cgit v1.2.3-58-ga151 From e58f41e41185c6906bd11c73c4e76aa5fc3ea685 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:10:33 +0000 Subject: ASoC: simple-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 67a56f32f983..5a3f59aa4ba5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -21,6 +21,7 @@ struct simple_card_data { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platform; unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -391,6 +392,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->dai_props = dai_props; @@ -416,6 +418,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } else { struct asoc_simple_card_info *cinfo; struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *platform; cinfo = dev->platform_data; if (!cinfo) { @@ -436,10 +439,12 @@ static int asoc_simple_card_probe(struct platform_device *pdev) codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; + platform = dai_link->platform; + platform->name = cinfo->platform; + card->name = (cinfo->card) ? cinfo->card : cinfo->name; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; - dai_link->platform_name = cinfo->platform; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; -- cgit v1.2.3-58-ga151 From 24f3bead9b72cbd91956be28ab6a74d51bdd4c6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:10:46 +0000 Subject: ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-scu-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 91efc8653746..85b46f0eae0f 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -25,6 +25,7 @@ struct simple_card_data { struct simple_dai_props { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -262,6 +263,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->dai_props = dai_props; -- cgit v1.2.3-58-ga151 From 46c73187f2986e40b427485bb8f4401aa9143ed0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:10:58 +0000 Subject: ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5b2ecf8c2652..fb6635f8d5d7 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -26,6 +26,7 @@ struct graph_card_data { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platform; unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -325,6 +326,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); -- cgit v1.2.3-58-ga151 From 77b9b84132f0b9ca0802a25277eb7be49713661f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:11:12 +0000 Subject: ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-scu-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index eeb3c1975fe3..b83bb31021a9 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -28,6 +28,7 @@ struct graph_card_data { struct graph_dai_props { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -365,6 +366,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->dai_props = dai_props; -- cgit v1.2.3-58-ga151 From c2f0898b86486a459fa8c91d194f3b699498c0c1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 31 Aug 2018 03:11:25 +0000 Subject: ASoC: simple-card-util: remove dai_link compatible code for platform Now no simple/audio cards are using legacy dai_link style for platform. Let's remove compatible code. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 644cd62ba027..b400dbf1f834 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -376,13 +376,9 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (dai_link->platform) { - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; - } else { - if (!dai_link->platform_of_node) - dai_link->platform_of_node = dai_link->cpu_of_node; - } + if (!dai_link->platform->of_node) + dai_link->platform->of_node = dai_link->cpu_of_node; + return 0; } -- cgit v1.2.3-58-ga151 From 919869214b8e0b24926a278e121879f60df485bb Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Fri, 31 Aug 2018 10:14:06 -0500 Subject: ASoC: tas6424: Print full register name in error message The current short version of the register name may be ambiguous when another fault register detection is added. Use the full name. While here fix comment about clearing faults, the CLEAR_FAULT register actually only clears sticky bits, which are only warnings, fault bits can only cleared by resolving the fault. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tas6424.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 14999b999fd3..3315ce8a15e6 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -408,7 +408,7 @@ static void tas6424_fault_check_work(struct work_struct *work) ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, ®); if (ret < 0) { - dev_err(dev, "failed to read FAULT1 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret); goto out; } @@ -451,7 +451,7 @@ static void tas6424_fault_check_work(struct work_struct *work) check_global_fault2_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, ®); if (ret < 0) { - dev_err(dev, "failed to read FAULT2 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret); goto out; } @@ -524,7 +524,7 @@ check_warn_reg: /* Store current warn value so we can detect any changes next time */ tas6424->last_warn = reg; - /* Clear any faults by toggling the CLEAR_FAULT control bit */ + /* Clear any warnings by toggling the CLEAR_FAULT control bit */ ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3, TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT); if (ret < 0) -- cgit v1.2.3-58-ga151 From 5fb6589acc3860304436aa436e7ea33712de6fc2 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Fri, 31 Aug 2018 10:14:07 -0500 Subject: ASoC: tas6424: Add channel fault reporting The TAS6426 has a register that reports channel faults such as overcurrent and continuous DC output. Add reporting of this here. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tas6424.c | 52 ++++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/tas6424.h | 10 +++++++++ 2 files changed, 56 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aac559fffc1a..36aebdb8f55c 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -41,6 +41,7 @@ struct tas6424_data { struct regmap *regmap; struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES]; struct delayed_work fault_check_work; + unsigned int last_cfault; unsigned int last_fault1; unsigned int last_fault2; unsigned int last_warn; @@ -406,6 +407,51 @@ static void tas6424_fault_check_work(struct work_struct *work) unsigned int reg; int ret; + ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, ®); + if (ret < 0) { + dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret); + goto out; + } + + if (!reg) { + tas6424->last_cfault = reg; + goto check_global_fault1_reg; + } + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS6424 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1)) + dev_crit(dev, "experienced a channel 1 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2)) + dev_crit(dev, "experienced a channel 2 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3)) + dev_crit(dev, "experienced a channel 3 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4)) + dev_crit(dev, "experienced a channel 4 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1)) + dev_crit(dev, "experienced a channel 1 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2)) + dev_crit(dev, "experienced a channel 2 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3)) + dev_crit(dev, "experienced a channel 3 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4)) + dev_crit(dev, "experienced a channel 4 DC fault\n"); + + /* Store current fault1 value so we can detect any changes next time */ + tas6424->last_cfault = reg; + +check_global_fault1_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, ®); if (ret < 0) { dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret); @@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work) goto check_global_fault2_reg; } - /* - * Only flag errors once for a given occurrence. This is needed as - * the TAS6424 will take time clearing the fault condition internally - * during which we don't want to bombard the system with the same - * error message over and over. - */ if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV)) dev_crit(dev, "experienced a PVDD overvoltage fault\n"); diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index b5958c45ed0e..c67a7835ca66 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -115,6 +115,16 @@ #define TAS6424_LDGBYPASS_SHIFT 0 #define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT) +/* TAS6424_GLOB_FAULT1_REG */ +#define TAS6424_FAULT_OC_CH1 BIT(7) +#define TAS6424_FAULT_OC_CH2 BIT(6) +#define TAS6424_FAULT_OC_CH3 BIT(5) +#define TAS6424_FAULT_OC_CH4 BIT(4) +#define TAS6424_FAULT_DC_CH1 BIT(3) +#define TAS6424_FAULT_DC_CH2 BIT(2) +#define TAS6424_FAULT_DC_CH3 BIT(1) +#define TAS6424_FAULT_DC_CH4 BIT(0) + /* TAS6424_GLOB_FAULT1_REG */ #define TAS6424_FAULT_CLOCK BIT(4) #define TAS6424_FAULT_PVDD_OV BIT(3) -- cgit v1.2.3-58-ga151 From 63a886f38dd96868e33488eccee8ed427144d397 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sun, 2 Sep 2018 19:38:10 -0700 Subject: ASoC: fix soc-core.c kernel-doc warning Fix kernel-doc warning: ../sound/soc/soc-core.c:2918: warning: Excess function parameter 'legacy_dai_naming' description in 'snd_soc_register_dais' Signed-off-by: Randy Dunlap Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2a73630d0680..e9c2304afaf1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2944,8 +2944,6 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs - * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the - * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, size_t count) -- cgit v1.2.3-58-ga151 From 80863ee222d37b1797cea74d2257ad6d68444d30 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Fri, 31 Aug 2018 13:24:31 -0500 Subject: ASoC: tlv320aic31xx: Add short circuit detection support These devices support detecting and reporting short circuits across the output stages. Add support for reporting these issue. Do this by registering an interrupt if available and enabling this error to trigger that interrupt in the device. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 55 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 16 ++++++++++++ 2 files changed, 71 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bf92d36b8f8a..2abe51d9f879 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -167,6 +167,7 @@ struct aic31xx_priv { u8 p_div; int rate_div_line; bool master_dapm_route_applied; + int irq; }; struct aic31xx_rate_divs { @@ -1391,6 +1392,40 @@ static const struct acpi_device_id aic31xx_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); #endif +static irqreturn_t aic31xx_irq(int irq, void *data) +{ + struct aic31xx_priv *aic31xx = data; + struct device *dev = aic31xx->dev; + unsigned int value; + bool handled = false; + int ret; + + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_HPLSCDETECT) + dev_err(dev, "Short circuit on Left output is detected\n"); + if (value & AIC31XX_HPRSCDETECT) + dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & ~(AIC31XX_HPLSCDETECT | + AIC31XX_HPRSCDETECT)) + dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); + +exit: + if (handled) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1413,6 +1448,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } aic31xx->dev = &i2c->dev; + aic31xx->irq = i2c->irq; aic31xx->codec_type = id->driver_data; @@ -1456,6 +1492,25 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + if (aic31xx->irq > 0) { + regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, + AIC31XX_GPIO1_FUNC_MASK, + AIC31XX_GPIO1_INT1 << + AIC31XX_GPIO1_FUNC_SHIFT); + + regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_SC); + + ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, + NULL, aic31xx_irq, + IRQF_ONESHOT, "aic31xx-irq", + aic31xx); + if (ret) { + dev_err(aic31xx->dev, "Unable to request IRQ\n"); + return ret; + } + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..52e171988906 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -191,6 +191,22 @@ struct aic31xx_pdata { #define AIC31XX_SC BIT(3) #define AIC31XX_ENGINE BIT(2) +/* AIC31XX_GPIO1 */ +#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2) +#define AIC31XX_GPIO1_FUNC_SHIFT 2 +#define AIC31XX_GPIO1_DISABLED 0x00 +#define AIC31XX_GPIO1_INPUT 0x01 +#define AIC31XX_GPIO1_GPI 0x02 +#define AIC31XX_GPIO1_GPO 0x03 +#define AIC31XX_GPIO1_CLKOUT 0x04 +#define AIC31XX_GPIO1_INT1 0x05 +#define AIC31XX_GPIO1_INT2 0x06 +#define AIC31XX_GPIO1_ADC_WCLK 0x07 +#define AIC31XX_GPIO1_SBCLK 0x08 +#define AIC31XX_GPIO1_SWCLK 0x09 +#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10 +#define AIC31XX_GPIO1_SDOUT 0x11 + /* AIC31XX_DACSETUP */ #define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) -- cgit v1.2.3-58-ga151 From b5c088689847372794500f83b65673aaa8ca4d8d Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:05:11 +0000 Subject: ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime() Add warning message to rsnd_kctrl_accept_runtime(), when kctrl update is rejected due to corresponding dai-link is idle. So that user can notice the reason of kctrl update failure. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [kuninori: adjust to upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f8425d8b44d2..ab7e317bbed8 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1263,8 +1263,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + + if (!runtime) { + dev_warn(dev, "Can't update kctrl when idle\n"); + return 0; + } - return !!runtime; + return 1; } struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) -- cgit v1.2.3-58-ga151 From fb2815f44a9eb341ed8990263855a266960a5135 Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 3 Sep 2018 07:05:42 +0000 Subject: ASoC: rsnd: add support for 16/24 bit slot widths The slot width (system word length) was fixed at 32 bits. This patch allows also setting it to 16 or 24 bits. Signed-off-by: Dragos Tarcatu Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 32 +++++++++++++++++++++------- sound/soc/sh/rcar/rsnd.h | 8 ++++++- sound/soc/sh/rcar/ssi.c | 54 +++++++++++++++++++++++++++++++++++------------- 3 files changed, 72 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index ab7e317bbed8..ce0a3a61c441 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -540,6 +540,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, return rdai->ssi_lane; } +int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width) +{ + if (width > 0) + rdai->chan_width = width; + + return rdai->chan_width; +} + struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) { if ((id < 0) || (id >= rsnd_rdai_nr(priv))) @@ -720,6 +728,16 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct device *dev = rsnd_priv_to_dev(priv); + switch (slot_width) { + case 16: + case 24: + case 32: + break; + default: + dev_err(dev, "unsupported slot width value: %d\n", slot_width); + return -EINVAL; + } + switch (slots) { case 2: case 6: @@ -727,6 +745,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, /* TDM Extend Mode */ rsnd_rdai_channels_set(rdai, slots); rsnd_rdai_ssi_lane_set(rdai, 1); + rsnd_rdai_width_set(rdai, slot_width); break; default: dev_err(dev, "unsupported TDM slots (%d)\n", slots); @@ -755,7 +774,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = { 192000, }; -static int rsnd_soc_hw_rule(struct rsnd_priv *priv, +static int rsnd_soc_hw_rule(struct rsnd_dai *rdai, unsigned int *list, int list_num, struct snd_interval *baseline, struct snd_interval *iv) { @@ -772,14 +791,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, if (!snd_interval_test(iv, list[i])) continue; - rate = rsnd_ssi_clk_query(priv, + rate = rsnd_ssi_clk_query(rdai, baseline->min, list[i], NULL); if (rate > 0) { p.min = min(p.min, list[i]); p.max = max(p.max, list[i]); } - rate = rsnd_ssi_clk_query(priv, + rate = rsnd_ssi_clk_query(rdai, baseline->max, list[i], NULL); if (rate > 0) { p.min = min(p.min, list[i]); @@ -799,7 +818,6 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_interval ic; struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* @@ -811,7 +829,7 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, ic.min = ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); - return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list, + return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list, ARRAY_SIZE(rsnd_soc_hw_rate_list), &ic, ir); } @@ -837,7 +855,6 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_interval ic; struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* @@ -849,7 +866,7 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ic.min = ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); - return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list, + return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list, ARRAY_SIZE(rsnd_soc_hw_channels_list), ir, &ic); } @@ -1072,6 +1089,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, rdai->capture.rdai = rdai; rsnd_rdai_channels_set(rdai, 2); /* default 2ch */ rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */ + rsnd_rdai_width_set(rdai, 32); /* default 32bit width */ for (io_i = 0;; io_i++) { playback = of_parse_phandle(dai_np, "playback", io_i); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 96d93330b1e1..698b08155b06 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,6 +460,7 @@ struct rsnd_dai { int max_channels; /* 2ch - 16ch */ int ssi_lane; /* 1lane - 4lane */ + int chan_width; /* 16/24/32 bit width */ unsigned int clk_master:1; unsigned int bit_clk_inv:1; @@ -493,6 +494,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai, int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, int ssi_lane); +#define rsnd_rdai_width_set(rdai, width) \ + rsnd_rdai_width_ctrl(rdai, width) +#define rsnd_rdai_width_get(rdai) \ + rsnd_rdai_width_ctrl(rdai, 0) +int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, @@ -702,7 +708,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, +unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx); /* diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8304e4ec9242..f707f53748bd 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -42,7 +42,13 @@ #define DWL_24 (5 << 19) /* Data Word Length */ #define DWL_32 (6 << 19) /* Data Word Length */ +/* + * System word length + */ +#define SWL_16 (1 << 16) /* R/W System Word Length */ +#define SWL_24 (2 << 16) /* R/W System Word Length */ #define SWL_32 (3 << 16) /* R/W System Word Length */ + #define SCKD (1 << 15) /* Serial Bit Clock Direction */ #define SWSD (1 << 14) /* Serial WS Direction */ #define SCKP (1 << 13) /* Serial Bit Clock Polarity */ @@ -220,14 +226,32 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io) return 0; } -unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, +static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai) +{ + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); + int width = rsnd_rdai_width_get(rdai); + + switch (width) { + case 32: return SWL_32; + case 24: return SWL_24; + case 16: return SWL_16; + } + + dev_err(dev, "unsupported slot width value: %d\n", width); + return 0; +} + +unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx) { + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; int j, ret; unsigned int main_rate; + int width = rsnd_rdai_width_get(rdai); for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { @@ -240,12 +264,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, if (j == 0) continue; - /* - * this driver is assuming that - * system word is 32bit x chan - * see rsnd_ssi_init() - */ - main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j]; + main_rate = width * param1 * param2 * ssi_clk_mul_table[j]; ret = rsnd_adg_clk_query(priv, main_rate); if (ret < 0) @@ -292,7 +311,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; } - main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); return -EIO; @@ -312,7 +331,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, * SSICR : FORCE, SCKD, SWSD * SSIWSR : CONT */ - ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx); + ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) | + SCKD | SWSD | CKDV(idx); ssi->wsr = CONT; ssi->rate = rate; @@ -357,11 +377,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, is_tdm = rsnd_runtime_is_ssi_tdm(io); - /* - * always use 32bit system word. - * see also rsnd_ssi_master_clk_enable() - */ - cr_own |= FORCE | SWL_32; + cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai); if (rdai->bit_clk_inv) cr_own |= SCKP; @@ -494,7 +510,17 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, struct snd_pcm_hw_params *params) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); int chan = params_channels(params); + unsigned int fmt_width = snd_pcm_format_width(params_format(params)); + + if (fmt_width > rdai->chan_width) { + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "invalid combination of slot-width and format-data-width\n"); + return -EINVAL; + } /* * snd_pcm_ops::hw_params will be called *before* -- cgit v1.2.3-58-ga151 From 3791b3ee4bb13c381868da89e9e6deb11de660ad Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 3 Sep 2018 07:06:01 +0000 Subject: ASoC: rsnd: add support for the DSP_A/DSP_B formats Signed-off-by: Dragos Tarcatu Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index ce0a3a61c441..8a417768b7ef 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -689,6 +689,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) rdai->frm_clk_inv = 0; break; case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_B: rdai->sys_delay = 1; rdai->data_alignment = 0; rdai->frm_clk_inv = 1; @@ -698,6 +699,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) rdai->data_alignment = 1; rdai->frm_clk_inv = 1; break; + case SND_SOC_DAIFMT_DSP_A: + rdai->sys_delay = 0; + rdai->data_alignment = 0; + rdai->frm_clk_inv = 1; + break; } /* set clock inversion */ -- cgit v1.2.3-58-ga151 From ba5d553b7bd71e63d639863e2cb09e0c9543b8b7 Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 3 Sep 2018 07:06:29 +0000 Subject: ASoC: rsnd: add support for 8 bit S8 format This patch adds support for SNDRV_PCM_FMTBIT_S8 format. Signed-off-by: Dragos Tarcatu Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 10 +++++++--- sound/soc/sh/rcar/ssi.c | 3 +++ 2 files changed, 10 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8a417768b7ef..cd0ff1eef463 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -102,7 +102,9 @@ #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_192000 -#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) static const struct of_device_id rsnd_of_match[] = { { .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 }, @@ -280,6 +282,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) struct device *dev = rsnd_priv_to_dev(priv); switch (snd_pcm_format_width(runtime->format)) { + case 8: + return 16 << 16; case 16: return 8 << 16; case 24: @@ -331,7 +335,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) target = cmd ? cmd : ssiu; } - /* Non target mod or 24bit data needs normal DALIGN */ + /* Non target mod or non 16bit needs normal DALIGN */ if ((snd_pcm_format_width(runtime->format) != 16) || (mod != target)) return 0x76543210; @@ -367,7 +371,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) * HW 24bit data is located as 0x******00 * */ - if (snd_pcm_format_width(runtime->format) == 16) + if (snd_pcm_format_width(runtime->format) != 24) return 0; for (i = 0; i < ARRAY_SIZE(playback_mods); i++) { diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f707f53748bd..765ecc06c7c9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -400,6 +400,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own &= ~DWL_MASK; switch (snd_pcm_format_width(runtime->format)) { + case 8: + cr_own |= DWL_8; + break; case 16: cr_own |= DWL_16; break; -- cgit v1.2.3-58-ga151 From b735662fa473c0e3618a4d645ce797d31e0c9192 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:06:50 +0000 Subject: ASoC: rsnd: remove is_play parameter from hw_rule function Currently rsnd_dai_stream *io is set to either &rdai->playback or &rdai->capture based on whether it is a playback or capture stream, in __rsnd_soc_hw_rule_* functions, but this is not necessary, rsnd_dai_stream *io handler can be get from rule->private. This patch removes 'is_play' parameter from hw_rule function. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 54 +++++++++++------------------------------------- 1 file changed, 12 insertions(+), 42 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index cd0ff1eef463..c66b3dade947 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -819,16 +819,14 @@ static int rsnd_soc_hw_rule(struct rsnd_dai *rdai, return snd_interval_refine(iv, &p); } -static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule, - int is_play) +static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval ic; - struct snd_soc_dai *dai = rule->private; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + struct rsnd_dai_stream *io = rule->private; + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); /* * possible sampling rate limitation is same as @@ -844,28 +842,14 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, &ic, ir); } -static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_rate(params, rule, 1); -} - -static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_rate(params, rule, 0); -} - -static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule, - int is_play) +static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval ic; - struct snd_soc_dai *dai = rule->private; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + struct rsnd_dai_stream *io = rule->private; + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); /* * possible sampling rate limitation is same as @@ -881,18 +865,6 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ir, &ic); } -static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_channels(params, rule, 1); -} - -static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_channels(params, rule, 0); -} - static const struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | @@ -949,14 +921,12 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - is_play ? rsnd_soc_hw_rule_rate_playback : - rsnd_soc_hw_rule_rate_capture, - dai, + rsnd_soc_hw_rule_rate, + is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - is_play ? rsnd_soc_hw_rule_channels_playback : - rsnd_soc_hw_rule_channels_capture, - dai, + rsnd_soc_hw_rule_channels, + is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_RATE, -1); } -- cgit v1.2.3-58-ga151 From 0e289012b47a2de1f029a6b61c75998e2f159dd9 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:07:07 +0000 Subject: ASoC: rsnd: ssi: Fix issue in dma data address assignment Same SSI device may be used in different dai links, by only having one dma struct in rsnd_ssi, after the first instance's dma config be initilized, the following instances can no longer configure dma, this causes issue, when their dma data address are different from the first instance. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 4 +--- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 698b08155b06..20e6a2ebebed 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -431,6 +431,7 @@ struct rsnd_dai_stream { char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; + struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ u32 parent_ssi_status; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 765ecc06c7c9..89cc433e2fc9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -78,7 +78,6 @@ struct rsnd_ssi { struct rsnd_mod mod; - struct rsnd_mod *dma; u32 flags; u32 cr_own; @@ -899,7 +898,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); int ret; /* @@ -914,7 +912,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, return ret; /* SSI probe might be called many times in MUX multi path */ - ret = rsnd_dma_attach(io, mod, &ssi->dma); + ret = rsnd_dma_attach(io, mod, &io->dma); return ret; } -- cgit v1.2.3-58-ga151 From 599da084e041b877ef89211dcbb4c7bd8380049d Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:07:26 +0000 Subject: ASoC: rsnd: ssi: Check runtime channel number rather than hw_params The number of channel handled by SSI maybe differs from the one set in hw_params, currently SSI checks hw_params's channel number, and constrains to use same channel number, when it is being used by multiple clients. This patch corrects to check runtime channel number rather than channel number set in hw_params. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 27 +++++++-------------------- 1 file changed, 7 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 89cc433e2fc9..3f6dd9f07bc6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -307,6 +307,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return -EINVAL; } + if (ssi->chan != chan) { + dev_err(dev, "SSI parent/child should use same chan\n"); + return -EINVAL; + } + return 0; } @@ -334,6 +339,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, SCKD | SWSD | CKDV(idx); ssi->wsr = CONT; ssi->rate = rate; + ssi->chan = chan; dev_dbg(dev, "%s[%d] outputs %u Hz\n", rsnd_mod_name(mod), @@ -359,6 +365,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod, ssi->cr_clk = 0; ssi->rate = 0; + ssi->chan = 0; rsnd_adg_ssi_clk_stop(mod); } @@ -511,9 +518,7 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dai *rdai = rsnd_io_to_rdai(io); - int chan = params_channels(params); unsigned int fmt_width = snd_pcm_format_width(params_format(params)); if (fmt_width > rdai->chan_width) { @@ -524,24 +529,6 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, return -EINVAL; } - /* - * snd_pcm_ops::hw_params will be called *before* - * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0 - * in 1st call. - */ - if (ssi->usrcnt) { - /* - * Already working. - * It will happen if SSI has parent/child connection. - * it is error if child <-> parent SSI uses - * different channels. - */ - if (ssi->chan != chan) - return -EIO; - } - - ssi->chan = chan; - return 0; } -- cgit v1.2.3-58-ga151 From 5e45a6fab3b90ca300e13191fc68baaa8e37d1d4 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:07:43 +0000 Subject: ASoc: rsnd: dma: Calculate dma address with consider of BUSIF DMA address calculated by rsnd_dma_addr() only considers BUSIF0 so far. But BUSIF1 ~ BUSIF7 also maybe used, in the future. This patch updates DMA address calculations, to also consider BUSIF number used by SSI. One note is that we can't support SSI9-4/5/6/7 so far, because its address is out of calculation rule. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 43 +++++++++++++++++++++++++++---------------- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 5 +++++ 3 files changed, 33 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fe63ef8600d0..73f743ea3d08 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -487,11 +487,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) -#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) #define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) @@ -517,6 +517,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, !!rsnd_io_to_mod_mix(io) || !!rsnd_io_to_mod_ctu(io); int id = rsnd_mod_id(mod); + int busif = rsnd_ssi_get_busif(io); struct dma_addr { dma_addr_t out_addr; dma_addr_t in_addr; @@ -533,25 +534,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, }, /* SSI */ /* Capture */ - {{{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 } }, /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) } } }, /* SSIU */ /* Capture */ - {{{ RDMA_SSIU_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, + {{{ RDMA_SSIU_O_N(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 } }, /* Playback */ - {{ 0, RDMA_SSIU_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } }, + {{ 0, RDMA_SSIU_I_N(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) } } }, }; + /* + * FIXME + * + * We can't support SSI9-4/5/6/7, because its address is + * out of calculation rule + */ + if ((id == 9) && (busif >= 4)) + dev_err(dev, "This driver doesn't support SSI%d-%d, so far", + id, busif); + /* it shouldn't happen */ if (use_cmd && !use_src) dev_err(dev, "DVC is selected without SRC\n"); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 20e6a2ebebed..cb27a679dc65 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -692,6 +692,7 @@ void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); +int rsnd_ssi_get_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); #define RSND_SSI_HDMI_PORT0 0xf0 diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3f6dd9f07bc6..85da4fc82011 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -150,6 +150,11 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) return use_busif; } +int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) +{ + return 0; /* BUSIF0 only for now */ +} + static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); -- cgit v1.2.3-58-ga151 From 92c7d384ff7282738c70720f8670c9b65c90c7df Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:08:00 +0000 Subject: ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF PDMACHCR setting for SSI only considers BUSIF0 so far. But BUSIF1 ~ BUSIF7 also maybe used, in the future. This patch updates table gen2_id_table_ssiu, to also consider BUSIF number used by SSI. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 37 ++++++++++++++++++++++++++----------- 1 file changed, 26 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 73f743ea3d08..d3b1a4ae876a 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -298,16 +298,26 @@ static struct rsnd_mod_ops rsnd_dmaen_ops = { * Audio DMAC peri peri */ static const u8 gen2_id_table_ssiu[] = { - 0x00, /* SSI00 */ - 0x04, /* SSI10 */ - 0x08, /* SSI20 */ - 0x0c, /* SSI3 */ - 0x0d, /* SSI4 */ - 0x0e, /* SSI5 */ - 0x0f, /* SSI6 */ - 0x10, /* SSI7 */ - 0x11, /* SSI8 */ - 0x12, /* SSI90 */ + /* SSI00 ~ SSI07 */ + 0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c, + /* SSI10 ~ SSI17 */ + 0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40, + /* SSI20 ~ SSI27 */ + 0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44, + /* SSI30 ~ SSI37 */ + 0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, + /* SSI40 ~ SSI47 */ + 0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52, + /* SSI5 */ + 0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI6 */ + 0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI7 */ + 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI8 */ + 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI90 ~ SSI97 */ + 0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56, }; static const u8 gen2_id_table_scu[] = { 0x2d, /* SCU_SRCI0 */ @@ -333,18 +343,23 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); const u8 *entry = NULL; - int id = rsnd_mod_id(mod); + int id = 255; int size = 0; if (mod == ssi) { + int busif = rsnd_ssi_get_busif(io); + entry = gen2_id_table_ssiu; size = ARRAY_SIZE(gen2_id_table_ssiu); + id = (rsnd_mod_id(mod) * 8) + busif; } else if (mod == src) { entry = gen2_id_table_scu; size = ARRAY_SIZE(gen2_id_table_scu); + id = rsnd_mod_id(mod); } else if (mod == dvc) { entry = gen2_id_table_cmd; size = ARRAY_SIZE(gen2_id_table_cmd); + id = rsnd_mod_id(mod); } if ((!entry) || (size <= id)) { -- cgit v1.2.3-58-ga151 From 8c9d750333408420a1e4816b1820f10be2a84af6 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:08:20 +0000 Subject: ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0 Currently only BUSIF0 is supported by SSIU, all register setting is done only for BUSIF. Since BUSIF1 ~ BUSIF7 has been supported, so also support these BUSIF from SSIU. One note is that we can't support SSI9-4/5/6/7 so far, because its address is out of calculation rule. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 27 ++++++++++++++++--- sound/soc/sh/rcar/rsnd.h | 27 ++++++++++++++++--- sound/soc/sh/rcar/ssiu.c | 70 ++++++++++++++++++++++++++++++++++++++++-------- 3 files changed, 107 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0230301fe078..3032869a7f26 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -219,9 +219,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), /* FIXME: it needs SSI_MODE2/3 in the future */ - RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80), - RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80), - RSND_GEN_M_REG(SSI_BUSIF_DALIGN,0x8, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index cb27a679dc65..7ff58b0e8002 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -156,9 +156,30 @@ enum rsnd_reg { RSND_REG_SSI_MODE2, RSND_REG_SSI_CONTROL, RSND_REG_SSI_CTRL, - RSND_REG_SSI_BUSIF_MODE, - RSND_REG_SSI_BUSIF_ADINR, - RSND_REG_SSI_BUSIF_DALIGN, + RSND_REG_SSI_BUSIF0_MODE, + RSND_REG_SSI_BUSIF0_ADINR, + RSND_REG_SSI_BUSIF0_DALIGN, + RSND_REG_SSI_BUSIF1_MODE, + RSND_REG_SSI_BUSIF1_ADINR, + RSND_REG_SSI_BUSIF1_DALIGN, + RSND_REG_SSI_BUSIF2_MODE, + RSND_REG_SSI_BUSIF2_ADINR, + RSND_REG_SSI_BUSIF2_DALIGN, + RSND_REG_SSI_BUSIF3_MODE, + RSND_REG_SSI_BUSIF3_ADINR, + RSND_REG_SSI_BUSIF3_DALIGN, + RSND_REG_SSI_BUSIF4_MODE, + RSND_REG_SSI_BUSIF4_ADINR, + RSND_REG_SSI_BUSIF4_DALIGN, + RSND_REG_SSI_BUSIF5_MODE, + RSND_REG_SSI_BUSIF5_ADINR, + RSND_REG_SSI_BUSIF5_DALIGN, + RSND_REG_SSI_BUSIF6_MODE, + RSND_REG_SSI_BUSIF6_ADINR, + RSND_REG_SSI_BUSIF6_DALIGN, + RSND_REG_SSI_BUSIF7_MODE, + RSND_REG_SSI_BUSIF7_ADINR, + RSND_REG_SSI_BUSIF7_DALIGN, RSND_REG_SSI_INT_ENABLE, RSND_REG_SSI_SYS_STATUS0, RSND_REG_SSI_SYS_STATUS1, diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 016fbf5ac242..a9605a0163f2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -140,15 +140,59 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, rsnd_mod_write(mod, SSI_MODE, mode); if (rsnd_ssi_use_busif(io)) { - rsnd_mod_write(mod, SSI_BUSIF_ADINR, - rsnd_get_adinr_bit(mod, io) | - (rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE, - rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN, - rsnd_get_dalign(mod, io)); + int id = rsnd_mod_id(mod); + int busif = rsnd_ssi_get_busif(io); + + /* + * FIXME + * + * We can't support SSI9-4/5/6/7, because its address is + * out of calculation rule + */ + if ((id == 9) && (busif >= 4)) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "This driver doesn't support SSI%d-%d, so far", + id, busif); + } + +#define RSND_WRITE_BUSIF(i) \ + rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR, \ + rsnd_get_adinr_bit(mod, io) | \ + (rsnd_io_is_play(io) ? \ + rsnd_runtime_channel_after_ctu(io) : \ + rsnd_runtime_channel_original(io))); \ + rsnd_mod_write(mod, SSI_BUSIF##i##_MODE, \ + rsnd_get_busif_shift(io, mod) | 1); \ + rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN, \ + rsnd_get_dalign(mod, io)) + + switch (busif) { + case 0: + RSND_WRITE_BUSIF(0); + break; + case 1: + RSND_WRITE_BUSIF(1); + break; + case 2: + RSND_WRITE_BUSIF(2); + break; + case 3: + RSND_WRITE_BUSIF(3); + break; + case 4: + RSND_WRITE_BUSIF(4); + break; + case 5: + RSND_WRITE_BUSIF(5); + break; + case 6: + RSND_WRITE_BUSIF(6); + break; + case 7: + RSND_WRITE_BUSIF(7); + break; + } } if (hdmi) { @@ -194,10 +238,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + int busif = rsnd_ssi_get_busif(io); + if (!rsnd_ssi_use_busif(io)) return 0; - rsnd_mod_write(mod, SSI_CTRL, 0x1); + rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4)); if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0x1); @@ -209,10 +255,12 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + int busif = rsnd_ssi_get_busif(io); + if (!rsnd_ssi_use_busif(io)) return 0; - rsnd_mod_write(mod, SSI_CTRL, 0); + rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0); if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0); -- cgit v1.2.3-58-ga151 From 2e66d523cd055ac3fa920f7e630c4bfa80d24c24 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Mon, 3 Sep 2018 07:08:37 +0000 Subject: ASoC: rsnd: ssiu: Support to init different BUSIF instance Currently ssiu's .init is only called once during audio stream. But SSIU with different BUSIF, shall be initialized each time, even they are used in the same audio stream. This patch introduces ssiu_status for BUSIF0 to BUSIF7 in rsnd_ssiu, to make sure same .init for different BUSIF can always be executed. To avoid the first stopped stream to stop the whole SSIU, which may still has other BUSIF instance running, use usrcnt to count the usage of SSIU, only the last user of SSIU can stop the whole SSIU. Signed-off-by: Jiada Wang Signed-off-by: Timo Wischer [Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssiu.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index a9605a0163f2..39b67643b5dc 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -10,9 +10,12 @@ struct rsnd_ssiu { struct rsnd_mod mod; + u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */ + unsigned int usrcnt; }; #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr) +#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod) #define for_each_rsnd_ssiu(pos, priv, i) \ for (i = 0; \ (i < rsnd_ssiu_nr(priv)) && \ @@ -120,6 +123,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); int hdmi = rsnd_ssi_hdmi_port(io); int ret; u32 mode = 0; @@ -128,6 +132,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (ret < 0) return ret; + ssiu->usrcnt++; + if (rsnd_runtime_is_ssi_tdm(io)) { /* * TDM Extend Mode @@ -255,6 +261,7 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); int busif = rsnd_ssi_get_busif(io); if (!rsnd_ssi_use_busif(io)) @@ -262,6 +269,9 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0); + if (--ssiu->usrcnt) + return 0; + if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0); @@ -294,6 +304,16 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io, return rsnd_dai_connect(mod, io, mod->type); } +static u32 *rsnd_ssiu_get_status(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, + enum rsnd_mod_type type) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + int busif = rsnd_ssi_get_busif(io); + + return &ssiu->busif_status[busif]; +} + int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); @@ -317,7 +337,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) for_each_rsnd_ssiu(ssiu, priv, i) { ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu), - ops, NULL, rsnd_mod_get_status, + ops, NULL, rsnd_ssiu_get_status, RSND_MOD_SSIU, i); if (ret) return ret; -- cgit v1.2.3-58-ga151 From 6ab6a2474e0dce02f71e92adb9778a168a8931f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 3 Sep 2018 07:09:17 +0000 Subject: ASoC: rsnd: merge .nolock_start and .prepare Main purpose of .nolock_start is we need to call some function without spinlock. OTOH we have .prepare which main purpose is called under atomic context. Then, it is called without spinlock. In summary, our main callback init/quit, and start/stop are called under "atomic context and with spinlock". And some function need to be called under "non-atomic context or without spinlock". Let's merge .nolock_start and prepare to be more clear code. Then, let's rename nolock_stop to cleanup Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 13 ++----------- sound/soc/sh/rcar/dma.c | 20 ++++++++++---------- sound/soc/sh/rcar/rsnd.h | 26 ++++++++++---------------- 3 files changed, 22 insertions(+), 37 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 299fb457573d..e46415c807a0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -881,12 +881,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint; struct snd_pcm_runtime *runtime = substream->runtime; unsigned int max_channels = rsnd_rdai_channels_get(rdai); - int ret; int i; rsnd_dai_stream_init(io, substream); @@ -930,14 +928,7 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, -1); } - /* - * call rsnd_dai_call without spinlock - */ - ret = rsnd_dai_call(nolock_start, io, priv); - if (ret < 0) - rsnd_dai_call(nolock_stop, io, priv); - - return ret; + return 0; } static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, @@ -950,7 +941,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, /* * call rsnd_dai_call without spinlock */ - rsnd_dai_call(nolock_stop, io, priv); + rsnd_dai_call(cleanup, io, priv); rsnd_dai_stream_quit(io); } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index d3b1a4ae876a..f99c1ab3b0bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, return 0; } -static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) +static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv) { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); @@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, /* * DMAEngine release uses mutex lock. * Thus, it shouldn't be called under spinlock. - * Let's call it under nolock_start + * Let's call it under prepare */ if (dmaen->chan) dma_release_channel(dmaen->chan); @@ -126,9 +126,9 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, return 0; } -static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) +static int rsnd_dmaen_prepare(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv) { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); @@ -142,7 +142,7 @@ static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod, /* * DMAEngine request uses mutex lock. * Thus, it shouldn't be called under spinlock. - * Let's call it under nolock_start + * Let's call it under prepare */ dmaen->chan = rsnd_dmaen_request_channel(io, dma->mod_from, @@ -287,8 +287,8 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, static struct rsnd_mod_ops rsnd_dmaen_ops = { .name = "audmac", - .nolock_start = rsnd_dmaen_nolock_start, - .nolock_stop = rsnd_dmaen_nolock_stop, + .prepare = rsnd_dmaen_prepare, + .cleanup = rsnd_dmaen_cleanup, .start = rsnd_dmaen_start, .stop = rsnd_dmaen_stop, .pointer= rsnd_dmaen_pointer, diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 7a04b19e405c..e857311ee5c1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -295,15 +295,12 @@ struct rsnd_mod_ops { int (*fallback)(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv); - int (*nolock_start)(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv); - int (*nolock_stop)(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv); int (*prepare)(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv); + int (*cleanup)(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv); }; struct rsnd_dai_stream; @@ -323,7 +320,7 @@ struct rsnd_mod { * * 0xH0000CBA * - * A 0: nolock_start 1: nolock_stop + * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -335,8 +332,8 @@ struct rsnd_mod { * H 0: pointer * H 0: prepare */ -#define __rsnd_mod_shift_nolock_start 0 -#define __rsnd_mod_shift_nolock_stop 0 +#define __rsnd_mod_shift_prepare 0 +#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -348,12 +345,11 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ -#define __rsnd_mod_shift_prepare 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_nolock_start 1 -#define __rsnd_mod_add_nolock_stop -1 +#define __rsnd_mod_add_prepare 1 +#define __rsnd_mod_add_cleanup -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,10 +359,11 @@ struct rsnd_mod { #define __rsnd_mod_add_fallback 0 #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_add_pointer 0 -#define __rsnd_mod_add_prepare 0 #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_prepare 0 +#define __rsnd_mod_call_cleanup 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 @@ -376,9 +373,6 @@ struct rsnd_mod { #define __rsnd_mod_call_fallback 0 #define __rsnd_mod_call_hw_params 0 #define __rsnd_mod_call_pointer 0 -#define __rsnd_mod_call_nolock_start 0 -#define __rsnd_mod_call_nolock_stop 1 -#define __rsnd_mod_call_prepare 0 #define rsnd_mod_to_priv(mod) ((mod)->priv) #define rsnd_mod_name(mod) ((mod)->ops->name) -- cgit v1.2.3-58-ga151 From a45f8853a5f95e3760dfbd7ba09d3d597d247040 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Fri, 31 Aug 2018 20:14:35 +0300 Subject: ASoC: Add driver for PROTO Audio CODEC (with a WM8731) Add support for the MikroElektronika PROTO audio codec board. URL to the audio chip: http://www.mikroe.com/add-on-boards/audio-voice/audio-codec-proto/ Signed-off-by: Florian Meier Signed-off-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 11 +++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mikroe-proto.c | 165 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+) create mode 100644 sound/soc/atmel/mikroe-proto.c (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 64b784e96f84..81a0712d4f14 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -97,4 +97,15 @@ config SND_ATMEL_SOC_I2S help Say Y or M if you want to add support for Atmel ASoc driver for boards using I2S. + +config SND_SOC_MIKROE_PROTO + tristate "Support for Mikroe-PROTO board" + depends on OF + select SND_SOC_WM8731 + help + Say Y or M if you want to add support for MikroElektronika PROTO Audio + Board. This board contains the WM8731 codec, which can be configured + using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins. + Both playback and capture are supported. + endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index cd87cb4bcff5..9f41bfa0fea3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o snd-atmel-soc-classd-objs := atmel-classd.o snd-atmel-soc-pdmic-objs := atmel-pdmic.o snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o +snd-soc-mikroe-proto-objs := mikroe-proto.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o @@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o +obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c new file mode 100644 index 000000000000..d47aaa5bf75a --- /dev/null +++ b/sound/soc/atmel/mikroe-proto.c @@ -0,0 +1,165 @@ +/* + * ASoC driver for PROTO AudioCODEC (with a WM8731) + * + * Author: Florian Meier, + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include + +#include +#include +#include +#include + +#include "../codecs/wm8731.h" + +#define XTAL_RATE 12288000 /* This is fixed on this board */ + +static int snd_proto_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* Set proto sysclk */ + int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + XTAL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n", + ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget snd_proto_widget[] = { + SND_SOC_DAPM_MIC("Microphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route snd_proto_route[] = { + /* speaker connected to LHPOUT/RHPOUT */ + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Microphone Jack"}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_proto = { + .name = "snd_mikroe_proto", + .owner = THIS_MODULE, + .dapm_widgets = snd_proto_widget, + .num_dapm_widgets = ARRAY_SIZE(snd_proto_widget), + .dapm_routes = snd_proto_route, + .num_dapm_routes = ARRAY_SIZE(snd_proto_route), +}; + +static int snd_proto_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai; + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int dai_fmt; + int ret = 0; + + if (!np) { + dev_err(&pdev->dev, "No device node supplied\n"); + return -EINVAL; + } + + snd_proto.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&snd_proto, "model"); + if (ret) + return ret; + + dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + snd_proto.dai_link = dai; + snd_proto.num_links = 1; + + dai->name = "WM8731"; + dai->stream_name = "WM8731 HiFi"; + dai->codec_dai_name = "wm8731-hifi"; + dai->init = &snd_proto_init; + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "audio-codec node missing\n"); + return -EINVAL; + } + dai->codec_of_node = codec_np; + + cpu_np = of_parse_phandle(np, "i2s-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "i2s-controller missing\n"); + return -EINVAL; + } + dai->cpu_of_node = cpu_np; + dai->platform_of_node = cpu_np; + + dai_fmt = snd_soc_of_parse_daifmt(np, NULL, + &bitclkmaster, &framemaster); + if (bitclkmaster != framemaster) { + dev_err(&pdev->dev, "Must be the same bitclock and frame master\n"); + return -EINVAL; + } + if (bitclkmaster) { + dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + if (codec_np == bitclkmaster) + dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + else + dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + of_node_put(bitclkmaster); + of_node_put(framemaster); + dai->dai_fmt = dai_fmt; + + of_node_put(codec_np); + of_node_put(cpu_np); + + ret = snd_soc_register_card(&snd_proto); + if (ret && ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + + return ret; +} + +static int snd_proto_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_proto); +} + +static const struct of_device_id snd_proto_of_match[] = { + { .compatible = "mikroe,mikroe-proto", }, + {}, +}; +MODULE_DEVICE_TABLE(of, snd_proto_of_match); + +static struct platform_driver snd_proto_driver = { + .driver = { + .name = "snd-mikroe-proto", + .of_match_table = snd_proto_of_match, + }, + .probe = snd_proto_probe, + .remove = snd_proto_remove, +}; + +module_platform_driver(snd_proto_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-58-ga151 From e03546ddd3db5352a74dec247dbdaa29889e93f7 Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Fri, 17 Aug 2018 16:35:43 +0100 Subject: ASoC: core: Don't schedule DAPM work if already in target state When dapm_power_widgets() is called, the dapm_pre_sequence_async() and dapm_post_sequence_async() functions are scheduled for all DAPM contexts (apart from the card DAPM context) regardless of whether the DAPM context is already in the desired state. The overhead of this is not insignificant and the more DAPM contexts there are the more overhead there is. For example, on the Tegra124 Jetson TK1, when profiling the time taken to execute the dapm_power_widgets() the following times were observed. Times for function dapm_power_widgets() are (us): Min 23, Ave 190, Max 434, Count 39 Here 'Count' is the number of times that dapm_power_widgets() has been called. Please note that the above time were measured using ktime_get() to log the time on entry and exit from dapm_power_widgets(). So it should be noted that these times may not be purely the time take to execute this function if it is preempted. However, after applying this patch and measuring the time taken to execute dapm_power_widgets() again a significant improvement is seen as shown below. Times for function dapm_power_widgets() are (us): Min 4, Ave 16, Max 82, Count 39 Therefore, optimise the dapm_power_widgets() function by only scheduling the dapm_pre/post_sequence_async() work if the DAPM context is not in the desired state. Signed-off-by: Jon Hunter Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d7be3981f026..9feccd2e7c11 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1952,7 +1952,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1976,7 +1976,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } -- cgit v1.2.3-58-ga151 From 18d545bb2599d6e5b0747351eaeebb0160d261f9 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Tue, 4 Sep 2018 10:36:17 -0500 Subject: ASoC: tlv320aic31xx: Add overflow detection support Similar to short circuit detection, when the ADC/DAC is saturated and overflows poor audio quality can result and should be reported to the user. This device support Automatic Dynamic Range Compression (DRC) to reduce this but it is not enabled currently in this driver. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 34 ++++++++++++++++++++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 7 +++++++ 2 files changed, 39 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2abe51d9f879..608ad49ad978 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1409,7 +1409,7 @@ static irqreturn_t aic31xx_irq(int irq, void *data) if (value) handled = true; else - goto exit; + goto read_overflow; if (value & AIC31XX_HPLSCDETECT) dev_err(dev, "Short circuit on Left output is detected\n"); @@ -1419,6 +1419,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) AIC31XX_HPRSCDETECT)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); +read_overflow: + ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read overflow flag: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_DAC_OF_LEFT) + dev_warn(dev, "Left-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_RIGHT) + dev_warn(dev, "Right-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_SHIFTER) + dev_warn(dev, "DAC barrel shifter overflow has occurred\n"); + if (value & AIC31XX_ADC_OF) + dev_warn(dev, "ADC overflow has occurred\n"); + if (value & AIC31XX_ADC_OF_SHIFTER) + dev_warn(dev, "ADC barrel shifter overflow has occurred\n"); + if (value & ~(AIC31XX_DAC_OF_LEFT | + AIC31XX_DAC_OF_RIGHT | + AIC31XX_DAC_OF_SHIFTER | + AIC31XX_ADC_OF | + AIC31XX_ADC_OF_SHIFTER)) + dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value); + exit: if (handled) return IRQ_HANDLED; @@ -1499,7 +1528,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, - AIC31XX_SC); + AIC31XX_SC | + AIC31XX_ENGINE); ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, NULL, aic31xx_irq, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52e171988906..2636f2c6bc79 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -173,6 +173,13 @@ struct aic31xx_pdata { #define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1) #define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0) +/* AIC31XX_OFFLAG */ +#define AIC31XX_DAC_OF_LEFT BIT(7) +#define AIC31XX_DAC_OF_RIGHT BIT(6) +#define AIC31XX_DAC_OF_SHIFTER BIT(5) +#define AIC31XX_ADC_OF BIT(3) +#define AIC31XX_ADC_OF_SHIFTER BIT(1) + /* AIC31XX_INTRDACFLAG */ #define AIC31XX_HPLSCDETECT BIT(7) #define AIC31XX_HPRSCDETECT BIT(6) -- cgit v1.2.3-58-ga151 From 3db769f17714ae65f2faf44ff2bae9d52f4bd46b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 3 Sep 2018 02:12:40 +0000 Subject: ASoC: add for_each_link_codecs() macro ALSA SoC snd_soc_dai_link has snd_soc_dai_link_component array for codecs. To be more readable code, this patch adds new for_each_link_codecs() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/meson/axg-card.c | 5 +++-- sound/soc/soc-core.c | 17 ++++++++--------- 3 files changed, 15 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 96c19aabf21b..ce42c578fe82 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -978,6 +978,10 @@ struct snd_soc_dai_link { struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; +#define for_each_link_codecs(link, i, codec) \ + for ((i) = 0; \ + ((i) < link->num_codecs) && ((codec) = &link->codecs[i]); \ + (i)++) struct snd_soc_codec_conf { /* diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index b76a5f4f1785..3e68f1aa68ff 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -97,14 +97,15 @@ static void axg_card_clean_references(struct axg_card *priv) { struct snd_soc_card *card = &priv->card; struct snd_soc_dai_link *link; + struct snd_soc_dai_link_component *codec; int i, j; if (card->dai_link) { for (i = 0; i < card->num_links; i++) { link = &card->dai_link[i]; of_node_put(link->cpu_of_node); - for (j = 0; j < link->num_codecs; j++) - of_node_put(link->codecs[j].of_node); + for_each_link_codecs(link, j, codec) + of_node_put(codec->of_node); } } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e9c2304afaf1..eeab492e954f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1071,6 +1071,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) { int i, ret; + struct snd_soc_dai_link_component *codec; ret = snd_soc_init_platform(card, link); if (ret) { @@ -1084,19 +1085,19 @@ static int soc_init_dai_link(struct snd_soc_card *card, return ret; } - for (i = 0; i < link->num_codecs; i++) { + for_each_link_codecs(link, i, codec) { /* * Codec must be specified by 1 of name or OF node, * not both or neither. */ - if (!!link->codecs[i].name == - !!link->codecs[i].of_node) { + if (!!codec->name == + !!codec->of_node) { dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", link->name); return -EINVAL; } /* Codec DAI name must be specified */ - if (!link->codecs[i].dai_name) { + if (!codec->dai_name) { dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", link->name); return -EINVAL; @@ -3796,10 +3797,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); */ void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *component = dai_link->codecs; + struct snd_soc_dai_link_component *component; int index; - for (index = 0; index < dai_link->num_codecs; index++, component++) { + for_each_link_codecs(dai_link, index, component) { if (!component->of_node) break; of_node_put(component->of_node); @@ -3851,9 +3852,7 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, dai_link->num_codecs = num_codecs; /* Parse the list */ - for (index = 0, component = dai_link->codecs; - index < dai_link->num_codecs; - index++, component++) { + for_each_link_codecs(dai_link, index, component) { ret = of_parse_phandle_with_args(of_node, name, "#sound-dai-cells", index, &args); -- cgit v1.2.3-58-ga151 From 0b7990e38971da403ce223d8bdc758a817eb72f8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 3 Sep 2018 02:12:56 +0000 Subject: ASoC: add for_each_rtd_codec_dai() macro ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai. To be more readable code, this patch adds new for_each_rtd_codec_dai() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 7 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- sound/soc/meson/axg-card.c | 6 +- sound/soc/soc-core.c | 38 ++--- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 154 ++++++++++----------- 10 files changed, 118 insertions(+), 126 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index ce42c578fe82..6b68b31e3140 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1158,6 +1158,13 @@ struct snd_soc_pcm_runtime { unsigned int dev_registered:1; unsigned int pop_wait:1; }; +#define for_each_rtd_codec_dai(rtd, i, dai)\ + for ((i) = 0; \ + ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ + (i)++) +#define for_each_rtd_codec_dai_reverse(rtd, i, dai) \ + for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) + /* mixer control */ struct soc_mixer_control { diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 21a6490746a6..99e1320c485f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -488,11 +488,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* * Use channel 4 and 5 for the first amp diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index a892b37eab7c..a737c915d46a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -353,11 +353,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); if (ret < 0) { diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 582174d98c6c..5b4e90180827 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index b3670c8a5b8d..82675ed057de 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 7a89b4aad182..ef05fbc40c32 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; unsigned int mclk_clock; + struct snd_soc_dai *codec_dai; int i, ret; switch (mt8173_rt5650_priv.pll_from) { @@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, break; } - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock, params_rate(params) * 512); diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 3e68f1aa68ff..197e10a96e28 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -168,8 +168,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, if (be->mclk_fs) { mclk = params_rate(params) * be->mclk_fs; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) @@ -197,8 +196,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai; int ret, i; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, be->codec_masks[i].tx, be->codec_masks[i].rx, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eeab492e954f..390da15c4a8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -452,12 +452,12 @@ int snd_soc_suspend(struct device *dev) /* mute any active DACs */ list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) continue; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -625,12 +625,12 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) continue; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -674,15 +674,14 @@ int snd_soc_resume(struct device *dev) /* activate pins from sleep state */ list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; if (cpu_dai->active) pinctrl_pm_select_default_state(cpu_dai->dev); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (codec_dai->active) pinctrl_pm_select_default_state(codec_dai->dev); } @@ -877,6 +876,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + /* we can use for_each_rtd_codec_dai() after this */ codec_dais = rtd->codec_dais; for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); @@ -959,6 +959,7 @@ static void soc_remove_link_dais(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, int order) { int i; + struct snd_soc_dai *codec_dai; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -967,8 +968,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, } /* remove the CODEC DAI */ - for (i = 0; i < rtd->num_codecs; i++) - soc_remove_dai(rtd->codec_dais[i], order); + for_each_rtd_codec_dai(rtd, i, codec_dai) + soc_remove_dai(codec_dai, order); soc_remove_dai(rtd->cpu_dai, order); } @@ -1511,6 +1512,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + struct snd_soc_dai *codec_dai; int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", @@ -1524,8 +1526,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, return ret; /* probe the CODEC DAI */ - for (i = 0; i < rtd->num_codecs; i++) { - ret = soc_probe_dai(rtd->codec_dais[i], order); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = soc_probe_dai(codec_dai, order); if (ret) return ret; } @@ -1712,14 +1714,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; unsigned int i; int ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, @@ -2266,11 +2266,11 @@ int snd_soc_poweroff(struct device *dev) /* deactivate pins to sleep state */ list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; pinctrl_pm_select_sleep_state(cpu_dai->dev); - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { pinctrl_pm_select_sleep_state(codec_dai->dev); } } @@ -2776,10 +2776,10 @@ int snd_soc_register_card(struct snd_soc_card *card) /* deactivate pins to sleep state */ list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!codec_dai->active) pinctrl_pm_select_sleep_state(codec_dai->dev); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9feccd2e7c11..0a738cb439be 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2370,12 +2370,13 @@ static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + struct snd_soc_dai *codec_dai; int i, count = 0; mutex_lock(&rtd->card->dapm_mutex); - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + struct snd_soc_component *cmpnt = codec_dai->component; count += dapm_widget_show_component(cmpnt, buf + count); } @@ -4110,11 +4111,11 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_dapm_widget *sink, *source; int i; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI playback if widgets are valid */ if (codec_dai->playback_widget && cpu_dai->playback_widget) { @@ -4202,11 +4203,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { + struct snd_soc_dai *codec_dai; int i; soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); - for (i = 0; i < rtd->num_codecs; i++) - soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); + for_each_rtd_codec_dai(rtd, i, codec_dai) + soc_dapm_dai_stream_event(codec_dai, stream, event); dapm_power_widgets(rtd->card, event); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb6f4f1b65a9..79f5dd541d29 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->playback_active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->playback_active++; } else { cpu_dai->capture_active++; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->capture_active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->capture_active++; } cpu_dai->active++; cpu_dai->component->active++; - for (i = 0; i < rtd->num_codecs; i++) { - rtd->codec_dais[i]->active++; - rtd->codec_dais[i]->component->active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + codec_dai->active++; + codec_dai->component->active++; } } @@ -94,25 +95,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->playback_active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->playback_active--; } else { cpu_dai->capture_active--; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->capture_active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->capture_active--; } cpu_dai->active--; cpu_dai->component->active--; - for (i = 0; i < rtd->num_codecs; i++) { - rtd->codec_dais[i]->component->active--; - rtd->codec_dais[i]->active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + codec_dai->component->active--; + codec_dai->active--; } } @@ -253,6 +255,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); @@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_rates; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_rates; if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", @@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_channels || rtd->dai_link->symmetric_channels; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_channels; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_channels; if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", @@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_samplebits || rtd->dai_link->symmetric_samplebits; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_samplebits; if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", @@ -304,17 +307,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; + struct snd_soc_dai *codec_dai; unsigned int symmetry, i; symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || cpu_driver->symmetric_channels || link->symmetric_channels || cpu_driver->symmetric_samplebits || link->symmetric_samplebits; - for (i = 0; i < rtd->num_codecs; i++) + for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry = symmetry || - rtd->codec_dais[i]->driver->symmetric_rates || - rtd->codec_dais[i]->driver->symmetric_channels || - rtd->codec_dais[i]->driver->symmetric_samplebits; + codec_dai->driver->symmetric_rates || + codec_dai->driver->symmetric_channels || + codec_dai->driver->symmetric_samplebits; return symmetry; } @@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) unsigned int bits = 0, cpu_bits; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->playback.sig_bits == 0) { bits = 0; break; @@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) } cpu_bits = cpu_dai->driver->playback.sig_bits; } else { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->capture.sig_bits == 0) { bits = 0; break; @@ -372,6 +374,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; @@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) cpu_stream = &cpu_dai_drv->capture; /* first calculate min/max only for CODECs in the DAI link */ - for (i = 0; i < rtd->num_codecs; i++) { + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* * Skip CODECs which don't support the current stream type. @@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue; - codec_dai_drv = rtd->codec_dais[i]->driver; + codec_dai_drv = codec_dai->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - for (i = 0; i < rtd->num_codecs; i++) - pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) + pinctrl_pm_select_default_state(codec_dai->dev); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } component = NULL; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); @@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->active) { - ret = soc_pcm_apply_symmetry(substream, - rtd->codec_dais[i]); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -620,8 +621,7 @@ machine_err: i = rtd->num_codecs; codec_dai_err: - while (--i >= 0) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai_reverse(rtd, i, codec_dai) { if (codec_dai->driver->ops->shutdown) codec_dai->driver->ops->shutdown(substream, codec_dai); } @@ -641,9 +641,9 @@ out: pm_runtime_put_autosuspend(component->dev); } - for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->active) - pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!cpu_dai->active) cpu_dai->rate = 0; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (!codec_dai->active) codec_dai->rate = 0; } @@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->shutdown) codec_dai->driver->ops->shutdown(substream, codec_dai); } @@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put_autosuspend(component->dev); } - for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->active) - pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); @@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - for (i = 0; i < rtd->num_codecs; i++) - snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, + for_each_rtd_codec_dai(rtd, i, codec_dai) + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); @@ -920,6 +917,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { struct snd_pcm_hw_params codec_params; /* @@ -1018,8 +1015,7 @@ interface_err: i = rtd->num_codecs; codec_err: - while (--i >= 0) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai_reverse(rtd, i, codec_dai) { if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; @@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->active == 1) { codec_dai->rate = 0; codec_dai->channels = 0; @@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - for (i = 0; i < rtd->num_codecs; i++) { - if ((playback && rtd->codec_dais[i]->playback_active == 1) || - (!playback && rtd->codec_dais[i]->capture_active == 1)) - snd_soc_dai_digital_mute(rtd->codec_dais[i], 1, + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); } @@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); } @@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); @@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); @@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->delay) codec_delay = max(codec_delay, codec_dai->driver->ops->delay(substream, @@ -1388,6 +1379,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *dai; int i; if (dir == SND_SOC_DAPM_DIR_OUT) { @@ -1398,8 +1390,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, if (rtd->cpu_dai->playback_widget == widget) return true; - for (i = 0; i < rtd->num_codecs; ++i) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { if (dai->playback_widget == widget) return true; } @@ -1412,8 +1403,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, if (rtd->cpu_dai->capture_widget == widget) return true; - for (i = 0; i < rtd->num_codecs; ++i) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { if (dai->capture_widget == widget) return true; } @@ -1907,6 +1897,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + struct snd_soc_dai *codec_dai; int i; if (rtd->dai_link->be_hw_params_fixup) @@ -1923,10 +1914,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, return err; } - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->active) { + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (codec_dai->active) { err = soc_pcm_apply_symmetry(fe_substream, - rtd->codec_dais[i]); + codec_dai); if (err < 0) return err; } @@ -3041,8 +3032,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) -- cgit v1.2.3-58-ga151 From 3bbf5d34fd4a0c41246290b70338095ae291851b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Sep 2018 15:20:58 +0100 Subject: ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked Currently DAPM has a lot of similar code to handle errors from snd_soc_dapm_new_control_unlocked, and much of this code does not really accurately reflect what the function returns. Firstly, most places will check for a return value of -EPROBE_DEFER and silence any error messages in that case. The one notable exception here being dapm_kcontrol_data_alloc which does currently print any error messages in the case of snd_soc_dapm_new_control_unlocked returning NULL or an error. Additionally the error prints being silenced in these case are redundant as snd_soc_dapm_new_control_unlocked can only return -EPROBE_DEFER or NULL when failing. Secondly, most places will treat a return value of NULL as an -ENOMEM. This is not correct either since any error except EPROBE_DEFER will cause a return value of NULL from snd_soc_dapm_new_control_unlocked. Centralise this handling and the error messages within snd_soc_dapm_new_control_unlocked and update the callers to simply check IS_ERR and return. Note that this update is slightly simpler in the case of dapm_kcontrol_data_alloc where that is fairly close to the handling that was already in place. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 118 ++++++++--------------------------------------- sound/soc/soc-topology.c | 11 ----- 2 files changed, 19 insertions(+), 110 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a738cb439be..d13a25ce1275 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -364,10 +364,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +398,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -3433,23 +3425,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: mutex_unlock(&dapm->card->dapm_mutex); + return w; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); @@ -3464,24 +3441,20 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3490,22 +3463,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->pinctrl = devm_pinctrl_get(dapm->dev); if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; default: @@ -3519,7 +3484,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3596,6 +3561,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); } /** @@ -3621,19 +3593,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3944,21 +3903,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) { - ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; + if (IS_ERR(w)) goto outfree_kcontrol_news; - } w->params = params; w->num_params = num_params; @@ -3999,21 +3945,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4028,21 +3961,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..17f81b9a5754 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1565,17 +1565,6 @@ widget: widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } -- cgit v1.2.3-58-ga151 From 94e630a35d3383b42f12a873a5404bdf61e38e42 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Sep 2018 15:20:59 +0100 Subject: ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control Move the function snd_soc_dapm_new_control to be next to snd_soc_dapm_new_controls and add some kernel doc for it. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 37 +++++++++++++++++++++++-------------- 1 file changed, 23 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d13a25ce1275..c111e69b9a09 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3417,20 +3417,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) -{ - struct snd_soc_dapm_widget *w; - - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - w = snd_soc_dapm_new_control_unlocked(dapm, widget); - mutex_unlock(&dapm->card->dapm_mutex); - - return w; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); - struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) @@ -3570,6 +3556,29 @@ request_failed: return ERR_PTR(ret); } +/** + * snd_soc_dapm_new_control - create new dapm control + * @dapm: DAPM context + * @widget: widget template + * + * Creates new DAPM control based upon a template. + * + * Returns a widget pointer on success or an error pointer on failure + */ +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + mutex_unlock(&dapm->card->dapm_mutex); + + return w; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); + /** * snd_soc_dapm_new_controls - create new dapm controls * @dapm: DAPM context -- cgit v1.2.3-58-ga151 From 778ff5bb8689eb4fd05a72a409e32a3a34e23faf Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Sep 2018 15:21:00 +0100 Subject: ASoC: dapm: Move connection of CODEC to CODEC DAIs Currently, snd_soc_dapm_connect_dai_link_widgets connects up the routes representing normal DAIs, however CODEC to CODEC links are hooked up through separate infrastructure in soc_link_dai_widgets. Improve the consistency of the code by using snd_soc_dapm_connect_dai_link for both types of DAIs. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 6 --- sound/soc/soc-core.c | 47 --------------------- sound/soc/soc-dapm.c | 103 ++++++++++++++++++++++++++++++----------------- 3 files changed, 66 insertions(+), 90 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index fdaaafdc7a00..cb177fa21ce7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -406,12 +406,6 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai); int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 390da15c4a8b..4e9367aacc0c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1463,48 +1463,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, return 0; } -static int soc_link_dai_widgets(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dapm_widget *sink, *source; - int ret; - - if (rtd->num_codecs > 1) - dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); - - /* link the DAI widgets */ - sink = codec_dai->playback_widget; - source = cpu_dai->capture_widget; - if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, - dai_link->num_params, - source, sink); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - sink->name, source->name, ret); - return ret; - } - } - - sink = cpu_dai->playback_widget; - source = codec_dai->capture_widget; - if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, - dai_link->num_params, - source, sink); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - sink->name, source->name, ret); - return ret; - } - } - - return 0; -} - static int soc_probe_link_dais(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, int order) { @@ -1606,11 +1564,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, } else { INIT_DELAYED_WORK(&rtd->delayed_work, codec2codec_close_delayed_work); - - /* link the DAI widgets */ - ret = soc_link_dai_widgets(card, dai_link, rtd); - if (ret) - return ret; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c111e69b9a09..bbfcb7fc05cc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3860,12 +3860,10 @@ outfree_w_param: return NULL; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; @@ -3877,7 +3875,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", source->name, sink->name); if (!link_name) - return -ENOMEM; + return ERR_PTR(-ENOMEM); memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; @@ -3889,9 +3887,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.kcontrol_news = NULL; /* allocate memory for control, only in case of multiple configs */ - if (num_params > 1) { - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); + if (rtd->dai_link->num_params > 1) { + w_param_text = devm_kcalloc(card->dev, + rtd->dai_link->num_params, + sizeof(char *), GFP_KERNEL); if (!w_param_text) { ret = -ENOMEM; goto param_fail; @@ -3900,7 +3899,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.num_kcontrols = 1; template.kcontrol_news = snd_soc_dapm_alloc_kcontrol(card, - link_name, params, num_params, + link_name, + rtd->dai_link->params, + rtd->dai_link->num_params, w_param_text, &private_value); if (!template.kcontrol_news) { ret = -ENOMEM; @@ -3915,23 +3916,19 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (IS_ERR(w)) goto outfree_kcontrol_news; - w->params = params; - w->num_params = num_params; + w->params = rtd->dai_link->params; + w->num_params = rtd->dai_link->num_params; w->priv = rtd; - ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); - if (ret) - goto outfree_w; - return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + return w; -outfree_w: - devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); - snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); + snd_soc_dapm_free_kcontrol(card, &private_value, + rtd->dai_link->num_params, w_param_text); param_fail: devm_kfree(card->dev, link_name); - return ret; + return ERR_PTR(ret); } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, @@ -4041,33 +4038,65 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; - struct snd_soc_dapm_widget *sink, *source; + struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; + struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; int i; + if (rtd->dai_link->params) { + if (rtd->num_codecs > 1) + dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); + + playback_cpu = cpu_dai->capture_widget; + capture_cpu = cpu_dai->playback_widget; + } else { + playback = cpu_dai->playback_widget; + capture = cpu_dai->capture_widget; + playback_cpu = playback; + capture_cpu = capture; + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI playback if widgets are valid */ - if (codec_dai->playback_widget && cpu_dai->playback_widget) { - source = cpu_dai->playback_widget; - sink = codec_dai->playback_widget; + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + playback = snd_soc_dapm_new_dai(card, rtd, + playback_cpu, + codec); + + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); + } + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, source->name, - codec_dai->component->name, sink->name); + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); - snd_soc_dapm_add_path(&card->dapm, source, sink, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); } /* connect BE DAI capture if widgets are valid */ - if (codec_dai->capture_widget && cpu_dai->capture_widget) { - source = codec_dai->capture_widget; - sink = cpu_dai->capture_widget; + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + capture = snd_soc_dapm_new_dai(card, rtd, + codec, + capture_cpu); + + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); + } + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, source->name, - cpu_dai->component->name, sink->name); + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); - snd_soc_dapm_add_path(&card->dapm, source, sink, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } } @@ -4122,7 +4151,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. */ - if (rtd->dai_link->dynamic || rtd->dai_link->params) + if (rtd->dai_link->dynamic) continue; dapm_connect_dai_link_widgets(card, rtd); -- cgit v1.2.3-58-ga151 From 4a75aae17b2a802a7267206414050408392c374c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Sep 2018 15:21:01 +0100 Subject: ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links Currently multi-CODEC is not supported on CODEC to CODEC links. There are common applications where this would be useful, such as connecting two mono amplifiers to an audio CODEC. Adding support simply requires an update of snd_soc_dai_link_event to loop over the attached CODEC DAIs. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 123 +++++++++++++++++++++++++++++---------------------- 1 file changed, 70 insertions(+), 53 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bbfcb7fc05cc..40f27c95da61 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3614,7 +3614,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; const struct snd_soc_pcm_stream *config = w->params + w->params_select; @@ -3629,17 +3629,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) return -EINVAL; - /* We only support a single source and sink, pick the first */ - source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_OUT]); - sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_IN]); - - source = source_p->source->priv; - sink = sink_p->sink->priv; - /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3681,59 +3670,90 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; + + if (source->driver->ops->startup) { + ret = source->driver->ops->startup(&substream, + source); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + source->active++; } - source->active++; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, source); - if (ret < 0) - goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + if (sink->driver->ops->startup) { + ret = sink->driver->ops->startup(&substream, + sink); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + sink->active++; } - sink->active++; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, sink); - if (ret < 0) - goto out; break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to unmute: %d\n", ret); + ret = 0; + } break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to mute: %d\n", ret); + ret = 0; + } + + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; - source->active--; - if (source->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - source->driver->ops->shutdown(&substream, source); + source->active--; + if (source->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + source->driver->ops->shutdown(&substream, + source); + } } - sink->active--; - if (sink->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - sink->driver->ops->shutdown(&substream, sink); + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + sink->active--; + if (sink->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + sink->driver->ops->shutdown(&substream, sink); + } } break; @@ -4043,9 +4063,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, int i; if (rtd->dai_link->params) { - if (rtd->num_codecs > 1) - dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); - playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { -- cgit v1.2.3-58-ga151 From 243bcfafcd9a23a20867fd488dc3a35264918d87 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Sep 2018 15:21:02 +0100 Subject: ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime Larger CODECs may contain many several hundred widgets and which set of parameters is selected only needs to be recorded on a per DAI basis. As such move the selected CODEC to CODEC link params to be stored in the runtime rather than the DAPM widget, to save some memory. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 19 +++++++++++-------- 3 files changed, 13 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index cb177fa21ce7..bd8163f151cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -584,9 +584,6 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ struct pinctrl *pinctrl; /* attached pinctrl */ - const struct snd_soc_pcm_stream *params; /* params for dai links */ - unsigned int num_params; /* number of params for dai links */ - unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6b68b31e3140..821bf99992b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1130,6 +1130,8 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; + unsigned int params_select; /* currently selected param for dai link */ + /* Dynamic PCM BE runtime data */ struct snd_soc_dpcm_runtime dpcm[2]; int fe_compr; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40f27c95da61..9f4edcd19e02 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1018,9 +1018,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_kcontrol *kcontrol; struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; + struct snd_soc_pcm_runtime *rtd = w->priv; /* create control for links with > 1 config */ - if (w->num_params <= 1) + if (rtd->dai_link->num_params <= 1) return 0; /* add kcontrol */ @@ -3617,13 +3618,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; - const struct snd_soc_pcm_stream *config = w->params + w->params_select; + const struct snd_soc_pcm_stream *config; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; int ret; + config = rtd->dai_link->params + rtd->params_select; + if (WARN_ON(!config) || WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) @@ -3772,8 +3775,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; - ucontrol->value.enumerated.item[0] = w->params_select; + ucontrol->value.enumerated.item[0] = rtd->params_select; return 0; } @@ -3782,18 +3786,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; /* Can't change the config when widget is already powered */ if (w->power) return -EBUSY; - if (ucontrol->value.enumerated.item[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == rtd->params_select) return 0; - if (ucontrol->value.enumerated.item[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params) return -EINVAL; - w->params_select = ucontrol->value.enumerated.item[0]; + rtd->params_select = ucontrol->value.enumerated.item[0]; return 0; } @@ -3936,8 +3941,6 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, if (IS_ERR(w)) goto outfree_kcontrol_news; - w->params = rtd->dai_link->params; - w->num_params = rtd->dai_link->num_params; w->priv = rtd; return w; -- cgit v1.2.3-58-ga151 From c24fb71fa4f764f02c17cbf88a969f109794e602 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 6 Sep 2018 10:39:01 +0100 Subject: ASoC: hdac_hdmi: remove redundant check for !port condition The !port check is redundant as it being performed in the following check. Remove it. Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..dc6a0dfea050 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1961,9 +1961,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head); - if (!port) - return 0; - if (!port || !port->eld.eld_valid) return 0; -- cgit v1.2.3-58-ga151 From 501683d0cd54714de78501efe945bbe4356b922b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 6 Sep 2018 03:21:16 +0000 Subject: ASoC: rsnd: gen: use tab instead of white-space commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0") added new SSIU registers. But it is using white-space for it. This patch fixup it to use tab. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 48 ++++++++++++++++++++++++------------------------ 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3032869a7f26..1f7881cc16b2 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -222,30 +222,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80), RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80), RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), - RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), - RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), - RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), + RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), + RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), + RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), }; static const struct rsnd_regmap_field_conf conf_scu[] = { -- cgit v1.2.3-58-ga151 From dabdbe3ae0cb9a67872fa4ac80ffdef61391f645 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 6 Sep 2018 03:22:01 +0000 Subject: ASoC: rsnd: don't use %p for dev_dbg() rsnd driver sometimes want to know which address is used when debugging. But it will indicate "(____ptrval____)" if it used "%p" on dev_dbg(). Let's use "%pa" or "%px" for it. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/dma.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 3a3064dda57f..b100c44ec3a3 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -577,7 +577,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) int i; for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "%s : %p : %ld\n", + dev_dbg(dev, "%s : %pa : %ld\n", clk_name[i], clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", @@ -590,7 +590,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) * by BRGCKR::BRGCKR_31 */ for_each_rsnd_clkout(clk, adg, i) - dev_dbg(dev, "clkout %d : %p : %ld\n", i, + dev_dbg(dev, "clkout %d : %pa : %ld\n", i, clk, clk_get_rate(clk)); } #else diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index f99c1ab3b0bd..c19342d18998 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -393,7 +393,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg) struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data); + dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data); iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg)); } -- cgit v1.2.3-58-ga151 From 9ab708aef61f5620113269a9d1bdb1543d1207d0 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 6 Sep 2018 11:41:52 +0100 Subject: ASoC: sgtl5000: avoid division by zero if lo_vag is zero In the case where lo_vag <= SGTL5000_LINE_OUT_GND_BASE, lo_vag is set to zero and later vol_quot is computed by dividing by lo_vag causing a division by zero error. Fix this by avoiding a zero division and set vol_quot to zero in this specific case so that the lowest setting for i is correctly set. Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 60764f6201b1..add18d6d77da 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component) * Searching for a suitable index solving this formula: * idx = 40 * log10(vag_val / lo_cagcntrl) + 15 */ - vol_quot = (vag * 100) / lo_vag; + vol_quot = lo_vag ? (vag * 100) / lo_vag : 0; lo_vol = 0; for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) { if (vol_quot >= vol_quot_table[i]) -- cgit v1.2.3-58-ga151 From fc269c0396448cabe1afd648c0b335669aa347b7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 6 Sep 2018 17:41:55 +0100 Subject: ASoC: dapm: Avoid uninitialised variable warning Commit 4a75aae17b2a ("ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links") adds loops that iterate over multiple CODECs in snd_soc_dai_link_event. This also introduced a compiler warning for a potentially uninitialised variable in the case no CODECs are present. This should never be the case as the DAI link must by definition contain at least 1 CODEC however probably best to avoid the compiler warning by initialising ret to zero. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e02..e496bc642c99 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; -- cgit v1.2.3-58-ga151 From 3b857472f34faa7d11001afa5e158833812c98d7 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Tue, 7 Aug 2018 12:19:16 -0500 Subject: ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation Playback of 44.1Khz contents with HDMI plugged returns "Invalid pipe config" because HDMI paths in the FW topology are configured to operate at 48Khz. This patch filters out sampling rates not supported at hdac_hdmi_create_dais() to let user space SRC to do the converting. Signed-off-by: Yong Zhi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index dc6a0dfea050..41d90dc6ebf7 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdev->dev, dai_name, GFP_KERNEL); -- cgit v1.2.3-58-ga151 From 3004136b90bedc9e254ff659adb7a60299e9495e Mon Sep 17 00:00:00 2001 From: Grant Grundler Date: Thu, 6 Sep 2018 17:27:28 -0700 Subject: ASoC: max98373: usleep_range() needs include/delay.h Commit ca917f9fe1a0fab added use of usleep_range() but not the corresponding "include ". The result is with Chrome OS won't build because warnings are forced to be errors: mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration] usleep_range(10000, 11000); ^ Including delay.h "fixes" this. Signed-off-by: Grant Grundler Reviewed-by: Benson Leung Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f766d0d2..d6868c9a9ce6 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -2,6 +2,7 @@ // Copyright (c) 2017, Maxim Integrated #include +#include #include #include #include -- cgit v1.2.3-58-ga151 From 2e558a8127de7b2ed3302f9adcf332ba3feeadb2 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 7 Sep 2018 22:40:33 +0300 Subject: ASoC: dapm: Fix a couple uninitialized ret variables Smatch complains that these variables could be uninitialized. The first one in snd_soc_dai_link_event() is probably a false positive, because probably we know the lists are not empty. I would normally ignore the warning, but GCC complains here as well so I just silenced the warning. The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it leads to a bogus dereference in the caller. Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e496bc642c99..0dcdcc23dcfd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3938,8 +3938,10 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) + if (IS_ERR(w)) { + ret = PTR_ERR(w); goto outfree_kcontrol_news; + } w->priv = rtd; -- cgit v1.2.3-58-ga151 From 0712e0288b7602efbde95d95bca4c651ccad01b8 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 10 Sep 2018 11:40:48 +0300 Subject: ASoC: qdsp6: q6asm-dai: clean up a return Smatch complains that if both "psubstream" and "csubstream" are NULL then "ret" is uninitialized. That probably can't happen, but it's cleaner to just return zero anyway so let's do that. Signed-off-by: Dan Carpenter Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9db9a2944ef2..c75fab38905d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) } } - return ret; + return 0; } static void q6asm_dai_pcm_free(struct snd_pcm *pcm) -- cgit v1.2.3-58-ga151 From e14614dc5153ad41f7d1e5b125e4cd155ca79aa2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:00:15 +0000 Subject: ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index d3b69682d9c2..6291ec7f9dd6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev) struct ssc_device *ssc = dev_get_drvdata(dev); int ret; - ret = snd_soc_register_component(dev, &atmel_ssc_component, + ret = devm_snd_soc_register_component(dev, &atmel_ssc_component, &atmel_ssc_dai, 1); if (ret) { dev_err(dev, "Could not register DAI: %d\n", ret); - goto err; + return ret; } if (ssc->pdata->use_dma) @@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + return ret; } return 0; - -err_unregister_dai: - snd_soc_unregister_component(dev); -err: - return ret; } static void asoc_ssc_exit(struct device *dev) @@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev) atmel_pcm_dma_platform_unregister(dev); else atmel_pcm_pdc_platform_unregister(dev); - - snd_soc_unregister_component(dev); } /** -- cgit v1.2.3-58-ga151 From 570f75b93551a6d70ecbd0a7b6d962b4ca4722f0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:00:49 +0000 Subject: ASoC: bcm: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/bcm/cygnus-ssp.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index b733f1446353..b7c358b48d8d 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev) cygaud->active_ports = 0; dev_dbg(dev, "Registering %d DAIs\n", active_port_count); - err = snd_soc_register_component(dev, &cygnus_ssp_component, + err = devm_snd_soc_register_component(dev, &cygnus_ssp_component, cygnus_ssp_dai, active_port_count); if (err) { dev_err(dev, "snd_soc_register_dai failed\n"); @@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev) if (cygaud->irq_num <= 0) { dev_err(dev, "platform_get_irq failed\n"); err = cygaud->irq_num; - goto err_irq; + return err; } err = audio_clk_init(pdev, cygaud); if (err) { dev_err(dev, "audio clock initialization failed\n"); - goto err_irq; + return err; } err = cygnus_soc_platform_register(dev, cygaud); if (err) { dev_err(dev, "platform reg error %d\n", err); - goto err_irq; + return err; } return 0; - -err_irq: - snd_soc_unregister_component(dev); - return err; } static int cygnus_ssp_remove(struct platform_device *pdev) { cygnus_soc_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- cgit v1.2.3-58-ga151 From 10ccaa39d7628470a3de4aae9d2346a55cbee46e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:01:19 +0000 Subject: ASoC: hdac_hda: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hda.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 8c25a1332fa7..2aaa83028e55 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -448,7 +448,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) return -ENOMEM; /* ASoC specific initialization */ - ret = snd_soc_register_component(&hdev->dev, + ret = devm_snd_soc_register_component(&hdev->dev, &hdac_hda_codec, hdac_hda_dais, ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { @@ -464,7 +464,6 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) static int hdac_hda_dev_remove(struct hdac_device *hdev) { - snd_soc_unregister_component(&hdev->dev); return 0; } -- cgit v1.2.3-58-ga151 From 4fe1984ebc086ee39dd57983a7fee84c96c954a7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:01:34 +0000 Subject: ASoC: rt5668: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/rt5668.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 3c19d03f2446..4412cd2910cd 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2587,14 +2587,12 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, } - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, rt5668_dai, ARRAY_SIZE(rt5668_dai)); } static int rt5668_i2c_remove(struct i2c_client *i2c) { - snd_soc_unregister_component(&i2c->dev); - return 0; } -- cgit v1.2.3-58-ga151 From 007ac42db9ff4c36e91d192353421c6209058e06 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:01:50 +0000 Subject: ASoC: tscs454: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/tscs454.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index ff85a0bf6170..93d84e5ae2d5 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c, /* Sync pg sel reg with cache */ regmap_write(tscs454->regmap, R_PAGESEL, 0x00); - ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, tscs454_dais, ARRAY_SIZE(tscs454_dais)); if (ret) { dev_err(&i2c->dev, "Failed to register component (%d)\n", ret); -- cgit v1.2.3-58-ga151 From bfacaa8c8956dd6076641043b7be848267a708eb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:02:38 +0000 Subject: ASoC: nuc900: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 81b09d740ed9..6384bb6dacfd 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (ret) goto out; - ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, + ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) goto out; @@ -373,8 +373,6 @@ out: static int nuc900_ac97_drvremove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); - nuc900_ac97_data = NULL; snd_soc_set_ac97_ops(NULL); -- cgit v1.2.3-58-ga151 From 642a722d3116fbd22e59ac027f81b5ecd285f17c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:02:54 +0000 Subject: ASoC: omap: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi-audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index 8a99a8837dc9..673a9eb153b2 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) default: return -EINVAL; } - ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component, + ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component, dai_drv, 1); if (ret) return ret; @@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "snd_soc_register_card failed (%d)\n", ret); - snd_soc_unregister_component(ad->dssdev); return ret; } @@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev) struct hdmi_audio_data *ad = platform_get_drvdata(pdev); snd_soc_unregister_card(ad->card); - snd_soc_unregister_component(ad->dssdev); return 0; } -- cgit v1.2.3-58-ga151 From afa88ee37b1383490df003ea005d1d9bc8afa8a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:03:25 +0000 Subject: ASoC: sh: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index c2b496398e6b..17622ceb98c0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev) if (ret != 0) return ret; - return snd_soc_register_component(&pdev->dev, &sh4_hac_component, + return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } static int hac_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } -- cgit v1.2.3-58-ga151 From fb77436a444e9836e6b1b0a457bc9c09cdce22f6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 7 Sep 2018 01:03:53 +0000 Subject: ASoC: txx9: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-ac97.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e2ad00e3cae1..1cfca698ae4b 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; - return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, + return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } -- cgit v1.2.3-58-ga151 From 18fbe800e6066050ab6ae7751708da04975cdc22 Mon Sep 17 00:00:00 2001 From: zhong jiang Date: Sat, 8 Sep 2018 16:36:19 +0800 Subject: ASoC: q6core: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to use the kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..ca1be7305524 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -105,12 +105,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +122,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; -- cgit v1.2.3-58-ga151 From ca92cc4636fdedf0d7ee88a5e50cd2b85c246a3b Mon Sep 17 00:00:00 2001 From: zhong jiang Date: Sat, 8 Sep 2018 16:36:20 +0800 Subject: ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..52a9915da0f5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); -- cgit v1.2.3-58-ga151 From e36a1d0d249aa09f94d551cadf043a7f9f7fae00 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 10 Sep 2018 15:28:39 +0100 Subject: ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai snd_soc_dapm_new_dai may return an error pointer and currently this isn't checked for in dapm_connect_dai_link_widgets. Add code to check the return value and not add routes in that case. Fixes: 778ff5bb8689 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs") Reported-by: Dan Carpenter Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0dcdcc23dcfd..43983c69f6aa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4087,6 +4087,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, playback = snd_soc_dapm_new_dai(card, rtd, playback_cpu, codec); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + continue; + } snd_soc_dapm_add_path(&card->dapm, playback_cpu, playback, NULL, NULL); @@ -4099,7 +4106,9 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, snd_soc_dapm_add_path(&card->dapm, playback, codec, NULL, NULL); } + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI capture if widgets are valid */ codec = codec_dai->capture_widget; @@ -4108,6 +4117,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture = snd_soc_dapm_new_dai(card, rtd, codec, capture_cpu); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + continue; + } snd_soc_dapm_add_path(&card->dapm, capture, capture_cpu, NULL, NULL); -- cgit v1.2.3-58-ga151 From 8dcb0c90c691de5b79608d04ec7941ef9b3fee9c Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 10 Sep 2018 22:50:27 +0530 Subject: ASoC: AMD: Fix simultaneous playback and capture on different channel If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++++++++------- sound/soc/amd/acp-pcm-dma.c | 8 ++++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879cccbd2c0..717a017f0db6 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index e359938e3d7e..8f3bc6e37f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -846,8 +846,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e8f4fa..dbbb1a85638d 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- cgit v1.2.3-58-ga151 From 1c8bc7b3de5e76cb89aacdc7be1475a028af505f Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sat, 25 Aug 2018 10:46:18 +0200 Subject: ASoC: pxa: switch to new ac97 bus support Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..29f577e6dfc0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include #include +#include #include #include #include @@ -27,43 +28,35 @@ #include #include -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- cgit v1.2.3-58-ga151 From ae7d1247d8673ebfd686b17e759d4be391165368 Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Tue, 11 Sep 2018 14:59:21 +0530 Subject: ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx() In functions snd_soc_get_volsw_sx() or snd_soc_put_volsw_sx(), if the result of (min + max) is negative, then fls() returns signed integer with value as 32. This leads to signed integer overflow as complete operation is considered as signed integer. UBSAN: Undefined behaviour in sound/soc/soc-ops.c:382:50 signed integer overflow: -2147483648 - 1 cannot be represented in type 'int' Call trace: [] __dump_stack lib/dump_stack.c:15 [inline] [] dump_stack+0xec/0x158 lib/dump_stack.c:51 [] ubsan_epilogue+0x18/0x50 lib/ubsan.c:164 [] handle_overflow+0xf8/0x130 lib/ubsan.c:195 [] __ubsan_handle_sub_overflow+0x34/0x44 lib/ubsan.c:211 [] snd_soc_get_volsw_sx+0x1a8/0x1f8 sound/soc/soc-ops.c:382 Typecast the operation to unsigned int to fix the issue. Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 592efb370c44..f4dc3d445aae 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; unsigned int val; int ret; @@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; int err = 0; unsigned int val, val_mask, val2 = 0; -- cgit v1.2.3-58-ga151 From a6ebf4c9770e918e601aa9bf4bc3cf4001dd3d4d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 07:02:04 +0000 Subject: ASoC: rt5668: remove empty rt5668_i2c_remove() rt5668_i2c_remove() is empty, and no longer needed. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/rt5668.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 4412cd2910cd..85ba04d6e7ae 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2591,11 +2591,6 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, rt5668_dai, ARRAY_SIZE(rt5668_dai)); } -static int rt5668_i2c_remove(struct i2c_client *i2c) -{ - return 0; -} - static void rt5668_i2c_shutdown(struct i2c_client *client) { struct rt5668_priv *rt5668 = i2c_get_clientdata(client); @@ -2626,7 +2621,6 @@ static struct i2c_driver rt5668_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5668_acpi_match), }, .probe = rt5668_i2c_probe, - .remove = rt5668_i2c_remove, .shutdown = rt5668_i2c_shutdown, .id_table = rt5668_i2c_id, }; -- cgit v1.2.3-58-ga151 From 2eda3cb108b699a6ff78a87e25143c153bc88e41 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 06:54:26 +0000 Subject: ASoC: soc-core: avoid nested code on soc_remove_dai() Nested code is not readable. This patch avoid it on soc_remove_dai(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e9367aacc0c..dde9b70b58b5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,17 +942,18 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { - err = dai->driver->remove(dai); - if (err < 0) - dev_err(dai->dev, - "ASoC: failed to remove %s: %d\n", - dai->name, err); - } - dai->probed = 0; + if (!dai || !dai->probed || + dai->driver->remove_order != order) + return; + + if (dai->driver->remove) { + err = dai->driver->remove(dai); + if (err < 0) + dev_err(dai->dev, + "ASoC: failed to remove %s: %d\n", + dai->name, err); } + dai->probed = 0; } static void soc_remove_link_dais(struct snd_soc_card *card, -- cgit v1.2.3-58-ga151 From 4f1b327e65a9516a46ea491ce72a5161be176af8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 06:59:01 +0000 Subject: ASoC: soc-core: remove unused num_dai_links ALSA SoC is counting card->dai_link_list user, but no-one is using it. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 4 ---- 2 files changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 821bf99992b8..e17a7ae9a155 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1060,7 +1060,6 @@ struct snd_soc_card { struct snd_soc_dai_link *dai_link; /* predefined links only */ int num_links; /* predefined links only */ struct list_head dai_link_list; /* all links */ - int num_dai_links; struct list_head rtd_list; int num_rtd; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dde9b70b58b5..7e18937c2214 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1013,7 +1013,6 @@ static void soc_remove_dai_links(struct snd_soc_card *card) link->name); list_del(&link->list); - card->num_dai_links--; } } @@ -1182,7 +1181,6 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, card->add_dai_link(card, dai_link); list_add_tail(&dai_link->list, &card->dai_link_list); - card->num_dai_links++; return 0; } @@ -1220,7 +1218,6 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { if (link == dai_link) { list_del(&link->list); - card->num_dai_links--; return; } } @@ -2712,7 +2709,6 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); INIT_LIST_HEAD(&card->dai_link_list); - card->num_dai_links = 0; INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; -- cgit v1.2.3-58-ga151 From 24d6638302b48328a58c13439276d4531af4ca7d Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Tue, 11 Sep 2018 01:39:32 +0900 Subject: ASoC: rockchip: add missing INTERLEAVED PCM attribute This patch adds SNDRV_PCM_INFO_INTERLEAVED into PCM hardware info. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index f77538319221..9e7b5fa4cf59 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, -- cgit v1.2.3-58-ga151 From e1e38ea14ea3294546e6350d05a1376197a73589 Mon Sep 17 00:00:00 2001 From: zhong jiang Date: Wed, 12 Sep 2018 11:41:49 +0800 Subject: ASoC: remove unneeded static set .owner field in platform_driver platform_driver_register will set the .owner field. So it is safe to remove the redundant assignment. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang Signed-off-by: Mark Brown --- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 - sound/soc/mediatek/mt6797/mt6797-mt6351.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 1 - 3 files changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 89f34efd9747..e5d49e6e2f99 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -150,7 +150,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = { static struct platform_driver mt2701_wm8960_machine = { .driver = { .name = "mt2701-wm8960", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt2701_wm8960_machine_dt_match, #endif diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index b1558c57b9ca..6e578e830e42 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -205,7 +205,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = { static struct platform_driver mt6797_mt6351_driver = { .driver = { .name = "mt6797-mt6351", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt6797_mt6351_dt_match, #endif diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 929b3fe289b0..a472d5eb2950 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = { .probe = snd_rk_mc_probe, .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = rockchip_sound_of_match, }, -- cgit v1.2.3-58-ga151 From e894efef9ac7c10b7727798dcc711cccf07569f9 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 12 Sep 2018 10:15:00 +0100 Subject: ASoC: core: add support to card rebind Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 85 +++++++++++++++++++++++++++++++++++----------------- 2 files changed, 60 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index e17a7ae9a155..1e093b399bc0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1097,6 +1097,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1373,6 +1374,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7e18937c2214..807f112fad80 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2679,6 +2680,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dai(rtd, j, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2688,7 +2716,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2719,28 +2746,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int j; - - for_each_rtd_codec_dai(rtd, j, codec_dai) { - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2750,12 +2772,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3099,7 +3117,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3139,6 +3157,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3166,6 +3196,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- cgit v1.2.3-58-ga151 From a7c439d6128de2cbc087ae7524b47f613ff8bc6c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 15:50:27 +0900 Subject: ASoC: soc-core: remove dai->driver NULL check It is strange if it has "dai" but doesn't have "dai->driver". And more over "dai->driver->xxx" is used everywhere without "dai->driver" pointer NULL checking. It got Oops already if "dai->driver" was NULL. Let's remove un-needed "dai->driver" NULL check. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 807f112fad80..325dc1964850 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2519,8 +2519,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->driver == NULL) - return -EINVAL; if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; return dai->driver->ops->set_fmt(dai, fmt); @@ -2667,9 +2665,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction) { - if (!dai->driver) - return -ENOTSUPP; - if (dai->driver->ops->mute_stream) return dai->driver->ops->mute_stream(dai, mute, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && -- cgit v1.2.3-58-ga151 From 597d18325acdb48eb516ca9ef33d5148e79ca3bb Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 13 Sep 2018 14:08:15 -0500 Subject: ASoC: es8328: Fix fall-through annotations Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index e9fc2fd97d2f..3aedd609626c 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; -- cgit v1.2.3-58-ga151 From 982e386379f01aa6254e3313cff1edcb04c66685 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 13 Sep 2018 14:11:07 -0500 Subject: ASoC: hisilicon: fix fall-through annotations Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 53344a3b7a60..a69e5b11b3da 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; -- cgit v1.2.3-58-ga151 From 24b7a0aa1abec41b62c9a29a75e511d29f95033b Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Fri, 14 Sep 2018 01:36:04 +0000 Subject: ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c Remove duplicated include. Signed-off-by: YueHaibing Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c75fab38905d..c3806d7037fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -8,7 +8,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.3-58-ga151 From 4a9ed39477bd1635cf23b49e10f9e364329bbe46 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 06:51:14 +0000 Subject: ASoC: soc-core: manage platform name under snd_soc_init_platform() Now "platform" is controlled by snd_soc_dai_link_component, thus its "name" can be initialized in snd_soc_init_platform(), instead of soc_bind_dai_link() local. This patch do it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 33 +++++++++++++++++---------------- 1 file changed, 17 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 325dc1964850..d856b08f5f99 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -845,7 +845,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_component *component; struct snd_soc_dai **codec_dais; struct device_node *platform_of_node; - const char *platform_name; int i; if (dai_link->ignore) @@ -892,11 +891,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; - /* if there's no platform we match on the empty platform */ - platform_name = dai_link->platform->name; - if (!platform_name && !dai_link->platform->of_node) - platform_name = "snd-soc-dummy"; - /* find one from the set of registered platforms */ list_for_each_entry(component, &component_list, list) { platform_of_node = component->dev->of_node; @@ -907,7 +901,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, if (platform_of_node != dai_link->platform->of_node) continue; } else { - if (strcmp(component->name, platform_name)) + if (strcmp(component->name, dai_link->platform->name)) continue; } @@ -1020,24 +1014,31 @@ static void soc_remove_dai_links(struct snd_soc_card *card) static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + struct snd_soc_dai_link_component *platform = dai_link->platform; + /* * FIXME * * this function should be removed in the future */ /* convert Legacy platform link */ - if (dai_link->platform) - return 0; - - dai_link->platform = devm_kzalloc(card->dev, + if (!platform) { + platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); - if (!dai_link->platform) - return -ENOMEM; + if (!platform) + return -ENOMEM; - dai_link->platform->name = dai_link->platform_name; - dai_link->platform->of_node = dai_link->platform_of_node; - dai_link->platform->dai_name = NULL; + dai_link->platform = platform; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; + } + + /* if there's no platform we match on the empty platform */ + if (!platform->name && + !platform->of_node) + platform->name = "snd-soc-dummy"; return 0; } -- cgit v1.2.3-58-ga151 From be6ac0a9ced99403c435b2b2fe9ac4bd55749823 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 11 Sep 2018 06:51:45 +0000 Subject: ASoC: soc-core: add snd_soc_is_matching_component() To find (CPU/)Codec/Platform, we need to find component first (= on CPU/Codec/Platform), and find DAI from it (= CPU/Codec). These are similar operation but difficult to be simple, and has many duplicate code to finding component. This patch adds new snd_soc_is_matching_component(), and reduce duplicate codes. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d856b08f5f99..da2b2a758b6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -737,6 +737,24 @@ static struct snd_soc_component *soc_find_component( return NULL; } +static int snd_soc_is_matching_component( + const struct snd_soc_dai_link_component *dlc, + struct snd_soc_component *component) +{ + struct device_node *component_of_node; + + component_of_node = component->dev->of_node; + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + if (dlc->of_node && component_of_node != dlc->of_node) + return 0; + if (dlc->name && strcmp(component->name, dlc->name)) + return 0; + + return 1; +} + /** * snd_soc_find_dai - Find a registered DAI * @@ -753,19 +771,12 @@ struct snd_soc_dai *snd_soc_find_dai( { struct snd_soc_component *component; struct snd_soc_dai *dai; - struct device_node *component_of_node; lockdep_assert_held(&client_mutex); /* Find CPU DAI from registered DAIs*/ list_for_each_entry(component, &component_list, list) { - component_of_node = component->dev->of_node; - if (!component_of_node && component->dev->parent) - component_of_node = component->dev->parent->of_node; - - if (dlc->of_node && component_of_node != dlc->of_node) - continue; - if (dlc->name && strcmp(component->name, dlc->name)) + if (!snd_soc_is_matching_component(dlc, component)) continue; list_for_each_entry(dai, &component->dai_list, list) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) @@ -844,7 +855,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_component *component; struct snd_soc_dai **codec_dais; - struct device_node *platform_of_node; int i; if (dai_link->ignore) @@ -893,17 +903,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* find one from the set of registered platforms */ list_for_each_entry(component, &component_list, list) { - platform_of_node = component->dev->of_node; - if (!platform_of_node && component->dev->parent->of_node) - platform_of_node = component->dev->parent->of_node; - - if (dai_link->platform->of_node) { - if (platform_of_node != dai_link->platform->of_node) - continue; - } else { - if (strcmp(component->name, dai_link->platform->name)) - continue; - } + if (!snd_soc_is_matching_component(dai_link->platform, + component)) + continue; snd_soc_rtdcom_add(rtd, component); } -- cgit v1.2.3-58-ga151 From fbb673f7c6555d5434ad005f86b0d4368b1203d9 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 17 Sep 2018 19:03:09 +0800 Subject: ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly The value of period_bytes will get the zero before the hw_params() is not run completely. Move the function snd_pcm_lib_period_bytes() to copy work, and make sure that is not zero. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514-spi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 18686ffb0cd5..13809821e1f8 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + if (!period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, @@ -149,13 +157,11 @@ done: static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - size_t period_bytes; u8 buf[8]; if (!rt5514_dsp->substream) return; - period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - if (rt5514_dsp->buf_size % period_bytes) - rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * - period_bytes; - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); -- cgit v1.2.3-58-ga151 From 29ca7d32d7f10737e8d165fcf40fe31d44b06bee Mon Sep 17 00:00:00 2001 From: zhong jiang Date: Tue, 18 Sep 2018 16:16:24 +0800 Subject: ASoC: remove redundant include module.h already contained moduleparam.h, so it is safe to remove the redundant include. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/soc-dapm.c | 1 - 4 files changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..b613103d801b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -10,7 +10,6 @@ */ #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c..2a3e5fbd04e4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -13,7 +13,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 43edaf8cd276..593a11960888 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -11,7 +11,6 @@ */ #include -#include #include #include #include diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 43983c69f6aa..ee6b9758ec15 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -18,7 +18,6 @@ // device reopen. #include -#include #include #include #include -- cgit v1.2.3-58-ga151 From bf0fa00fd8410b377a3403adb58e32fc703e86e8 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 18 Sep 2018 19:51:08 +0800 Subject: ASoC: rt5682: Improve HP performance We change the settings while HP power-up for better performance. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 32 +++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 14 ++++++++++++++ 2 files changed, 43 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 731c6a849f69..83202e9e5abd 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1437,6 +1437,28 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); +static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); + break; + default: + return 0; + } + + return 0; +} + static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1449,8 +1471,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_LOGIC_CTRL_2, 0x0012); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x6000); - snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, - RT5682_NG2_EN_MASK, RT5682_NG2_EN); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); break; @@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, NULL, 0), + RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref1"}, {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, @@ -2607,6 +2629,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 8068140ebe3f..d82a8301fd74 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1214,6 +1214,20 @@ #define RT5682_JDH_NO_PLUG (0x1 << 4) #define RT5682_JDH_PLUG (0x0 << 4) +/* Bias current control 8 (0x0111) */ +#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2) +#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2) +#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2) +#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2) +#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 -- cgit v1.2.3-58-ga151 From 3f24f37adbc9a1059420a9c8f857e3490a4bce5e Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 18 Sep 2018 19:51:24 +0800 Subject: ASoC: rt5682: Remove HP volume control This patch removed Headphone Playback Volume control. Due to codec settings, we don't want the user to change HP analog gain. The user could use DAC1 Playback Volume control to change playback volume. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index afe7d5b19313..fad0bed82d79 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -749,7 +749,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); @@ -1108,10 +1107,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work) } static const struct snd_kcontrol_new rt5682_snd_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, - RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), - /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), -- cgit v1.2.3-58-ga151 From afd603e4ded0fad9e3102d514020af8494da1604 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 18 Sep 2018 19:50:38 +0800 Subject: ASoC: rt5682: Update calibration function The ADC/DAC path should open while calibration process. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7213b1cfb18a..2725eb72fb1b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2468,17 +2468,22 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); + regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2495,8 +2500,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); mutex_unlock(&rt5682->calibrate_mutex); -- cgit v1.2.3-58-ga151 From 28b20dde5e1c943ab899549a655ac4935cffccbb Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 18 Sep 2018 19:51:38 +0800 Subject: ASoC: rt5682: Fix the boost volume at the begining of playback This patch fixed the boost volume at the begining of playback while DAC volume set to lower level. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2725eb72fb1b..18099668e960 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -68,6 +68,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {0x01c1, 0x1000}, + {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; static const struct reg_default rt5682_reg[] = { @@ -1468,6 +1469,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_CTRL_2, 0x6000); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); break; case SND_SOC_DAPM_POST_PMD: @@ -1475,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x0); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x0000); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; default: -- cgit v1.2.3-58-ga151 From c50535ed6a10fcae1b64ae83c0f6b1eeb5535afc Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Tue, 18 Sep 2018 12:53:13 +0530 Subject: ASoC: AMD: Fix capture unstable in beginning for some runs alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 77b265bd0505..3135e9eafd18 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1036,16 +1036,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { period_bytes = frames_to_bytes(runtime, runtime->period_size); - dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); - if (dscr == rtd->dma_dscr_idx_1) - pos = period_bytes; - else - pos = 0; bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) + if (bytescount >= rtd->bytescount) bytescount -= rtd->bytescount; - delay = do_div(bytescount, period_bytes); - runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) { + pos = 0; + } else { + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } + if (bytescount > 0) { + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- cgit v1.2.3-58-ga151 From 37efe23dcca3c59cee662f1c28835020bef31cc0 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 18 Sep 2018 19:51:53 +0800 Subject: ASoC: rt5682: Minor code modification Minor code changes are: - improve the readability in patch list - add i2c remove function - regmap_register_patch changes to regmap_multi_reg_write Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 18099668e960..340f90497d07 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -67,7 +67,7 @@ struct rt5682_priv { }; static const struct reg_sequence patch_list[] = { - {0x01c1, 0x1000}, + {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; @@ -2584,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_calibrate(rt5682); - ret = regmap_register_patch(rt5682->regmap, patch_list, + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, ARRAY_SIZE(patch_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); @@ -2659,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_rt5682, + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2694,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; -- cgit v1.2.3-58-ga151 From 65ba4dd5200a537eae0f6b29e120f3971eac5a4d Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Tue, 18 Sep 2018 12:11:57 -0700 Subject: ASoC: rt5677-spi: Drop unused GPIO include This SPI driver does not use the legacy GPIO header so just delete it. Signed-off-by: Linus Walleij Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..84501c2020c7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.3-58-ga151 From 7afecb3073e357ebfe4087e4ab8bb493c32bb652 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:28:04 +0000 Subject: ASoC: convert for_each_rtd_codec_dai() for missing part commit 0b7990e38971 ("ASoC: add for_each_rtd_codec_dai() macro") added for_each_rtd_codec_dai(), but it didn't convert few loop which is not using "rtd". This patch fixup it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 79f5dd541d29..e387fff352c8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1301,6 +1301,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; + struct snd_soc_dai *dai; int i; dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); @@ -1318,8 +1319,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->playback_widget == widget) return be; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { if (dai->playback_widget == widget) return be; } @@ -1338,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->capture_widget == widget) return be; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { if (dai->capture_widget == widget) return be; } @@ -1435,6 +1434,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_dapm_widget *widget; + struct snd_soc_dai *dai; int prune = 0; /* Destroy any old FE <--> BE connections */ @@ -1449,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ - for (i = 0; i < dpcm->be->num_codecs; i++) { - struct snd_soc_dai *dai = dpcm->be->codec_dais[i]; + for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ @@ -1685,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; + struct snd_soc_dai *dai; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) @@ -1701,16 +1701,15 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, struct snd_soc_pcm_stream *codec_stream; int i; - for (i = 0; i < be->num_codecs; i++) { + for_each_rtd_codec_dai(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ - if (!snd_soc_dai_stream_valid(be->codec_dais[i], - stream)) + if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = be->codec_dais[i]->driver; + codec_dai_drv = dai->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -1795,6 +1794,7 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_dai *dai; int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -1806,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); - for (i = 0; i < be->num_codecs; i++) { + for_each_rtd_codec_dai(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ - if (!snd_soc_dai_stream_valid(be->codec_dais[i], - stream)) + if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = be->codec_dais[i]->driver; + codec_dai_drv = dai->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -2784,6 +2783,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) struct snd_soc_dpcm *dpcm; struct list_head *clients = &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; + struct snd_soc_dai *dai; list_for_each_entry(dpcm, clients, list_be) { @@ -2793,8 +2793,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) if (be->dai_link->ignore_suspend) continue; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; dev_dbg(be->dev, "ASoC: BE digital mute %s\n", -- cgit v1.2.3-58-ga151 From 6d11b12879144da5f5aa08071a8a7f95f3b5a4e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:28:30 +0000 Subject: ASoC: rename for_each_rtd_codec_dai_reverse to rollback commit 0b7990e38971 ("ASoC: add for_each_rtd_codec_dai() macro") added for_each_rtd_codec_dai_reverse(). but _rollback() is better naming than _reverse(). This patch rename it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/soc-pcm.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1e093b399bc0..ec1ae9f4feeb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1164,7 +1164,7 @@ struct snd_soc_pcm_runtime { for ((i) = 0; \ ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) -#define for_each_rtd_codec_dai_reverse(rtd, i, dai) \ +#define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e387fff352c8..1eff1dbb0d00 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -621,7 +621,7 @@ machine_err: i = rtd->num_codecs; codec_dai_err: - for_each_rtd_codec_dai_reverse(rtd, i, codec_dai) { + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { if (codec_dai->driver->ops->shutdown) codec_dai->driver->ops->shutdown(substream, codec_dai); } @@ -1015,7 +1015,7 @@ interface_err: i = rtd->num_codecs; codec_err: - for_each_rtd_codec_dai_reverse(rtd, i, codec_dai) { + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; -- cgit v1.2.3-58-ga151 From 7fe072b4df5d0cc832eb758c1eed243c145a2dfc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:28:49 +0000 Subject: ASoC: add for_each_card_prelinks() macro To be more readable code, this patch adds new for_each_card_prelinks() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/fsl/pcm030-audio-fabric.c | 5 +++-- sound/soc/generic/simple-card-utils.c | 6 ++---- sound/soc/intel/boards/skl_hda_dsp_generic.c | 5 +++-- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +++++++------ sound/soc/mediatek/mt2701/mt2701-wm8960.c | 13 +++++++------ sound/soc/mediatek/mt6797/mt6797-mt6351.c | 13 +++++++------ sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +++++++------ sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 ++++--- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 7 ++++--- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 7 ++++--- sound/soc/meson/axg-card.c | 3 +-- sound/soc/qcom/apq8096.c | 7 +++---- sound/soc/qcom/sdm845.c | 7 +++---- sound/soc/samsung/tm2_wm5110.c | 13 +++++++------ sound/soc/soc-core.c | 16 +++++++--------- 16 files changed, 73 insertions(+), 66 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index ec1ae9f4feeb..f94b989e7a1a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1120,6 +1120,10 @@ struct snd_soc_card { void *drvdata; }; +#define for_each_card_prelinks(card, i, link) \ + for ((i) = 0; \ + ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \ + (i)++) /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..e339f36cea95 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op) struct device_node *platform_np; struct snd_soc_card *card = &pcm030_card; struct pcm030_audio_data *pdata; + struct snd_soc_dai_link *dai_link; int ret; int i; @@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op) return -ENODEV; } - for (i = 0; i < card->num_links; i++) - card->dai_link[i].platform_of_node = platform_np; + for_each_card_prelinks(card, i, dai_link) + dai_link->platform_of_node = platform_np; ret = request_module("snd-soc-wm9712"); if (ret) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b400dbf1f834..f34cc6cddfa2 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -404,11 +404,9 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); int asoc_simple_card_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; - int num_links; + int i; - for (num_links = 0, dai_link = card->dai_link; - num_links < card->num_links; - num_links++, dai_link++) { + for_each_card_prelinks(card, i, dai_link) { of_node_put(dai_link->cpu_of_node); of_node_put(dai_link->codecs->of_node); } diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b213e9b47505..b415dd4c85f5 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -104,6 +104,7 @@ static struct snd_soc_card hda_soc_card = { static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) { struct snd_soc_card *card = &hda_soc_card; + struct snd_soc_dai_link *dai_link; u32 codec_count, codec_mask; int i, num_links, num_route; @@ -125,8 +126,8 @@ static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) card->num_links = num_links; card->num_dapm_routes = num_route; - for (i = 0; i < num_links; i++) - skl_hda_be_dai_links[i].platform_name = pdata->platform; + for_each_card_prelinks(card, i, dai_link) + dai_link->platform_name = pdata->platform; return 0; } diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 666282b865a8..875f84691535 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private), GFP_KERNEL); struct device *dev = &pdev->dev; + struct snd_soc_dai_link *dai_link; if (!priv) return -ENOMEM; @@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_cs42448_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_links->platform_name) continue; - mt2701_cs42448_dai_links[i].platform_of_node = platform_node; + dai_links->platform_of_node = platform_node; } card->dev = dev; @@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_cs42448_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_links->codec_name) continue; - mt2701_cs42448_dai_links[i].codec_of_node = codec_node; + dai_links->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index e5d49e6e2f99..c67f62935e53 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt2701_wm8960_card; struct device_node *platform_node, *codec_node; + struct snd_soc_dai_link *dai_link; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_wm8960_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_links->platform_name) continue; - mt2701_wm8960_dai_links[i].platform_of_node = platform_node; + dai_links->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_wm8960_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_links->codec_name) continue; - mt2701_wm8960_dai_links[i].codec_of_node = codec_node; + dai_links->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 6e578e830e42..ff2e0ca4384e 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt6797_mt6351_card; struct device_node *platform_node, *codec_node; + struct snd_soc_dai_link *dai_link; int ret, i; card->dev = &pdev->dev; @@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt6797_mt6351_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt6797_mt6351_dai_links[i].platform_of_node = platform_node; + dai_links->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt6797_mt6351_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_links->codec_name) continue; - mt6797_mt6351_dai_links[i].codec_of_node = codec_node; + dai_links->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 902d111016d6..4d6596d5cb07 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; struct device_node *codec_node, *platform_node; + struct snd_soc_dai_link *dai_link; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_max98090_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_max98090_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_max98090_dais[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codec_name) continue; - mt8173_max98090_dais[i].codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } card->dev = &pdev->dev; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 5b4e90180827..da5b58ce791b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -178,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5514_card; struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -187,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_rt5514_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_rt5514_codecs[0].of_node = diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 82675ed057de..d83cd039b413 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -224,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -233,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_rt5676_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_rt5676_codecs[0].of_node = diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef05fbc40c32..7edf250c8fb1 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -239,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) struct device_node *platform_node; struct device_node *np; const char *codec_capture_dai; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -248,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_codecs[0].of_node = diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 197e10a96e28..aa54d2c612c9 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -101,8 +101,7 @@ static void axg_card_clean_references(struct axg_card *priv) int i, j; if (card->dai_link) { - for (i = 0; i < card->num_links; i++) { - link = &card->dai_link[i]; + for_each_card_prelinks(card, i, link) { of_node_put(link->cpu_of_node); for_each_link_codecs(link, j, codec) of_node_put(codec->of_node); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1543e85629f8..fb45f396ab4a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct snd_soc_dai_link *link = card->dai_link; - int i, num_links = card->num_links; + struct snd_soc_dai_link *link; + int i; - for (i = 0; i < num_links; i++) { + for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - link++; } } diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 2a781d87ee65..9effbecc571f 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -195,15 +195,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static void sdm845_add_be_ops(struct snd_soc_card *card) { - struct snd_soc_dai_link *link = card->dai_link; - int i, num_links = card->num_links; + struct snd_soc_dai_link *link; + int i; - for (i = 0; i < num_links; i++) { + for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) { link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; } - link++; } } diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 43332c32d7e9..dc93941e01c3 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev) struct snd_soc_card *card = &tm2_card; struct tm2_machine_priv *priv; struct of_phandle_args args; + struct snd_soc_dai_link *dai_link; int num_codecs, ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev) } /* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */ - for (i = 0; i < card->num_links; i++) { + for_each_card_prelinks(card, i, dai_link) { unsigned int dai_index = 0; /* WM5110 */ - card->dai_link[i].cpu_name = NULL; - card->dai_link[i].platform_name = NULL; + dai_link->cpu_name = NULL; + dai_link->platform_name = NULL; if (num_codecs > 1 && i == card->num_links - 1) dai_index = 1; /* HDMI */ - card->dai_link[i].codec_of_node = codec_dai_node[dai_index]; - card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index]; - card->dai_link[i].platform_of_node = cpu_dai_node[dai_index]; + dai_link->codec_of_node = codec_dai_node[dai_index]; + dai_link->cpu_of_node = cpu_dai_node[dai_index]; + dai_link->platform_of_node = cpu_dai_node[dai_index]; } if (num_codecs > 1) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index da2b2a758b6d..532d8c59ed1e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1889,9 +1889,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; /* machine matches, so override the rtd data */ - for (i = 0; i < card->num_links; i++) { - - dai_link = &card->dai_link[i]; + for_each_card_prelinks(card, i, dai_link) { /* ignore this FE */ if (dai_link->dynamic) { @@ -1955,8 +1953,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_check_tplg_fes(card); /* bind DAIs */ - for (i = 0; i < card->num_links; i++) { - ret = soc_bind_dai_link(card, &card->dai_link[i]); + for_each_card_prelinks(card, i, dai_link) { + ret = soc_bind_dai_link(card, dai_link); if (ret != 0) goto base_error; } @@ -1969,8 +1967,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* add predefined DAI links to the list */ - for (i = 0; i < card->num_links; i++) - snd_soc_add_dai_link(card, card->dai_link+i); + for_each_card_prelinks(card, i, dai_link) + snd_soc_add_dai_link(card, dai_link); /* card bind complete so register a sound card */ ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, @@ -2714,12 +2712,12 @@ static int snd_soc_bind_card(struct snd_soc_card *card) int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; + struct snd_soc_dai_link *link; if (!card->name || !card->dev) return -EINVAL; - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai_link *link = &card->dai_link[i]; + for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { -- cgit v1.2.3-58-ga151 From 98061fdbfccc02aa0fd6637c67a0524aab385b8d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:29:16 +0000 Subject: ASoC: add for_each_card_links() macro To be more readable code, this patch adds new for_each_card_links() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 8 ++++---- 2 files changed, 10 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index f94b989e7a1a..1fffbaa819d9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1125,6 +1125,12 @@ struct snd_soc_card { ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \ (i)++) +#define for_each_card_links(card, link) \ + list_for_each_entry(dai_link, &(card)->dai_link_list, list) +#define for_each_card_links_safe(card, link, _link) \ + list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list) + + /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { struct device *dev; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 532d8c59ed1e..495173635642 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -816,7 +816,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card, lockdep_assert_held(&client_mutex); - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link->id != id) continue; @@ -1004,7 +1004,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) soc_remove_link_components(card, rtd, order); } - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK) dev_warn(card->dev, "Topology forgot to remove link %s?\n", link->name); @@ -1219,7 +1219,7 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, if (dai_link->dobj.type && card->remove_dai_link) card->remove_dai_link(card, dai_link); - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link == dai_link) { list_del(&link->list); return; @@ -2033,7 +2033,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ - list_for_each_entry(dai_link, &card->dai_link_list, list) { + for_each_card_links(card, dai_link) { if (soc_is_dai_link_bound(card, dai_link)) continue; -- cgit v1.2.3-58-ga151 From bcb1fd1fcd6507ba5a1f8610550135dc367aedb7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:29:35 +0000 Subject: ASoC: add for_each_card_rtds() macro To be more readable code, this patch adds new for_each_card_rtds() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +++ sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +-- sound/soc/soc-core.c | 48 ++++++++++++++-------------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 12 +++---- 6 files changed, 38 insertions(+), 34 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1fffbaa819d9..164418dbf40e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1130,6 +1130,10 @@ struct snd_soc_card { #define for_each_card_links_safe(card, link, _link) \ list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list) +#define for_each_card_rtds(card, rtd) \ + list_for_each_entry(rtd, &(card)->rtd_list, list) +#define for_each_card_rtds_safe(card, rtd, _rtd) \ + list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list) /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 41d90dc6ebf7..4e9854889a95 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1604,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->pcm && (rtd->pcm->device == device)) return rtd->pcm; } diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 6c36da560877..afc559866095 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -765,7 +765,7 @@ static int sst_soc_prepare(struct device *dev) snd_soc_poweroff(drv->soc_card->dev); /* set the SSPs to idle */ - list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + for_each_card_rtds(drv->soc_card, rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { @@ -786,7 +786,7 @@ static void sst_soc_complete(struct device *dev) return; /* restart SSPs */ - list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + for_each_card_rtds(drv->soc_card, rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 495173635642..7efcf3475d6f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -342,7 +342,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->no_pcm && !strcmp(rtd->dai_link->name, dai_link)) return rtd->pcm->streams[stream].substream; @@ -399,7 +399,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd, *_rtd; - list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + for_each_card_rtds_safe(card, rtd, _rtd) { list_del(&rtd->list); soc_free_pcm_runtime(rtd); } @@ -412,7 +412,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!strcmp(rtd->dai_link->name, dai_link)) return rtd; } @@ -452,7 +452,7 @@ int snd_soc_suspend(struct device *dev) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DACs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) @@ -467,7 +467,7 @@ int snd_soc_suspend(struct device *dev) } /* suspend all pcms */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -477,7 +477,7 @@ int snd_soc_suspend(struct device *dev) if (card->suspend_pre) card->suspend_pre(card); - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -488,10 +488,10 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -548,7 +548,7 @@ int snd_soc_suspend(struct device *dev) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -592,7 +592,7 @@ static void soc_resume_deferred(struct work_struct *work) card->resume_pre(card); /* resume control bus DAIs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -610,7 +610,7 @@ static void soc_resume_deferred(struct work_struct *work) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -625,7 +625,7 @@ static void soc_resume_deferred(struct work_struct *work) } /* unmute any active DACs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) @@ -639,7 +639,7 @@ static void soc_resume_deferred(struct work_struct *work) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -674,7 +674,7 @@ int snd_soc_resume(struct device *dev) return 0; /* activate pins from sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; @@ -694,7 +694,7 @@ int snd_soc_resume(struct device *dev) * have that problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; bus_control |= cpu_dai->driver->bus_control; } @@ -839,7 +839,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link == dai_link) return true; } @@ -994,13 +994,13 @@ static void soc_remove_dai_links(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) soc_remove_link_dais(card, rtd, order); } for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) soc_remove_link_components(card, rtd, order); } @@ -2014,7 +2014,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { ret = soc_probe_link_components(card, rtd, order); if (ret < 0) { dev_err(card->dev, @@ -2048,7 +2048,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { ret = soc_probe_link_dais(card, rtd, order); if (ret < 0) { dev_err(card->dev, @@ -2169,7 +2169,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; /* make sure any delayed work runs */ - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); /* free the ALSA card at first; this syncs with pending operations */ @@ -2211,13 +2211,13 @@ int snd_soc_poweroff(struct device *dev) /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); snd_soc_dapm_shutdown(card); /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; int i; @@ -2686,7 +2686,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) return ret; /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; int j; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ee6b9758ec15..8c5b065c8880 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4183,7 +4183,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; /* for each BE DAI link... */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { /* * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1eff1dbb0d00..09d0f668c78e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1307,7 +1307,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(be, &card->rtd_list, list) { + for_each_card_rtds(card, be) { if (!be->dai_link->no_pcm) continue; @@ -1326,7 +1326,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } else { - list_for_each_entry(be, &card->rtd_list, list) { + for_each_card_rtds(card, be) { if (!be->dai_link->no_pcm) continue; @@ -1382,7 +1382,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, int i; if (dir == SND_SOC_DAPM_DIR_OUT) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!rtd->dai_link->no_pcm) continue; @@ -1395,7 +1395,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, } } } else { /* SND_SOC_DAPM_DIR_IN */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!rtd->dai_link->no_pcm) continue; @@ -2761,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); /* shutdown all old paths first */ - list_for_each_entry(fe, &card->rtd_list, list) { + for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 0); if (ret) goto out; } /* bring new paths up */ - list_for_each_entry(fe, &card->rtd_list, list) { + for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 1); if (ret) goto out; -- cgit v1.2.3-58-ga151 From f70f18f7d459b7958a4d3944396e2bc4a9f7ed72 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:29:55 +0000 Subject: ASoC: add for_each_card_components() macro To be more readable code, this patch adds new for_each_card_components() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/intel/boards/broadwell.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++-- sound/soc/soc-core.c | 5 +++-- 6 files changed, 14 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 164418dbf40e..34efab6baff6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1135,6 +1135,9 @@ struct snd_soc_card { #define for_each_card_rtds_safe(card, rtd, _rtd) \ list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list) +#define for_each_card_components(card, component) \ + list_for_each_entry(component, &(card)->component_dev_list, card_list) + /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { struct device *dev; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 7b0ee67b4fc8..68e6543e6cb0 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { static int broadwell_suspend(struct snd_soc_card *card){ struct snd_soc_component *component; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); @@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){ static int broadwell_resume(struct snd_soc_card *card){ struct snd_soc_component *component; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "enabling jack detect for resume.\n"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b6dc524830b2..8587bd3d1cc1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1048,7 +1048,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card) if (!BYT_RT5640_JDSRC(byt_rt5640_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5640_codec_name)) { dev_dbg(component->dev, "disabling jack detect before suspend\n"); snd_soc_component_set_jack(component, NULL, NULL); @@ -1067,7 +1067,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card) if (!BYT_RT5640_JDSRC(byt_rt5640_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5640_codec_name)) { dev_dbg(component->dev, "re-enabling jack detect after resume\n"); snd_soc_component_set_jack(component, &priv->jack, NULL); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index f8a68bdb3885..8dffeecda55b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -742,7 +742,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card) if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5651_codec_name)) { dev_dbg(component->dev, "disabling jack detect before suspend\n"); snd_soc_component_set_jack(component, NULL, NULL); @@ -761,7 +761,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card) if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5651_codec_name)) { dev_dbg(component->dev, "re-enabling jack detect after resume\n"); snd_soc_component_set_jack(component, &priv->jack, NULL); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index e054318185ea..51f0d45d6f8f 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -347,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_component *component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strncmp(component->name, ctx->codec_name, sizeof(ctx->codec_name))) { @@ -364,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_component *component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strncmp(component->name, ctx->codec_name, sizeof(ctx->codec_name))) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7efcf3475d6f..673a694ede3e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -510,7 +510,7 @@ int snd_soc_suspend(struct device *dev) snd_soc_dapm_sync(&card->dapm); /* suspend all COMPONENTs */ - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); /* If there are paths active then the COMPONENT will be held with @@ -602,7 +602,7 @@ static void soc_resume_deferred(struct work_struct *work) cpu_dai->driver->resume(cpu_dai); } - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (component->suspended) { if (component->driver->resume) component->driver->resume(component); @@ -1354,6 +1354,7 @@ static int soc_probe_component(struct snd_soc_card *card, component->driver->num_dapm_routes); list_add(&dapm->list, &card->dapm_list); + /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); return 0; -- cgit v1.2.3-58-ga151 From 1a1035a9854fd893d487a84edccc1d5804e1d716 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:30:41 +0000 Subject: ASoC: add for_each_comp_order() macro To be more readable code, this patch adds new for_each_comp_order() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 18 ++++++------------ 2 files changed, 11 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 34efab6baff6..93aa894a57ef 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -372,6 +372,11 @@ #define SND_SOC_COMP_ORDER_LATE 1 #define SND_SOC_COMP_ORDER_LAST 2 +#define for_each_comp_order(order) \ + for (order = SND_SOC_COMP_ORDER_FIRST; \ + order <= SND_SOC_COMP_ORDER_LAST; \ + order++) + /* * Bias levels * diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 673a694ede3e..d8625ac2b201 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -992,14 +992,12 @@ static void soc_remove_dai_links(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link *link, *_link; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { for_each_card_rtds(card, rtd) soc_remove_link_dais(card, rtd, order); } - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { for_each_card_rtds(card, rtd) soc_remove_link_components(card, rtd, order); } @@ -1617,8 +1615,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) int order; int ret; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); @@ -1640,8 +1637,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) struct snd_soc_component *comp, *_comp; int order; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { list_for_each_entry_safe(comp, _comp, &card->aux_comp_list, card_aux_list) { @@ -2013,8 +2009,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* probe all components used by DAI links on this card */ - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { for_each_card_rtds(card, rtd) { ret = soc_probe_link_components(card, rtd, order); if (ret < 0) { @@ -2047,8 +2042,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* probe all DAI links on this card */ - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { for_each_card_rtds(card, rtd) { ret = soc_probe_link_dais(card, rtd, order); if (ret < 0) { -- cgit v1.2.3-58-ga151 From d2e24d64652bf9d272e5496ae8a562bc64facff3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:30:54 +0000 Subject: ASoC: add for_each_dpcm_fe() macro To be more readable code, this patch adds new for_each_dpcm_fe() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 3 +++ sound/soc/soc-pcm.c | 6 +++--- 2 files changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 9bb92f187af8..f130de6cfe8e 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,6 +103,9 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; +#define for_each_dpcm_fe(be, stream, dpcm) \ + list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) + /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 09d0f668c78e..e7916630e6fa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1252,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, be_substream = snd_soc_dpcm_get_substream(be, stream); - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; @@ -3219,7 +3219,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; @@ -3246,7 +3246,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; -- cgit v1.2.3-58-ga151 From 8d6258a4dd267838e2f10643c3d91b79fe75ef6e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Sep 2018 01:31:09 +0000 Subject: ASoC: add for_each_dpcm_be() macro To be more readable code, this patch adds new for_each_dpcm_be() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 7 +++++++ sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/soc-compress.c | 4 ++-- sound/soc/soc-pcm.c | 48 +++++++++++++++++++++----------------------- 6 files changed, 35 insertions(+), 30 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index f130de6cfe8e..4be3a2b7c106 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -106,6 +106,13 @@ struct snd_soc_dpcm_runtime { #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) +#define for_each_dpcm_be(fe, stream, dpcm) \ + list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ + list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_rollback(fe, stream, dpcm) \ + list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be) + /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 1033ac6631b0..01052a0808b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, int ret; /* Fetch the Back-End dma_data from DPCM */ - list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; struct snd_soc_dai *dai = be->cpu_dai; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6a55aa753003..ad702377a6c3 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -258,7 +258,7 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod, struct snd_pcm_hw_params *be_params; int stream = substream->stream; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { be_params = &dpcm->hw_params; if (params_channels(fe_params) != params_channels(be_params)) ctu->channels = params_channels(be_params); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index beccfbac7581..cd38a43b976f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -158,7 +158,7 @@ static int rsnd_src_hw_params(struct rsnd_mod *mod, struct snd_soc_dpcm *dpcm; struct snd_pcm_hw_params *be_params; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { be_params = &dpcm->hw_params; if (params_rate(fe_params) != params_rate(be_params)) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 409d082e80d1..699397a09167 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) ret = dpcm_be_dai_shutdown(fe, stream); /* mark FE's links ready to prune */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7916630e6fa..03f36e534050 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -174,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, { struct snd_soc_dpcm *dpcm; - list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) { + for_each_dpcm_be(fe, dir, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -1211,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; /* only add new dpcms */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { if (dpcm->be == be && dpcm->fe == fe) return 0; } @@ -1272,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; - list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", dpcm->be->dai_link->name); @@ -1438,7 +1438,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, int prune = 0; /* Destroy any old FE <--> BE connections */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { unsigned int i; /* is there a valid CPU DAI widget for this BE */ @@ -1544,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; } @@ -1555,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; /* disable any enabled and non active backends */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -1584,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) int err, count = 0; /* only startup BE DAIs that are either sinks or sources to this FE DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -1638,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) unwind: /* disable any enabled and non active backends */ - list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_rollback(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -1695,7 +1695,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_format) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; @@ -1736,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_chan) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; @@ -1788,7 +1788,7 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_chan) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; @@ -1891,7 +1891,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, } /* apply symmetry for BE */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -1976,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; /* only shutdown BEs that are either sinks or sources to this FE DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2040,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) /* only hw_params backends that are either sinks or sources * to this frontend DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2109,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; int ret; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2160,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) unwind: /* disable any enabled and non active backends */ - list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_rollback(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -2240,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dpcm *dpcm; int ret = 0; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2426,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; int ret = 0; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2636,7 +2636,7 @@ close: dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; @@ -2781,11 +2781,9 @@ out: int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; - struct list_head *clients = - &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; struct snd_soc_dai *dai; - list_for_each_entry(dpcm, clients, list_be) { + for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; int i; @@ -2834,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) ret = dpcm_fe_dai_startup(fe_substream); if (ret < 0) { /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -2857,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) ret = dpcm_fe_dai_shutdown(fe_substream); /* mark FE's links ready to prune */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -3326,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; -- cgit v1.2.3-58-ga151 From fc795bf7224efda8c35e505966c2757856064247 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Sep 2018 09:56:47 +0800 Subject: ASoC: rt5663: Remove the boost volume in the beginning of playback The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..2444fad7c2df 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- cgit v1.2.3-58-ga151 From 0310820c2738e92003d9dd8cabee77ff958a16dc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 21 Sep 2018 07:46:28 +0000 Subject: ASoC: tidyup for_each_card_prelinks() dai_link commit 7fe072b4df5d0 ("ASoC: add for_each_card_prelinks() macro") added new for_each_card_prelinks() macro, but it had typo. This patch fixup it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 ++++---- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 875f84691535..97f9f38ce6b3 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -311,9 +311,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -326,9 +326,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index c67f62935e53..6bc1d3d58e64 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -107,9 +107,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -122,9 +122,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index ff2e0ca4384e..cc41eb531653 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -172,7 +172,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) for_each_card_prelinks(card, i, dai_link) { if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -183,9 +183,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); -- cgit v1.2.3-58-ga151 From c78d42c7fbd6f2a27a665bcd1ab60c9df617f7c9 Mon Sep 17 00:00:00 2001 From: zhong jiang Date: Fri, 21 Sep 2018 18:24:58 +0800 Subject: ASoC: qcom: qdsp6: remove duplicated include from q6adm.c We include wait.h twice in q6adm.c. it is unnecessary. hence remove it. Further, order the include files as alphabet. Signed-off-by: zhong jiang Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include -#include -#include #include -#include -#include #include +#include +#include +#include #include #include -#include -#include -#include #include +#include +#include +#include +#include #include #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 -- cgit v1.2.3-58-ga151 From 624d1a7cd8991e33dad96ab4629a52c412540e65 Mon Sep 17 00:00:00 2001 From: Dmytro Prokopchuk Date: Fri, 21 Sep 2018 04:59:59 +0000 Subject: ASoC: rsnd: fixup SSI clock during suspend/resume modes Prepare <-> Cleanup functions pair has balanced calls. But in case of suspend mode no call to rsnd_soc_dai_shutdown() function, so cleanup isn't called. OTOH during resume mode function rsnd_soc_dai_prepare() is called, but calling rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero, bacause was not cleanup before). We need to call rsnd_ssi_prepare(), because it enables SSI clocks by calling rsnd_ssi_master_clk_start(). This patch allows to call prepare/cleanup functions always. Signed-off-by: Dmytro Prokopchuk Tested-by: Hiroyuki Yokoyama [kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 7 +++---- sound/soc/sh/rcar/rsnd.h | 14 +++++++------- 2 files changed, 10 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0bbc4b0ea2c6..6d1947515dc8 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -134,10 +134,9 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e857311ee5c1..4464d1d0a042 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -318,9 +318,8 @@ struct rsnd_mod { /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -331,9 +330,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_prepare 0 -#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -345,11 +343,13 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ +#define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_prepare 1 -#define __rsnd_mod_add_cleanup -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,7 +363,7 @@ struct rsnd_mod { #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 #define __rsnd_mod_call_prepare 0 -#define __rsnd_mod_call_cleanup 1 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- cgit v1.2.3-58-ga151 From 368dee9459472b44f760a35cd07a6f3b90b3e549 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 21 Sep 2018 05:23:01 +0000 Subject: ASoC: add for_each_component() macro To be more readable code, this patch adds new for_each_component() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d8625ac2b201..b9b33c8cac2e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -54,6 +54,9 @@ static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); +#define for_each_component(component) \ + list_for_each_entry(component, &component_list, list) + /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. @@ -176,7 +179,7 @@ static int dai_list_show(struct seq_file *m, void *v) mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) + for_each_component(component) list_for_each_entry(dai, &component->dai_list, list) seq_printf(m, "%s\n", dai->name); @@ -192,7 +195,7 @@ static int component_list_show(struct seq_file *m, void *v) mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) + for_each_component(component) seq_printf(m, "%s\n", component->name); mutex_unlock(&client_mutex); @@ -725,7 +728,7 @@ static struct snd_soc_component *soc_find_component( lockdep_assert_held(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (of_node) { if (component->dev->of_node == of_node) return component; @@ -775,7 +778,7 @@ struct snd_soc_dai *snd_soc_find_dai( lockdep_assert_held(&client_mutex); /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (!snd_soc_is_matching_component(dlc, component)) continue; list_for_each_entry(dai, &component->dai_list, list) { @@ -902,7 +905,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->codec_dai = codec_dais[0]; /* find one from the set of registered platforms */ - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (!snd_soc_is_matching_component(dai_link->platform, component)) continue; @@ -1874,7 +1877,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int i; - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { /* does this component override FEs ? */ if (!component->driver->ignore_machine) @@ -3091,6 +3094,7 @@ static void snd_soc_component_add(struct snd_soc_component *component) snd_soc_component_setup_regmap(component); } + /* see for_each_component */ list_add(&component->list, &component_list); INIT_LIST_HEAD(&component->dobj_list); @@ -3226,7 +3230,7 @@ static int __snd_soc_unregister_component(struct device *dev) int found = 0; mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (dev != component->dev) continue; @@ -3258,7 +3262,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, ret = NULL; mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (dev != component->dev) continue; @@ -3658,7 +3662,7 @@ int snd_soc_get_dai_id(struct device_node *ep) */ ret = -ENOTSUPP; mutex_lock(&client_mutex); - list_for_each_entry(pos, &component_list, list) { + for_each_component(pos) { struct device_node *component_of_node = pos->dev->of_node; if (!component_of_node && pos->dev->parent) @@ -3688,7 +3692,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, int ret = -EPROBE_DEFER; mutex_lock(&client_mutex); - list_for_each_entry(pos, &component_list, list) { + for_each_component(pos) { component_of_node = pos->dev->of_node; if (!component_of_node && pos->dev->parent) component_of_node = pos->dev->parent->of_node; -- cgit v1.2.3-58-ga151 From 15a0c64572463eddf59e80aa643d3a87809a7d9b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 21 Sep 2018 05:23:17 +0000 Subject: ASoC: add for_each_component_dais() macro To be more readable code, this patch adds new for_each_component_dais() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 11 ++++++----- 2 files changed, 11 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 93aa894a57ef..f1dab1f4b194 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -864,6 +864,11 @@ struct snd_soc_component { #endif }; +#define for_each_component_dais(component, dai)\ + list_for_each_entry(dai, &(component)->dai_list, list) +#define for_each_component_dais_safe(component, dai, _dai)\ + list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list) + struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b9b33c8cac2e..62e8e36062df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -180,7 +180,7 @@ static int dai_list_show(struct seq_file *m, void *v) mutex_lock(&client_mutex); for_each_component(component) - list_for_each_entry(dai, &component->dai_list, list) + for_each_component_dais(component, dai) seq_printf(m, "%s\n", dai->name); mutex_unlock(&client_mutex); @@ -781,7 +781,7 @@ struct snd_soc_dai *snd_soc_find_dai( for_each_component(component) { if (!snd_soc_is_matching_component(dlc, component)) continue; - list_for_each_entry(dai, &component->dai_list, list) { + for_each_component_dais(component, dai) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) && (!dai->driver->name || strcmp(dai->driver->name, dlc->dai_name))) @@ -1312,7 +1312,7 @@ static int soc_probe_component(struct snd_soc_card *card, } } - list_for_each_entry(dai, &component->dai_list, list) { + for_each_component_dais(component, dai) { ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret != 0) { dev_err(component->dev, @@ -2842,7 +2842,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component) { struct snd_soc_dai *dai, *_dai; - list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { + for_each_component_dais_safe(component, dai, _dai) { dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", dai->name); list_del(&dai->list); @@ -2894,6 +2894,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; + /* see for_each_component_dais */ list_add_tail(&dai->list, &component->dai_list); component->num_dai++; @@ -3728,7 +3729,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, ret = 0; /* find target DAI */ - list_for_each_entry(dai, &pos->dai_list, list) { + for_each_component_dais(pos, dai) { if (id == 0) break; id--; -- cgit v1.2.3-58-ga151 From b0ef5011b981ece1fde8063243a56d3038b87adb Mon Sep 17 00:00:00 2001 From: Matt Flax Date: Tue, 25 Sep 2018 16:40:18 +1000 Subject: ASoC: cs4265: Add a MIC pre. route The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d9eebf6af7a8..ab27d2b94d02 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"LINEOUTR", NULL, "DAC"}, {"SPDIFOUT", NULL, "SPDIF"}, + {"Pre-amp MIC", NULL, "MICL"}, + {"Pre-amp MIC", NULL, "MICR"}, + {"ADC Mux", "MIC", "Pre-amp MIC"}, {"ADC Mux", "LINEIN", "LINEINL"}, {"ADC Mux", "LINEIN", "LINEINR"}, - {"ADC Mux", "MIC", "MICL"}, - {"ADC Mux", "MIC", "MICR"}, {"ADC", NULL, "ADC Mux"}, {"DOUT", NULL, "ADC"}, {"DAI1 Capture", NULL, "DOUT"}, -- cgit v1.2.3-58-ga151 From 85aa0fe73edd856365d074a5aa38c614c8b2ca45 Mon Sep 17 00:00:00 2001 From: Andreas Färber Date: Tue, 25 Sep 2018 16:23:49 +0200 Subject: ASoC: max98088: add OF support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit MAX98088 is an older version of the MAX98089 device. Signed-off-by: Andreas Färber [m.felsch@pengutronix.de: add CONFIG_OF compile switch] [m.felsch@pengutronix.de: adapt commit message] Signed-off-by: Marco Felsch Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index fb515aaa54fc..9450d5d9c492 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1742,9 +1742,19 @@ static const struct i2c_device_id max98088_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id max98088_of_match[] = { + { .compatible = "maxim,max98088" }, + { .compatible = "maxim,max98089" }, + { } +}; +MODULE_DEVICE_TABLE(of, max98088_of_match); +#endif + static struct i2c_driver max98088_i2c_driver = { .driver = { .name = "max98088", + .of_match_table = of_match_ptr(max98088_of_match), }, .probe = max98088_i2c_probe, .id_table = max98088_i2c_id, -- cgit v1.2.3-58-ga151 From 42cfb412e24ffbb46d6de9590293bc44f921a0fb Mon Sep 17 00:00:00 2001 From: Matthias Kaehlcke Date: Tue, 25 Sep 2018 11:09:14 -0700 Subject: ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..e3b9dd634c6d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_dma_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- cgit v1.2.3-58-ga151 From bec5ecdf41d404691980c9c82ba867113cc8dee5 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 26 Sep 2018 14:46:35 +0200 Subject: ASoC: pxa: avoid AC97_BUS build warning Selecting AC97_BUS_NEW from SND_PXA2XX_SOC_AC97 leads to a Kconfig warning if any other driver selects AC97_BUS: WARNING: unmet direct dependencies detected for AC97_BUS_COMPAT Depends on [n]: SOUND [=y] && !UML && SND [=y] && AC97_BUS_NEW [=y] && !AC97_BUS [=y] Selected by [y]: - SND_SOC_WM9713 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && AC97_BUS_NEW [=y] I don't know if that combination is supposed to work. Assuming it is not, this adds a dependency on all users for PXA to avoids the combination. Fixes: 1c8bc7b3de5e ("ASoC: pxa: switch to new ac97 bus support") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 29f577e6dfc0..943b44de1464 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -88,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,6 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,6 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 + depends on !AC97_BUS select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -116,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -126,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,6 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -194,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- cgit v1.2.3-58-ga151 From 53c156ab9d8aae2083d3c895365a3e39b864a5a7 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 26 Sep 2018 14:59:40 +0200 Subject: ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency Selecting SND_SOC_WM8731 is only allowed when all its dependencies are already there: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] Selected by [m]: - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && (ARCH_AT91 [=y] || COMPILE_TEST [=y]) && ATMEL_SSC [=y] && SND_SOC_I2C_AND_SPI [=m] - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && COMPILE_TEST [=y] && SND_SOC_I2C_AND_SPI [=m] Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Arnd Bergmann Reviewed-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 81a0712d4f14..64f86f0b87e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -101,6 +101,7 @@ config SND_ATMEL_SOC_I2S config SND_SOC_MIKROE_PROTO tristate "Support for Mikroe-PROTO board" depends on OF + depends on SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 help Say Y or M if you want to add support for MikroElektronika PROTO Audio -- cgit v1.2.3-58-ga151 From 18380dcc52cc8965e5144ce33fdfad7e168679a5 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 26 Sep 2018 21:37:40 +0200 Subject: ASoC: wm9712: fix unused variable warning The 'ret' variable is now only used in an #ifdef, and causes a warning if it is declared outside of that block: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable] Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..e873baa9e778 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; } else { #ifdef CONFIG_SND_SOC_AC97_BUS + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { -- cgit v1.2.3-58-ga151 From 06da26e5ce157bf2af09ef4a9f5dc30af4599fa0 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 26 Sep 2018 15:33:03 +0800 Subject: ASoC: qcom: qdsp6: remove duplicated include Remove duplicated includes linux/of_platform.h and linux/wait.h Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 2b2c7233bb5f..e1cfa846a1dc 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -11,7 +11,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index ca1be7305524..cdfc8ab6cfc0 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -10,7 +10,6 @@ #include #include #include -#include #include #include "q6core.h" #include "q6dsp-errno.h" -- cgit v1.2.3-58-ga151 From 9c80c5a8831471e0a3e139aad1b0d4c0fdc50b2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Oct 2018 19:31:44 +0200 Subject: ASoC: intel: skylake: Add missing break in skl_tplg_get_token() skl_tplg_get_token() misses a break in the big switch() block for SKL_TKN_U8_CORE_ID entry. Spotted nicely by -Wimplicit-fallthrough compiler option. Fixes: 6277e83292a2 ("ASoC: Intel: Skylake: Parse vendor tokens to build module data") Cc: Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 52a9915da0f5..cf8848b779dc 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2460,6 +2460,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U8_CORE_ID: mconfig->core_id = tkn_elem->value; + break; case SKL_TKN_U8_MOD_TYPE: mconfig->m_type = tkn_elem->value; -- cgit v1.2.3-58-ga151 From 8e9f7265eda9f3a662ca1ca47a69042a7840735b Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 1 Oct 2018 19:44:30 +0300 Subject: ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR() The q6asm_audio_client_alloc() doesn't return NULL, it returns error pointers. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c3806d7037fc..a16c71c03058 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -318,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { pr_info("%s: Could not allocate memory\n", __func__); + ret = PTR_ERR(prtd->audio_client); kfree(prtd); - return -ENOMEM; + return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -- cgit v1.2.3-58-ga151 From cfe9ee5f2b7808ceec21df83a45e38afcc647153 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Oct 2018 21:36:27 +0200 Subject: ASoC: pxa-ssp: enable and disable extclk if given If a "extclk" clock is given, enable and disable it when appropriate. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..adcf8ba9d287 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } + if (priv->extclk) + clk_prepare_enable(priv->extclk); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; @@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } + if (priv->extclk) + clk_disable_unprepare(priv->extclk); + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } -- cgit v1.2.3-58-ga151 From 466dee75b3364d05b43ddfe41ef2e8887b6a3ea7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Oct 2018 21:32:34 +0200 Subject: ASoC: add fault detect recovery property to DT bindings The driver already has support for setting the FDRB bit in the CONFA register through platform data, but there was no property to set it in the device-tree bindings. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/st,sta32x.txt | 3 +++ sound/soc/codecs/sta32x.c | 2 ++ 2 files changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index 255de3ae5b2f..ff4a685a4303 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -39,6 +39,9 @@ Optional properties: - st,thermal-warning-recover: If present, thermal warning recovery is enabled. + - st,fault-detect-recovery: + If present, fault detect recovery is enabled. + - st,thermal-warning-adjustment: If present, thermal warning adjustment is enabled. diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index d5035f2f2b2b..22de1593443c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1038,6 +1038,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x) of_property_read_u8(np, "st,ch3-output-mapping", &pdata->ch3_output_mapping); + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; if (of_get_property(np, "st,thermal-warning-recovery", NULL)) pdata->thermal_warning_recovery = 1; if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) -- cgit v1.2.3-58-ga151 From 7e29317928bd79a03a9c35816afa709988b5d31b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Oct 2018 20:30:02 +0200 Subject: ASoC: adau1761: Use the standard fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation at the right place. It has to be put right before the next label. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e981653..bef3e9e74c26 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- cgit v1.2.3-58-ga151 From 641f7f2195735b4fe93b541ea3a792fe4fee2415 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Oct 2018 20:30:03 +0200 Subject: ASoC: pcm186x: Use the standard fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation. Unfortunately gcc doesn't understand the mixed comment lines. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/pcm186x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 690c26e7389e..809b7e9f03ca 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format) break; case SND_SOC_DAIFMT_DSP_A: priv->tdm_offset += 1; - /* Fall through... DSP_A uses the same basic config as DSP_B + /* fall through */ + /* DSP_A uses the same basic config as DSP_B * except we need to shift the TDM output by one BCK cycle */ case SND_SOC_DAIFMT_DSP_B: -- cgit v1.2.3-58-ga151 From 0beeb4baf56bd9deb920712a4034541fb33bbbe0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Oct 2018 20:30:04 +0200 Subject: ASoC: rt274: Add fall-through annotations As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, add the "fall through" annotations in rt274 driver. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index d88e67341083..0ef966d56bac 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid pll source, use BCLK\n"); + /* fall through */ case RT274_PLL2_S_BCLK: snd_soc_component_update_bits(component, RT274_PLL2_CTRL, RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); @@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid freq_in, assume 4.8M\n"); + /* fall through */ case 100: snd_soc_component_write(component, 0x7a, 0xaab6); snd_soc_component_write(component, 0x7b, 0x0301); -- cgit v1.2.3-58-ga151 From e4bfd61571f5db5c69a7a49de401543cc7d6c87c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Oct 2018 20:30:05 +0200 Subject: ASoC: intel: skylake: Add fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, add the "fall through" annotation in Intel SST skylake driver. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 00b7a91b18c9..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -495,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -- cgit v1.2.3-58-ga151 From 9c6c4d961e634413add345ee030e108e6d19cea2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Oct 2018 20:30:06 +0200 Subject: ASoC: topology: Use the standard fall-through annotations As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation. gcc can't understand the mixed texts, unfortunately. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 17f81b9a5754..045ef136903d 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -993,7 +993,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, kfree(se); continue; } - /* fall through and create texts */ + /* fall through */ case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: @@ -1310,7 +1310,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); goto err_se; } - /* fall through to create texts */ + /* fall through */ case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: -- cgit v1.2.3-58-ga151 From 7454a21c13f7ce9bf1a4f9b639039b78462cec09 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Oct 2018 21:34:36 +0200 Subject: ASoC: wm8782: add support for regulators Lookup regulators for Vdd and Vdda during probe, and enable them when the component is linked. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/wm8782.c | 63 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index 317db9a149a7..cf2cdbece122 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; +/* regulator power supply names */ +static const char *supply_names[] = { + "Vdda", /* analog supply, 2.7V - 3.6V */ + "Vdd", /* digital supply, 2.7V - 5.5V */ +}; + +struct wm8782_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; +}; + +static int wm8782_soc_probe(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static void wm8782_soc_remove(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +#ifdef CONFIG_PM +static int wm8782_soc_suspend(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int wm8782_soc_resume(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define wm8782_soc_suspend NULL +#define wm8782_soc_resume NULL +#endif /* CONFIG_PM */ + static const struct snd_soc_component_driver soc_component_dev_wm8782 = { + .probe = wm8782_soc_probe, + .remove = wm8782_soc_remove, + .suspend = wm8782_soc_suspend, + .resume = wm8782_soc_resume, .dapm_widgets = wm8782_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), .dapm_routes = wm8782_dapm_routes, @@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = { static int wm8782_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct wm8782_priv *priv; + int ret, i; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(dev, priv); + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + return devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_wm8782, &wm8782_dai, 1); } -- cgit v1.2.3-58-ga151 From 62a7fc32a6289dce88787da03f893deab08158c3 Mon Sep 17 00:00:00 2001 From: Andreas Färber Date: Fri, 5 Oct 2018 09:58:11 +0200 Subject: ASoC: max98088: Add master clock handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If master clock is provided through device tree, then update the master clock frequency during set_sysclk. Cc: Tushar Behera Signed-off-by: Andreas Färber Acked-by: Tushar Behera Reviewed-by: Javier Martinez Canillas [m.felsch@pengutronix.de: move mclk request to i2c_probe] [m.felsch@pengutronix.de: make use of snd_soc_component_get_bias_level()] Signed-off-by: Marco Felsch Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 9450d5d9c492..ca172a4b6849 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -42,6 +43,7 @@ struct max98088_priv { struct regmap *regmap; enum max98088_type devtype; struct max98088_pdata *pdata; + struct clk *mclk; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98088->sysclk) return 0; + if (!IS_ERR(max98088->mclk)) { + freq = clk_round_rate(max98088->mclk, freq); + clk_set_rate(max98088->mclk, freq); + } + /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 30MHz).. @@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_PREPARE: + /* + * SND_SOC_BIAS_PREPARE is called while preparing for a + * transition to ON or away from ON. If current bias_level + * is SND_SOC_BIAS_ON, then it is preparing for a transition + * away from ON. Disable the clock in that case, otherwise + * enable it. + */ + if (!IS_ERR(max98088->mclk)) { + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_ON) + clk_disable_unprepare(max98088->mclk); + else + clk_prepare_enable(max98088->mclk); + } break; case SND_SOC_BIAS_STANDBY: @@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (IS_ERR(max98088->regmap)) return PTR_ERR(max98088->regmap); + max98088->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(max98088->mclk)) + if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER) + return PTR_ERR(max98088->mclk); + max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); -- cgit v1.2.3-58-ga151 From 24ae67c5825004bcbce90e7c89fed63f25d96260 Mon Sep 17 00:00:00 2001 From: Marco Felsch Date: Fri, 5 Oct 2018 09:58:12 +0200 Subject: ASoC: max98988: make it selectable Currently the driver will build only if SND_SOC_ALL_CODECS is set. Adding a Kconfig menu description to build the driver standalone. Signed-off-by: Marco Felsch Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9989d35e0fc6..3c6bd6019b92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -640,7 +640,7 @@ config SND_SOC_LM49453 tristate config SND_SOC_MAX98088 - tristate + tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" config SND_SOC_MAX98090 tristate -- cgit v1.2.3-58-ga151 From 9641faa2db7e856f50a6d1169e1b9f01e7fcb2b0 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 10 Oct 2018 10:37:13 +0200 Subject: ASoC: max98988: add I2C dependency max98988 only builds with I2C support enabled, otherwise we get a build error: sound/soc/codecs/max98088.c:1789:1: error: data definition has no type or storage class [-Werror] module_i2c_driver(max98088_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/max98088.c:1789:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/max98088.c:1789:1: error: parameter names (without types) in function declaration [-Werror] sound/soc/codecs/max98088.c:1780:26: error: 'max98088_i2c_driver' defined but not used [-Werror=unused-variable] Fixes: 24ae67c58250 ("ASoC: max98988: make it selectable") Signed-off-by: Arnd Bergmann Reviewed-by: Marco Felsch Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c6bd6019b92..774d38310875 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -641,6 +641,7 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" + depends on I2C config SND_SOC_MAX98090 tristate -- cgit v1.2.3-58-ga151 From 82ab7e9a4d3fcec27f745be04063e17da1881dda Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 10 Oct 2018 02:20:42 +0000 Subject: ASoC: rsnd: use 32bit TDM width as default commit fb2815f44a9e ("ASoC: rsnd: add support for 16/24 bit slot widths") added TDM width check, and return error if it was not 16/24/32 bit. But it is too strict. This patch uses 32bit same as default. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..f930f51b686f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -744,8 +744,8 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, case 32: break; default: - dev_err(dev, "unsupported slot width value: %d\n", slot_width); - return -EINVAL; + /* use default */ + slot_width = 32; } switch (slots) { -- cgit v1.2.3-58-ga151 From 8036dbc490d16dc3d998246e14c9507ec8272ae2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 10 Oct 2018 02:22:29 +0000 Subject: ASoC: audio-graph-card: enable mclk-fs on codec node Current audio-graph-card is supporting mclk-fs on CPU node side only. But having Codec node also is good idea. It will be just ignored if not defined. "rcpu_ep" is same as "cpu_ep", This patch tidyup it, too. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fb6635f8d5d7..25c819e402e1 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -182,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, if (ret < 0) goto dai_link_of_err; - of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) -- cgit v1.2.3-58-ga151 From 4cbbc91609846c09a8350080cd7e6f7764fb2ec1 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 10 Oct 2018 23:26:06 +0000 Subject: ASoC: max98373: Sort Digital Volume in reverse order Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index d6868c9a9ce6..9b7ccd351cae 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -455,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"), SND_SOC_DAPM_SIGGEN("FBMON"), }; -static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0); +static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1); static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv, 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), @@ -605,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG, SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0), SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0, 0x7F, 0, max98373_digital_tlv), + 0, 0x7F, 1, max98373_digital_tlv), SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN, MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv), SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN, -- cgit v1.2.3-58-ga151 From 6c3beeca424a0c8d6c79184a880a8954bd498d57 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 10 Oct 2018 23:26:10 +0000 Subject: ASoC: max98373: Sort BDE Limiter Thresh Volume in reverse order Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 9b7ccd351cae..38871677cbae 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -479,7 +479,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv, 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv, - 0, 15, TLV_DB_SCALE_ITEM(0, -100, 0), + 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, @@ -670,13 +670,13 @@ SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3, SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3, 0, 0x3C, 0, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), /* Limiter */ SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN, MAX98373_LIMITER_EN_SHIFT, 1, 0), -- cgit v1.2.3-58-ga151 From d34c8f37c75b739efc26383145a43497143ada88 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 10 Oct 2018 23:26:13 +0000 Subject: ASoC: max98373: Sort max98373_bde_gain_tlv in reverse order Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 38871677cbae..37f8bcdd8e35 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -483,7 +483,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv, ); static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, - 0, 60, TLV_DB_SCALE_ITEM(0, -25, 0), + 0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0), ); static bool max98373_readable_register(struct device *dev, unsigned int reg) @@ -654,21 +654,21 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0), SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0), SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1, 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1, -- cgit v1.2.3-58-ga151 From a23f5dc8448694a0ffe2127a04aa5787b9cf9e5f Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 10 Oct 2018 23:26:17 +0000 Subject: ASoC: max98373: Sort DHT Rot Pnt Volume in reverse order Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 37f8bcdd8e35..a09d01318f79 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -471,12 +471,12 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv, 0, 9, TLV_DB_SCALE_ITEM(800, 100, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv, - 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), - 2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0), - 8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0), - 10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0), - 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), - 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0), + 2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0), + 7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0), + 10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0), + 14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv, 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), @@ -617,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN, SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG, MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv), SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG, - MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv), + MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv), SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG, MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv), SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG, -- cgit v1.2.3-58-ga151 From 747df19747bc9752cd40b9cce761e17a033aa5c2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 11 Oct 2018 20:32:05 +0200 Subject: ASoC: sta32x: set ->component pointer in private struct The ESD watchdog code in sta32x_watchdog() dereferences the pointer which is never assigned. This is a regression from a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") which went unnoticed since nobody seems to use that ESD workaround. Fixes: a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") Signed-off-by: Daniel Mack Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sta32x.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index d5035f2f2b2b..ce508b4cc85c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -879,6 +879,9 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { -- cgit v1.2.3-58-ga151 From 3809688980205622f75ed5d5890759430da4e7e4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Oct 2018 06:31:00 +0000 Subject: ASoC: pcm3168a: add HW constraint for non RIGHT_J RIGHT_J only can handle 16bit data bits. Current driver just errored if user requests non RIGHT_J + 16bit combination. But it is not useful for user. This patch adds HW constraint for it, and avoid error on such situation. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 3356c91f55b0..233a8df5d7a5 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -476,7 +476,43 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return 0; } +static int pcm3168a_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt; + unsigned int sample_min; + + if (tx) + fmt = pcm3168a->dac_fmt; + else + fmt = pcm3168a->adc_fmt; + + /* + * Available Data Bits + * + * RIGHT_J : 24 / 16 + * LEFT_J : 24 + * I2S : 24 + */ + switch (fmt) { + case PCM3168A_FMT_RIGHT_J: + sample_min = 16; + break; + default: + sample_min = 24; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + sample_min, 32); + + return 0; +} static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_dac, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, -- cgit v1.2.3-58-ga151 From 594680ea4a394f19d38a39a7d7c673fbad02a3d6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Oct 2018 06:31:18 +0000 Subject: ASoC: pcm3168a: add hw constraint for channel LEFT_J / I2S only can use TDM. This patch adds channel constraint for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 233a8df5d7a5..f0e2b886323e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -484,6 +484,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int fmt; unsigned int sample_min; + unsigned int channel_max; if (tx) fmt = pcm3168a->dac_fmt; @@ -496,19 +497,38 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, * RIGHT_J : 24 / 16 * LEFT_J : 24 * I2S : 24 + * + * TDM available + * + * I2S + * LEFT_J */ switch (fmt) { case PCM3168A_FMT_RIGHT_J: sample_min = 16; + channel_max = 2; + break; + case PCM3168A_FMT_LEFT_J: + sample_min = 24; + channel_max = 8; + break; + case PCM3168A_FMT_I2S: + sample_min = 24; + channel_max = 8; break; default: sample_min = 24; + channel_max = 2; } snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, sample_min, 32); + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, channel_max); + return 0; } static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { -- cgit v1.2.3-58-ga151 From 471a7ba89158c6d52dae69636c94c4aa1a6b7b22 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Oct 2018 06:31:49 +0000 Subject: ASoC: pcm3168a: add I2S/Left_J TDM support pcm3168a is supporting TDM on I2S/Left_J, but there is no settings for it. This patch add it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index f0e2b886323e..63aa02592bc0 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -33,6 +33,8 @@ #define PCM3168A_FMT_RIGHT_J_16 0x3 #define PCM3168A_FMT_DSP_A 0x4 #define PCM3168A_FMT_DSP_B 0x5 +#define PCM3168A_FMT_I2S_TDM 0x6 +#define PCM3168A_FMT_LEFT_J_TDM 0x7 #define PCM3168A_FMT_DSP_MASK 0x4 #define PCM3168A_NUM_SUPPLIES 6 @@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, bool tx, master_mode; u32 val, mask, shift, reg; unsigned int rate, fmt, ratio, max_ratio; + unsigned int chan; int i, min_frame_size; rate = params_rate(params); + chan = params_channels(params); ratio = pcm3168a->sysclk / rate; @@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* for TDM */ + if (chan > 2) { + switch (fmt) { + case PCM3168A_FMT_I2S: + fmt = PCM3168A_FMT_I2S_TDM; + break; + case PCM3168A_FMT_LEFT_J: + fmt = PCM3168A_FMT_LEFT_J_TDM; + break; + default: + dev_err(component->dev, "TDM is supported under I2S/Left_J only\n"); + return -EINVAL; + } + } + if (master_mode) val = ((i + 1) << shift); else -- cgit v1.2.3-58-ga151 From 2657c6a9037df1c98eb5dae10dd554ac8c37d38f Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 15 Oct 2018 12:24:44 +0530 Subject: ASoC: AMD: Add SND_JACK_LINEOUT jack type Some 3 pole connectors report impedance greater than threshold of 1000Ohm. Thus, da7219 reports them as LINEOUT. Adding the SND_JACK_LINEOUT type so that we don't fail to detect any 3 pole jack type. Also, changing SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 717a017f0db6..3f813ea5210a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); ret = snd_soc_card_jack_new(card, "Headset Jack", - SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &cz_jack, NULL, 0); -- cgit v1.2.3-58-ga151 From bca0ac1d96739c07ee1a158e4b1202260ad7480e Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Tue, 9 Oct 2018 15:35:47 +0800 Subject: ASoC: Intel: Boards: Add KBL Dialog Maxim I2S machine driver This patch adds Kabylake I2S machine driver with: DA7219 audio codec(SSP1) and MAXIM98927(SSP0) speaker amplifier. Signed-off-by: Mac Chiang Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 13 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_da7219_max98927.c | 983 +++++++++++++++++++++++++++ 3 files changed, 998 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 88e4b4284738..73ca1350aa31 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -280,6 +280,19 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. +config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH + tristate "KBL with DA7219 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_DA7219 + select SND_SOC_MAX98927 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98927 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6e88373cbe35..5381e27df9cc 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -17,6 +17,7 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o +snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c new file mode 100644 index 000000000000..3ab96ee7bd3c --- /dev/null +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -0,0 +1,983 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAX98927 and + * RT5663 codecs + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/da7219.h" +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "../../codecs/da7219-aad.h" + +#define KBL_DIALOG_CODEC_DAI "da7219-hifi" +#define MAX98927_CODEC_DAI "max98927-aif1" +#define MAXIM_DEV0_NAME "i2c-MX98927:00" +#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack kabylake_hdmi[3]; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct snd_soc_jack kabylake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_ECHO_REF_CP, + KBL_DPCM_AUDIO_REF_CP, + KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, + KBL_DPCM_AUDIO_HS_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_kcontrol_new kabylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget kabylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route kabylake_map[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, + + /* other jacks */ + { "DMic", NULL, "SoC DMIC" }, + + { "HDMI", NULL, "hif5 Output" }, + { "DP", NULL, "hif6 Output" }, + + /* CODEC BE connections */ + { "Left HiFi Playback", NULL, "ssp0 Tx" }, + { "Right HiFi Playback", NULL, "ssp0 Tx" }, + { "ssp0 Tx", NULL, "spk_out" }, + + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + + /* AEC capture path */ + { "echo_ref_out", NULL, "ssp0 Rx" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, + + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, +}; + +static const struct snd_soc_dapm_route kabylake_ssp1_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, + + /* CODEC BE connections */ + { "Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec1_out" }, + + { "hs_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "Capture" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + int ret = 0, j; + + for (j = 0; j < runtime->num_codecs; j++) { + struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static struct snd_soc_ops kabylake_ssp0_ops = { + .hw_params = kabylake_ssp0_hw_params, +}; + +static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_soc_dpcm *dpcm = container_of( + params, struct snd_soc_dpcm, hw_params); + struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; + struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + + /* + * The ADSP will convert the FE rate to 48k, stereo, 24 bit + */ + if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + } + + /* + * The speaker on the SSP0 supports S16_LE and not S24_LE. + * thus changing the mask here + */ + if (!strcmp(be_dai_link->name, "SSP0-Codec")) + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = rtd->card; + int ret; + + + ret = snd_soc_dapm_add_routes(&card->dapm, + kabylake_ssp1_map, + ARRAY_SIZE(kabylake_ssp1_map)); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + da7219_aad_jack_det(component, &ctx->kabylake_headset); + + ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); + if (ret) + dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); + + return ret; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_component *component = rtd->cpu_dai->component; + + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + + return 0; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_da7219_fe_ops = { + .startup = kbl_fe_startup, +}; + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static int kabylake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static struct snd_soc_ops kabylake_dmic_ops = { + .startup = kabylake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static const unsigned int ch_mono[] = { + 1, +}; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + + +static struct snd_soc_ops skylaye_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + +static struct snd_soc_codec_conf max98927_codec_conf[] = { + + { + .dev_name = MAXIM_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAXIM_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_dai_link_component ssp0_codec_components[] = { + { /* Left */ + .name = MAXIM_DEV0_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + + { /* For Right */ + .name = MAXIM_DEV1_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HS_PB] = { + .name = "Kbl Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .ops = &kabylake_da7219_fe_ops, + + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 1, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-DLGS7219:00", + .codec_dai_name = KBL_DIALOG_CODEC_DAI, + .init = kabylake_da7219_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_max98927_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 2, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 3, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 4, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &kabylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &kabylake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); + + return 0; +} + +/* kabylake audio machine driver for SPT + DA7219 */ +static struct snd_soc_card kbl_audio_card_da7219_m98927 = { + .name = "kblda7219m98927", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +/* kabylake audio machine driver for Maxim98927 */ +static struct snd_soc_card kbl_audio_card_max98927 = { + .name = "kblmax98927", + .owner = THIS_MODULE, + .dai_link = kabylake_max98927_dais, + .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_da7219_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98927, + }, + { + .name = "kbl_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98927, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_da7219_max98927", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_AUTHOR("Mac Chiang "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_da7219_max98927"); +MODULE_ALIAS("platform:kbl_max98927"); -- cgit v1.2.3-58-ga151 From 6530adeaaf5018cd13437dc82adcd9349657a00e Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Tue, 9 Oct 2018 15:37:08 +0800 Subject: ASoC: Intel: common: Add Kabylake Dialog+Maxim machine driver entry This patch adds da7219_max98927 machine driver entry into machine table Signed-off-by: Mac Chiang Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 0ee173ca437d..a317b7790fce 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { + .num_codecs = 1, + .codecs = {"MX98927"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98357_codecs, .pdata = &skl_dmic_data, }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98927_codecs, + .pdata = &skl_dmic_data + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- cgit v1.2.3-58-ga151 From 3b991038498bc5011b063d6a804503c577a79434 Mon Sep 17 00:00:00 2001 From: Christoph Hellwig Date: Sat, 13 Oct 2018 17:17:04 +0200 Subject: ASoC: intel: don't pass GFP_DMA32 to dma_alloc_coherent The DMA API does its own zone decisions based on the coherent_dma_mask. Signed-off-by: Christoph Hellwig Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/common/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 11041aedea31..1e067504b604 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, /* allocate DMA buffer to store FW data */ sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size, - &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL); + &sst_fw->dmable_fw_paddr, GFP_KERNEL); if (!sst_fw->dma_buf) { dev_err(dsp->dev, "error: DMA alloc failed\n"); kfree(sst_fw); -- cgit v1.2.3-58-ga151 From 66ecce3325383c8304063e7d5a30f4374ef5a33e Mon Sep 17 00:00:00 2001 From: Marcus Cooper Date: Wed, 17 Oct 2018 00:38:05 -0700 Subject: ASoC: sun4i-i2s: Add compatibility with A64 codec I2S The I2S block used for the audio codec in the A64 differs from other 3 I2S modules in A64 and isn't compatible with H3. But it is very similar to what is found in A10(sun4i). However, its TX FIFO is located at a different address. Signed-off-by: Marcus Cooper Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sun4i-i2s.txt | 2 ++ sound/soc/sunxi/sun4i-i2s.c | 21 +++++++++++++++++++++ 2 files changed, 23 insertions(+) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt index b9d50d6cdef3..61e71c1729e0 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt @@ -10,6 +10,7 @@ Required properties: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. @@ -26,6 +27,7 @@ Required properties for the following compatibles: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - resets: phandle to the reset line for this codec Example: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..c63d226e2436 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -961,6 +961,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { .field_rxchansel = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2), }; +static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .has_slave_select_bit = true, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5), + .field_fmt_bclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6), + .field_fmt_lrclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7), + .field_fmt_mode = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1), + .field_txchanmap = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31), + .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31), + .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2), + .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2), +}; + static int sun4i_i2s_init_regmap_fields(struct device *dev, struct sun4i_i2s *i2s) { @@ -1169,6 +1186,10 @@ static const struct of_device_id sun4i_i2s_match[] = { .compatible = "allwinner,sun8i-h3-i2s", .data = &sun8i_h3_i2s_quirks, }, + { + .compatible = "allwinner,sun50i-a64-codec-i2s", + .data = &sun50i_a64_codec_i2s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_i2s_match); -- cgit v1.2.3-58-ga151 From 13c3bf174becfb8b55adcfeb6f01724dc99347f0 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Wed, 17 Oct 2018 00:38:06 -0700 Subject: ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio BCLK / LRCK ratio should be sample size * channels, but it was hardcoded to 32 (0x1 is 32 as per A33 and A64 datasheets). Calculate it basing on sample size and number of channels. Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index fb37dd927e33..522a72fde78d 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include @@ -52,7 +53,6 @@ #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV 13 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV 9 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV 6 -#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16 (1 << 6) #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ 4 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16 (1 << 4) #define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT 2 @@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec, return best_val; } +static int sun8i_codec_get_lrck_div(unsigned int channels, + unsigned int word_size) +{ + unsigned int div = word_size * channels; + + if (div < 16 || div > 256) + return -EINVAL; + + return ilog2(div) - 4; +} + static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component); - int sample_rate; + int sample_rate, lrck_div; u8 bclk_div; /* @@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK, bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV); + lrck_div = sun8i_codec_get_lrck_div(params_channels(params), + params_physical_width(params)); + if (lrck_div < 0) + return lrck_div; + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK, - SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16); + lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV); sample_rate = sun8i_codec_get_hw_rate(params); if (sample_rate < 0) -- cgit v1.2.3-58-ga151 From 55b407f6468c481edc4a26b21c9f567ae57f01a4 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Wed, 17 Oct 2018 00:38:07 -0700 Subject: ASoC: sun8i-codec-analog: split regmap code into separate driver It will be reused by sun50i-codec-analog later. Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/Kconfig | 7 ++- sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 +++++++++++++++++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 +++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +------------------------ 5 files changed, 119 insertions(+), 77 deletions(-) create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h (limited to 'sound/soc') diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 22408bc2d6ec..83b770cdfdaa 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -23,7 +23,7 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. @@ -45,4 +45,9 @@ config SND_SUN4I_SPDIF help Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. + +config SND_SUN8I_ADDA_PR_REGMAP + tristate + select REGMAP + endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 4a9ef67386ca..74b99d55cfca 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -4,3 +4,4 @@ obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o +obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c new file mode 100644 index 000000000000..e68ce9d2884d --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver provides regmap to access to analog part of audio codec + * found on Allwinner A23, A31s, A33, H3 and A64 Socs + * + * Copyright 2016 Chen-Yu Tsai + * Copyright (C) 2018 Vasily Khoruzhick + */ + +#include +#include +#include +#include + +#include "sun8i-adda-pr-regmap.h" + +/* Analog control register access bits */ +#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ +#define ADDA_PR_RESET BIT(28) +#define ADDA_PR_WRITE BIT(24) +#define ADDA_PR_ADDR_SHIFT 16 +#define ADDA_PR_ADDR_MASK GENMASK(4, 0) +#define ADDA_PR_DATA_IN_SHIFT 8 +#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) +#define ADDA_PR_DATA_OUT_SHIFT 0 +#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) + +/* regmap access bits */ +static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Read back value */ + *val = readl(base) & ADDA_PR_DATA_OUT_MASK; + + return 0; +} + +static int adda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Set data to write */ + tmp = readl(base); + tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); + tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; + writel(tmp, base); + + /* Set write bit to signal a write */ + writel(readl(base) | ADDA_PR_WRITE, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + return 0; +} + +static const struct regmap_config adda_pr_regmap_cfg = { + .name = "adda-pr", + .reg_bits = 5, + .reg_stride = 1, + .val_bits = 8, + .reg_read = adda_reg_read, + .reg_write = adda_reg_write, + .fast_io = true, + .max_register = 31, +}; + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base) +{ + return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg); +} +EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init); + +MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver"); +MODULE_AUTHOR("Vasily Khoruzhick "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-adda-pr"); diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h new file mode 100644 index 000000000000..a5ae95dfebc1 --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2018 Vasily Khoruzhick + */ + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base); diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 485e79f292c4..916a46bbc1c8 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -27,6 +27,8 @@ #include #include +#include "sun8i-adda-pr-regmap.h" + /* Codec analog control register offsets and bit fields */ #define SUN8I_ADDA_HP_VOLC 0x00 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7 @@ -120,81 +122,6 @@ #define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6 #define SUN8I_ADDA_ADC_AP_EN_ADCG 0 -/* Analog control register access bits */ -#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ -#define ADDA_PR_RESET BIT(28) -#define ADDA_PR_WRITE BIT(24) -#define ADDA_PR_ADDR_SHIFT 16 -#define ADDA_PR_ADDR_MASK GENMASK(4, 0) -#define ADDA_PR_DATA_IN_SHIFT 8 -#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) -#define ADDA_PR_DATA_OUT_SHIFT 0 -#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) - -/* regmap access bits */ -static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Read back value */ - *val = readl(base) & ADDA_PR_DATA_OUT_MASK; - - return 0; -} - -static int adda_reg_write(void *context, unsigned int reg, unsigned int val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Set data to write */ - tmp = readl(base); - tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); - tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; - writel(tmp, base); - - /* Set write bit to signal a write */ - writel(readl(base) | ADDA_PR_WRITE, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - return 0; -} - -static const struct regmap_config adda_pr_regmap_cfg = { - .name = "adda-pr", - .reg_bits = 5, - .reg_stride = 1, - .val_bits = 8, - .reg_read = adda_reg_read, - .reg_write = adda_reg_write, - .fast_io = true, - .max_register = 24, -}; - /* mixer controls */ static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { SOC_DAPM_DOUBLE_R("DAC Playback Switch", @@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev) return PTR_ERR(base); } - regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg); + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); if (IS_ERR(regmap)) { dev_err(&pdev->dev, "Failed to create regmap\n"); return PTR_ERR(regmap); -- cgit v1.2.3-58-ga151 From 42371f327df0b8e9d479b929c5cd301846dd0f70 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Wed, 17 Oct 2018 00:38:09 -0700 Subject: ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls The internal codec on A64 is split into 2 parts. The analog path controls are routed through an embedded custom register bus accessed through the PRCM block. Add an ASoC component driver for it. This should be tied to the codec audio card as an auxiliary device. Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++++++++++++++++++ 3 files changed, 453 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c (limited to 'sound/soc') diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 83b770cdfdaa..8a055ca1819a 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -28,6 +28,14 @@ config SND_SUN8I_CODEC_ANALOG Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. +config SND_SUN50I_CODEC_ANALOG + tristate "Allwinner sun50i Codec Analog Controls Support" + depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST + select SND_SUNXI_ADDA_PR_REGMAP + help + Say Y or M if you want to add support for the analog controls for + the codec embedded in Allwinner A64 SoC. + config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 74b99d55cfca..a86be340a076 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -3,5 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o +obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c new file mode 100644 index 000000000000..8f5f999df631 --- /dev/null +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -0,0 +1,444 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver supports the analog controls for the internal codec + * found in Allwinner's A64 SoC. + * + * Copyright (C) 2016 Chen-Yu Tsai + * Copyright (C) 2017 Marcus Cooper + * Copyright (C) 2018 Vasily Khoruzhick + * + * Based on sun8i-codec-analog.c + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "sun8i-adda-pr-regmap.h" + +/* Codec analog control register offsets and bit fields */ +#define SUN50I_ADDA_HP_CTRL 0x00 +#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE 7 +#define SUN50I_ADDA_HP_CTRL_HPPA_EN 6 +#define SUN50I_ADDA_HP_CTRL_HPVOL 0 + +#define SUN50I_ADDA_OL_MIX_CTRL 0x01 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN 3 +#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL 2 +#define SUN50I_ADDA_OL_MIX_CTRL_DACL 1 +#define SUN50I_ADDA_OL_MIX_CTRL_DACR 0 + +#define SUN50I_ADDA_OR_MIX_CTRL 0x02 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP 3 +#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR 2 +#define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 +#define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 + +#define SUN50I_ADDA_LINEOUT_CTRL0 0x05 +#define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 +#define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 +#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL 5 +#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL 4 + +#define SUN50I_ADDA_LINEOUT_CTRL1 0x06 +#define SUN50I_ADDA_LINEOUT_CTRL1_VOL 0 + +#define SUN50I_ADDA_MIC1_CTRL 0x07 +#define SUN50I_ADDA_MIC1_CTRL_MIC1G 4 +#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN 3 +#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST 0 + +#define SUN50I_ADDA_MIC2_CTRL 0x08 +#define SUN50I_ADDA_MIC2_CTRL_MIC2G 4 +#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN 3 +#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST 0 + +#define SUN50I_ADDA_LINEIN_CTRL 0x09 +#define SUN50I_ADDA_LINEIN_CTRL_LINEING 0 + +#define SUN50I_ADDA_MIX_DAC_CTRL 0x0a +#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN 7 +#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN 6 +#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN 5 +#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN 4 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE 3 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE 2 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS 1 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS 0 + +#define SUN50I_ADDA_L_ADCMIX_SRC 0x0b +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN 3 +#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL 2 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL 1 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR 0 + +#define SUN50I_ADDA_R_ADCMIX_SRC 0x0c +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP 3 +#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR 2 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR 1 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL 0 + +#define SUN50I_ADDA_ADC_CTRL 0x0d +#define SUN50I_ADDA_ADC_CTRL_ADCREN 7 +#define SUN50I_ADDA_ADC_CTRL_ADCLEN 6 +#define SUN50I_ADDA_ADC_CTRL_ADCG 0 + +#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e +#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 + +#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 + +/* mixer controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0), +}; + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1); + +static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); + + +/* volume / mute controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0, + sun50i_codec_hp_vol_scale), + + SOC_DOUBLE("Headphone Playback Switch", + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE, + SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* ADC */ + SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0, + sun50i_codec_out_mixer_pregain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL, + SUN50I_ADDA_LINEIN_CTRL_LINEING, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN50I_ADDA_LINEOUT_CTRL1, + SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0, + sun50i_codec_lineout_vol_scale), + + SOC_DOUBLE("Line Out Playback Switch", + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LEN, + SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + +}; + +static const char * const sun50i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum, + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPIS, + SUN50I_ADDA_MIX_DAC_CTRL_RHPIS, + sun50i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun50i_codec_hp_src_enum), +}; + +static const char * const sun50i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum, + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL, + SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL, + sun50i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun50i_codec_lineout_src_enum), +}; + +static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0), + /* ADC */ + SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0), + /* + * Due to this component and the codec belonging to separate DAPM + * contexts, we need to manually link the above widgets to their + * stream widgets at the card level. + */ + + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HP"), + + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL, + SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0), + + /* Microphone input */ + SND_SOC_DAPM_INPUT("MIC2"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL, + SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Line input */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), +}; + +static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* ADC Routes */ + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + /* Headphone Routes */ + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "HP", NULL, "Headphone Amp" }, + + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + + /* Microphone Routes */ + { "Mic2 Amplifier", NULL, "MIC2"}, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Line-in Routes */ + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + + /* Line-out Routes */ + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", + "Right Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, +}; + +static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { + .controls = sun50i_a64_codec_controls, + .num_controls = ARRAY_SIZE(sun50i_a64_codec_controls), + .dapm_widgets = sun50i_a64_codec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets), + .dapm_routes = sun50i_a64_codec_routes, + .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes), +}; + +static const struct of_device_id sun50i_codec_analog_of_match[] = { + { + .compatible = "allwinner,sun50i-a64-codec-analog", + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match); + +static int sun50i_codec_analog_probe(struct platform_device *pdev) +{ + struct resource *res; + struct regmap *regmap; + void __iomem *base; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); + if (IS_ERR(regmap)) { + dev_err(&pdev->dev, "Failed to create regmap\n"); + return PTR_ERR(regmap); + } + + return devm_snd_soc_register_component(&pdev->dev, + &sun50i_codec_analog_cmpnt_drv, + NULL, 0); +} + +static struct platform_driver sun50i_codec_analog_driver = { + .driver = { + .name = "sun50i-codec-analog", + .of_match_table = sun50i_codec_analog_of_match, + }, + .probe = sun50i_codec_analog_probe, +}; +module_platform_driver(sun50i_codec_analog_driver); + +MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64"); +MODULE_AUTHOR("Vasily Khoruzhick "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-codec-analog"); -- cgit v1.2.3-58-ga151 From 7e95aac96b554193760aaeb64e263da58127bf27 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Wed, 17 Oct 2018 00:38:10 -0700 Subject: ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 Allwinner A64 uses the same digital codec part as in A33, so we need to build this driver on ARM64 as well. Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 8a055ca1819a..66aad0d3f9c7 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -12,7 +12,7 @@ config SND_SUN4I_CODEC config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF - depends on MACH_SUN8I || COMPILE_TEST + depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST select REGMAP_MMIO help This option enables the digital part of the internal audio codec for -- cgit v1.2.3-58-ga151 From a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler Acked-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- cgit v1.2.3-58-ga151 From 1e3cb6c321be2e5295dcaa94c2bf42a43a47a067 Mon Sep 17 00:00:00 2001 From: David Lin Date: Fri, 28 Sep 2018 11:10:04 +0800 Subject: ASoC: nau8822: new codec driver Add driver for NAU88C22. Signed-off-by: David Lin Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nau8822.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8822.c | 1136 ++++++++++++++++++++ sound/soc/codecs/nau8822.h | 204 ++++ 5 files changed, 1363 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000000..a471d162d4e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +NAU8822 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8822" + + - reg : the I2C address of the device. + +Example: + +codec: nau8822@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 774d38310875..9cc4f1848c9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -110,6 +110,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -1326,6 +1327,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d694c26192c..8ffab8c8dbfa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -107,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -371,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..622ce947f134 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1136 @@ +/* + * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver + * + * Copyright 2017 Nuvoton Technology Corp. + * + * Author: David Lin + * Co-author: John Hsu + * Co-author: Seven Li + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, ®_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Playback Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Playback Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Playback Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 1), + + SOC_DOUBLE_R_TLV("PGA Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX2_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..aa79c969cd44 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,204 @@ +/* + * nau8822.h -- NAU8822 Soc Audio Codec driver + * + * Author: David Lin + * Co-author: John Hsu + * Co-author: Seven Li + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ -- cgit v1.2.3-58-ga151 From fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,sta32x.txt | 6 +++++ sound/soc/codecs/sta32x.c | 28 ++++++++++++++++++++++ 2 files changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- cgit v1.2.3-58-ga151 From 4e9e07c5675706983ed649cfb92521a4d8aa1d6d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 17 Oct 2018 01:54:33 +0000 Subject: ASoC: pcm3168a: add hw constraint for capture channel LEFT_J / I2S only can use TDM. commit 594680ea4a394 ("ASoC: pcm3168a: add hw constraint for channel") commit 3809688980205 ("ASoC: pcm3168a: add HW constraint for non RIGHT_J") added channel constraint for it, but, it was only for playback. This patch adds constraint for capture. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 63aa02592bc0..52cc950c9fd1 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -529,11 +529,17 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; case PCM3168A_FMT_I2S: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; default: sample_min = 24; @@ -559,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { }; static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_adc, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params -- cgit v1.2.3-58-ga151 From 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- cgit v1.2.3-58-ga151 From 2eaa6e233091f51d8a629e423ad0bc080ffcb5d6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 17 Oct 2018 01:55:57 +0000 Subject: ASoC: rsnd: tidyup SSICR::SWSP for TDM R-Car datasheet is indicating that WS output settings of SSICR::SWSP is inverted on TDM mode from non TDM mode settings. But, it is meaning that TDM should use 0 here. Without this patch, sound input/output 1ch will be 2ch, 2ch will be 3ch ..., be jumbled on I2S + TDM settings. This patch fixup it. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index b42a0e0feab7..fcb4df23248c 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -392,7 +392,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv ^ is_tdm) + if (rdai->frm_clk_inv && !is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; -- cgit v1.2.3-58-ga151 From 9ab2a1bd81f7eaef4b496168dd93e0f23e8906fe Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 18 Oct 2018 10:34:30 +0300 Subject: ASoC: Intel: kbl_da7219_max98927: minor white space clean up I just added a couple missing tabs. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3ab96ee7bd3c..3fa1c3ca6d37 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -180,14 +180,14 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); - return ret; + return ret; } } if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); - return ret; + return ret; } } } -- cgit v1.2.3-58-ga151 From 3c01b0e129e9486c8004e43eba3a70de7393f645 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 11 Oct 2018 17:28:28 +0100 Subject: ASoC: dapm: Add support for hw_free on CODEC to CODEC links Currently, on power down for a CODEC to CODEC DAI link we only call digital_mute and shutdown. Provide a little more flexibility for drivers by adding a call to hw_free as well. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8c5b065c8880..a5178845065b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3737,25 +3737,30 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = 0; } + substream.stream = SNDRV_PCM_STREAM_CAPTURE; snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv; + if (source->driver->ops->hw_free) + source->driver->ops->hw_free(&substream, + source); + source->active--; - if (source->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; + if (source->driver->ops->shutdown) source->driver->ops->shutdown(&substream, source); - } } + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv; + if (sink->driver->ops->hw_free) + sink->driver->ops->hw_free(&substream, sink); + sink->active--; - if (sink->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + if (sink->driver->ops->shutdown) sink->driver->ops->shutdown(&substream, sink); - } } break; -- cgit v1.2.3-58-ga151 From 2c7b696a7589ab14854c132dc732973fbd498d5a Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler Date: Thu, 18 Oct 2018 13:18:28 +0200 Subject: ASoC: soc-core: fix trivial checkpatch issues Fix a few trivial aka cosmetic only checkpatch issues like long lines, wrong indentations, spurious blanks and newlines, missing newlines, multi-line comments etc. Signed-off-by: Marcel Ziswiler Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 146 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 88 insertions(+), 58 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 62e8e36062df..6ddcf12bc030 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -66,8 +66,9 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -/* If a DMI filed contain strings in this blacklist (e.g. - * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken +/* + * If a DMI filed contain strings in this blacklist (e.g. + * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken * as invalid and dropped when setting the card long name from DMI info. */ static const char * const dmi_blacklist[] = { @@ -222,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card) &card->pop_time); if (!card->debugfs_pop_time) dev_warn(card->dev, - "ASoC: Failed to create pop time debugfs file\n"); + "ASoC: Failed to create pop time debugfs file\n"); } static void soc_cleanup_card_debugfs(struct snd_soc_card *card) @@ -426,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); static void codec2codec_close_delayed_work(struct work_struct *work) { - /* Currently nothing to do for c2c links + /* + * Currently nothing to do for c2c links * Since c2c links are internal nodes in the DAPM graph and * don't interface with the outside world or application layer * we don't have to do any special handling on close. @@ -446,8 +448,9 @@ int snd_soc_suspend(struct device *dev) if (!card->instantiated) return 0; - /* Due to the resume being scheduled into a workqueue we could - * suspend before that's finished - wait for it to complete. + /* + * Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. */ snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); @@ -514,10 +517,13 @@ int snd_soc_suspend(struct device *dev) /* suspend all COMPONENTs */ for_each_card_components(card, component) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); - /* If there are paths active then the COMPONENT will be held with - * bias _ON and should not be suspended. */ + /* + * If there are paths active then the COMPONENT will be held + * with bias _ON and should not be suspended. + */ if (!component->suspended) { switch (snd_soc_dapm_get_bias_level(dapm)) { case SND_SOC_BIAS_STANDBY: @@ -571,18 +577,21 @@ int snd_soc_suspend(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_suspend); -/* deferred resume work, so resume can complete before we finished +/* + * deferred resume work, so resume can complete before we finished * setting our codec back up, which can be very slow on I2C */ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = - container_of(work, struct snd_soc_card, deferred_resume_work); + container_of(work, struct snd_soc_card, + deferred_resume_work); struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; int i; - /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + /* + * our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us */ @@ -699,6 +708,7 @@ int snd_soc_resume(struct device *dev) */ for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + bus_control |= cpu_dai->driver->bus_control; } if (bus_control) { @@ -777,7 +787,7 @@ struct snd_soc_dai *snd_soc_find_dai( lockdep_assert_held(&client_mutex); - /* Find CPU DAI from registered DAIs*/ + /* Find CPU DAI from registered DAIs */ for_each_component(component) { if (!snd_soc_is_matching_component(dlc, component)) continue; @@ -795,7 +805,6 @@ struct snd_soc_dai *snd_soc_find_dai( } EXPORT_SYMBOL_GPL(snd_soc_find_dai); - /** * snd_soc_find_dai_link - Find a DAI link * @@ -918,7 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, _err_defer: soc_free_pcm_runtime(rtd); - return -EPROBE_DEFER; + return -EPROBE_DEFER; } static void soc_remove_component(struct snd_soc_component *component) @@ -1074,7 +1083,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, } static int soc_init_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link) { int i, ret; struct snd_soc_dai_link_component *codec; @@ -1148,7 +1157,8 @@ static int soc_init_dai_link(struct snd_soc_card *card, void snd_soc_disconnect_sync(struct device *dev) { - struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL); + struct snd_soc_component *component = + snd_soc_lookup_component(dev, NULL); if (!component || !component->card) return; @@ -1179,7 +1189,8 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, } lockdep_assert_held(&client_mutex); - /* Notify the machine driver for extra initialization + /* + * Notify the machine driver for extra initialization * on the link created by topology. */ if (dai_link->dobj.type && card->add_dai_link) @@ -1214,7 +1225,8 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } lockdep_assert_held(&client_mutex); - /* Notify the machine driver for extra destruction + /* + * Notify the machine driver for extra destruction * on the link created by topology. */ if (dai_link->dobj.type && card->remove_dai_link) @@ -1274,7 +1286,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card, static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; int ret; @@ -1406,8 +1419,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } static int soc_probe_link_components(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - int order) + struct snd_soc_pcm_runtime *rtd, int order) { struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; @@ -1434,6 +1446,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order) if (dai->driver->probe) { int ret = dai->driver->probe(dai); + if (ret < 0) { dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", dai->name, ret); @@ -1541,7 +1554,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, } if (cpu_dai->driver->compress_new) { - /*create compress_device"*/ + /* create compress_device" */ ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", @@ -1555,7 +1568,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); + dai_link->stream_name, ret); return ret; } ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); @@ -1683,8 +1696,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } } - /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ - /* the component which has non_legacy_dai_naming is Codec */ + /* + * Flip the polarity for the "CPU" end of a CODEC<->CODEC link + * the component which has non_legacy_dai_naming is Codec + */ if (cpu_dai->component->driver->non_legacy_dai_naming) { unsigned int inv_dai_fmt; @@ -1718,9 +1733,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); - #ifdef CONFIG_DMI -/* Trim special characters, and replace '-' with '_' since '-' is used to +/* + * Trim special characters, and replace '-' with '_' since '-' is used to * separate different DMI fields in the card long name. Only number and * alphabet characters and a few separator characters are kept. */ @@ -1739,7 +1754,8 @@ static void cleanup_dmi_name(char *name) name[j] = '\0'; } -/* Check if a DMI field is valid, i.e. not containing any string +/* + * Check if a DMI field is valid, i.e. not containing any string * in the black list. */ static int is_dmi_valid(const char *field) @@ -1802,7 +1818,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) return 0; } - snprintf(card->dmi_longname, sizeof(card->snd_card->longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); @@ -1818,7 +1833,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (len < longname_buf_size) cleanup_dmi_name(card->dmi_longname + len); - /* some vendors like Lenovo may only put a self-explanatory + /* + * some vendors like Lenovo may only put a self-explanatory * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); @@ -1914,7 +1930,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->be_hw_params_fixup = component->driver->be_hw_params_fixup; - /* most BE links don't set stream name, so set it to + /* + * most BE links don't set stream name, so set it to * dai link name if it's NULL to help bind widgets. */ if (!dai_link->stream_name) @@ -1924,7 +1941,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) /* Inform userspace we are using alternate topology */ if (component->driver->topology_name_prefix) { - /* topology shortname created ? */ + /* topology shortname created? */ if (!card->topology_shortname_created) { comp_drv = component->driver; @@ -2029,7 +2046,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) goto probe_dai_err; - /* Find new DAI links added during probing components and bind them. + /* + * Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ for_each_card_links(card, dai_link) { @@ -2061,7 +2079,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_connect_dai_link_widgets(card); if (card->controls) - snd_soc_add_card_controls(card, card->controls, card->num_controls); + snd_soc_add_card_controls(card, card->controls, + card->num_controls); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, @@ -2207,8 +2226,10 @@ int snd_soc_poweroff(struct device *dev) if (!card->instantiated) return 0; - /* Flush out pmdown_time work - we actually do want to run it - * now, we're shutting down so no imminent restart. */ + /* + * Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. + */ for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); @@ -2301,6 +2322,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, data, control->name, prefix)); if (err < 0) { @@ -2418,8 +2440,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ -int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, - int source, unsigned int freq, int dir) +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) { if (component->driver->set_sysclk) return component->driver->set_sysclk(component, clk_id, source, @@ -2487,7 +2510,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, { if (component->driver->set_pll) return component->driver->set_pll(component, pll_id, source, - freq_in, freq_out); + freq_in, freq_out); return -EINVAL; } @@ -2533,8 +2556,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); * Generates the TDM tx and rx slot default masks for DAI. */ static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, - unsigned int *tx_mask, - unsigned int *rx_mask) + unsigned int *tx_mask, + unsigned int *rx_mask) { if (*tx_mask || *rx_mask) return 0; @@ -2684,7 +2707,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) return ret; /* deactivate pins to sleep state */ - for_each_card_rtds(card, rtd) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; int j; @@ -2799,7 +2822,7 @@ static char *fmt_single_name(struct device *dev, int *id) } } else { - /* I2C component devices are named "bus-addr" */ + /* I2C component devices are named "bus-addr" */ if (sscanf(name, "%x-%x", &id1, &id2) == 2) { char tmp[NAME_SIZE]; @@ -2807,7 +2830,8 @@ static char *fmt_single_name(struct device *dev, int *id) *id = ((id1 & 0xffff) << 16) + id2; /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name); + snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, + name); strlcpy(name, tmp, NAME_SIZE); } else *id = 0; @@ -2874,7 +2898,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * component-less anymore. */ if (legacy_dai_naming && - (dai_drv->id == 0 || dai_drv->name == NULL)) { + (dai_drv->id == 0 || dai_drv->name == NULL)) { dai->name = fmt_single_name(dev, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); @@ -2910,7 +2934,8 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * @count: Number of DAIs */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, + size_t count) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -2921,8 +2946,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { - dai = soc_add_dai(component, dai_drv + i, - count == 1 && !component->driver->non_legacy_dai_naming); + dai = soc_add_dai(component, dai_drv + i, count == 1 && + !component->driver->non_legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; @@ -2966,7 +2991,8 @@ int snd_soc_register_dai(struct snd_soc_component *component, if (!dai) return -ENOMEM; - /* Create the DAI widgets here. After adding DAIs, topology may + /* + * Create the DAI widgets here. After adding DAIs, topology may * also add routes that need these widgets as source or sink. */ ret = snd_soc_dapm_new_dai_widgets(dapm, dai); @@ -3048,7 +3074,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component) #ifdef CONFIG_REGMAP /** - * snd_soc_component_init_regmap() - Initialize regmap instance for the component + * snd_soc_component_init_regmap() - Initialize regmap instance for the + * component * @component: The component for which to initialize the regmap instance * @regmap: The regmap instance that should be used by the component * @@ -3066,7 +3093,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap); /** - * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component + * snd_soc_component_exit_regmap() - De-initialize regmap instance for the + * component * @component: The component for which to de-initialize the regmap instance * * Calls regmap_exit() on the regmap instance associated to the component and @@ -3090,7 +3118,8 @@ static void snd_soc_component_add(struct snd_soc_component *component) if (!component->driver->write && !component->driver->read) { if (!component->regmap) - component->regmap = dev_get_regmap(component->dev, NULL); + component->regmap = dev_get_regmap(component->dev, + NULL); if (component->regmap) snd_soc_component_setup_regmap(component); } @@ -3235,23 +3264,24 @@ static int __snd_soc_unregister_component(struct device *dev) if (dev != component->dev) continue; - snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL); + snd_soc_tplg_component_remove(component, + SND_SOC_TPLG_INDEX_ALL); snd_soc_component_del_unlocked(component); found = 1; break; } mutex_unlock(&client_mutex); - if (found) { + if (found) snd_soc_component_cleanup(component); - } return found; } void snd_soc_unregister_component(struct device *dev) { - while (__snd_soc_unregister_component(dev)); + while (__snd_soc_unregister_component(dev)) + ; } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); @@ -3832,7 +3862,7 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, for_each_link_codecs(dai_link, index, component) { ret = of_parse_phandle_with_args(of_node, name, "#sound-dai-cells", - index, &args); + index, &args); if (ret) goto err; component->of_node = args.np; -- cgit v1.2.3-58-ga151 From 8307b2afd386ccce369821daa2196068c47fe8cd Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 15 Oct 2018 16:03:35 +0200 Subject: ASoC: stm32: sai: set sai as mclk clock provider Add master clock generation support in STM32 SAI. The master clock provided by SAI can be used to feed a codec. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.h | 3 + sound/soc/stm/stm32_sai_sub.c | 275 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 242 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index f25422174909..08de899c766b 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -91,6 +91,9 @@ #define SAI_XCR1_OSR_SHIFT 26 #define SAI_XCR1_OSR BIT(SAI_XCR1_OSR_SHIFT) +#define SAI_XCR1_MCKEN_SHIFT 27 +#define SAI_XCR1_MCKEN BIT(SAI_XCR1_MCKEN_SHIFT) + /******************* Bit definition for SAI_XCR2 register *******************/ #define SAI_XCR2_FTH_SHIFT 0 #define SAI_XCR2_FTH_MASK GENMASK(2, SAI_XCR2_FTH_SHIFT) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 56a227e0bd71..31d22abd3204 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include @@ -68,6 +69,8 @@ #define SAI_IEC60958_BLOCK_FRAMES 192 #define SAI_IEC60958_STATUS_BYTES 24 +#define SAI_MCLK_NAME_LEN 32 + /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) * @pdev: device data pointer @@ -80,6 +83,7 @@ * @pdata: SAI block parent data pointer * @np_sync_provider: synchronization provider node * @sai_ck: kernel clock feeding the SAI clock generator + * @sai_mclk: master clock from SAI mclk provider * @phys_addr: SAI registers physical base address * @mclk_rate: SAI block master clock frequency (Hz). set at init * @id: SAI sub block id corresponding to sub-block A or B @@ -110,6 +114,7 @@ struct stm32_sai_sub_data { struct stm32_sai_data *pdata; struct device_node *np_sync_provider; struct clk *sai_ck; + struct clk *sai_mclk; dma_addr_t phys_addr; unsigned int mclk_rate; unsigned int id; @@ -251,6 +256,177 @@ static const struct snd_kcontrol_new iec958_ctls = { .put = snd_pcm_iec958_put, }; +struct stm32_sai_mclk_data { + struct clk_hw hw; + unsigned long freq; + struct stm32_sai_sub_data *sai_data; +}; + +#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) +#define STM32_SAI_MAX_CLKS 1 + +static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, + unsigned long input_rate, + unsigned long output_rate) +{ + int version = sai->pdata->conf->version; + int div; + + div = DIV_ROUND_CLOSEST(input_rate, output_rate); + if (div > SAI_XCR1_MCKDIV_MAX(version)) { + dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); + return -EINVAL; + } + dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div); + + if (input_rate % div) + dev_dbg(&sai->pdev->dev, + "Rate not accurate. requested (%ld), actual (%ld)\n", + output_rate, input_rate / div); + + return div; +} + +static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, + unsigned int div) +{ + int version = sai->pdata->conf->version; + int ret, cr1, mask; + + if (div > SAI_XCR1_MCKDIV_MAX(version)) { + dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); + return -EINVAL; + } + + mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); + cr1 = SAI_XCR1_MCKDIV_SET(div); + ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); + if (ret < 0) + dev_err(&sai->pdev->dev, "Failed to update CR1 register\n"); + + return ret; +} + +static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *prate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + int div; + + div = stm32_sai_get_clk_div(sai, *prate, rate); + if (div < 0) + return div; + + mclk->freq = *prate / div; + + return mclk->freq; +} + +static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + + return mclk->freq; +} + +static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + unsigned int div; + int ret; + + div = stm32_sai_get_clk_div(sai, parent_rate, rate); + if (div < 0) + return div; + + ret = stm32_sai_set_clk_div(sai, div); + if (ret) + return ret; + + mclk->freq = rate; + + return 0; +} + +static int stm32_sai_mclk_enable(struct clk_hw *hw) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + + dev_dbg(&sai->pdev->dev, "Enable master clock\n"); + + return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, + SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); +} + +static void stm32_sai_mclk_disable(struct clk_hw *hw) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + + dev_dbg(&sai->pdev->dev, "Disable master clock\n"); + + regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); +} + +static const struct clk_ops mclk_ops = { + .enable = stm32_sai_mclk_enable, + .disable = stm32_sai_mclk_disable, + .recalc_rate = stm32_sai_mclk_recalc_rate, + .round_rate = stm32_sai_mclk_round_rate, + .set_rate = stm32_sai_mclk_set_rate, +}; + +static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) +{ + struct clk_hw *hw; + struct stm32_sai_mclk_data *mclk; + struct device *dev = &sai->pdev->dev; + const char *pname = __clk_get_name(sai->sai_ck); + char *mclk_name, *p, *s = (char *)pname; + int ret, i = 0; + + mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + if (!mclk) + return -ENOMEM; + + mclk_name = devm_kcalloc(dev, sizeof(char), + SAI_MCLK_NAME_LEN, GFP_KERNEL); + if (!mclk_name) + return -ENOMEM; + + /* + * Forge mclk clock name from parent clock name and suffix. + * String after "_" char is stripped in parent name. + */ + p = mclk_name; + while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { + *p++ = *s++; + i++; + } + STM_SAI_IS_SUB_A(sai) ? + strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); + + mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); + mclk->sai_data = sai; + hw = &mclk->hw; + + dev_dbg(dev, "Register master clock %s\n", mclk_name); + ret = devm_clk_hw_register(&sai->pdev->dev, hw); + if (ret) { + dev_err(dev, "mclk register returned %d\n", ret); + return ret; + } + sai->sai_mclk = hw->clk; + + /* register mclk provider */ + return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); +} + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -312,15 +488,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if ((dir == SND_SOC_CLOCK_OUT) && sai->master) { + if (dir == SND_SOC_CLOCK_OUT) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - sai->mclk_rate = freq; dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; + + if (sai->sai_mclk) { + ret = clk_set_rate_exclusive(sai->sai_mclk, + sai->mclk_rate); + if (ret) { + dev_err(cpu_dai->dev, + "Could not set mclk rate\n"); + return ret; + } + } } return 0; @@ -715,15 +901,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int cr1, mask, div = 0; - int sai_clk_rate, mclk_ratio, den, ret; - int version = sai->pdata->conf->version; + int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); - if (!sai->mclk_rate) { - dev_err(cpu_dai->dev, "Mclk rate is null\n"); - return -EINVAL; - } - if (!(rate % 11025)) clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); else @@ -731,14 +911,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { - /* - * mclk_rate = 256 * fs - * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate - * MCKDIV = sai_ck / (2 * mclk_rate) otherwise + /* mclk on (NODIV=0) + * mclk_rate = 256 * fs + * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate + * MCKDIV = sai_ck / (2 * mclk_rate) otherwise + * mclk off (NODIV=1) + * MCKDIV ignored. sck = sai_ck */ - if (2 * sai_clk_rate >= 3 * sai->mclk_rate) - div = DIV_ROUND_CLOSEST(sai_clk_rate, - 2 * sai->mclk_rate); + if (!sai->mclk_rate) + return 0; + + if (2 * sai_clk_rate >= 3 * sai->mclk_rate) { + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + 2 * sai->mclk_rate); + if (div < 0) + return div; + } } else { /* * TDM mode : @@ -750,8 +938,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, * Note: NOMCK/NODIV correspond to same bit. */ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { - div = DIV_ROUND_CLOSEST(sai_clk_rate, - (params_rate(params) * 128)); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + rate * 128); + if (div < 0) + return div; } else { if (sai->mclk_rate) { mclk_ratio = sai->mclk_rate / rate; @@ -764,31 +954,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, mclk_ratio); return -EINVAL; } - div = DIV_ROUND_CLOSEST(sai_clk_rate, - sai->mclk_rate); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + sai->mclk_rate); + if (div < 0) + return div; } else { /* mclk-fs not set, master clock not active */ den = sai->fs_length * params_rate(params); - div = DIV_ROUND_CLOSEST(sai_clk_rate, den); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + den); + if (div < 0) + return div; } } } - if (div > SAI_XCR1_MCKDIV_MAX(version)) { - dev_err(cpu_dai->dev, "Divider %d out of range\n", div); - return -EINVAL; - } - dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); - - mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); - cr1 = SAI_XCR1_MCKDIV_SET(div); - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); - if (ret < 0) { - dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); - return ret; - } - - return 0; + return stm32_sai_set_clk_div(sai, div); } static int stm32_sai_hw_params(struct snd_pcm_substream *substream, @@ -881,6 +1062,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, SAI_XCR1_NODIV); clk_disable_unprepare(sai->sai_ck); + + clk_rate_exclusive_put(sai->sai_mclk); + sai->substream = NULL; } @@ -903,6 +1087,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); int cr1 = 0, cr1_mask; + sai->cpu_dai = cpu_dai; + sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); /* * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice, @@ -1181,6 +1367,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return PTR_ERR(sai->sai_ck); } + if (STM_SAI_IS_F4(sai->pdata)) + return 0; + + /* Register mclk provider if requested */ + if (of_find_property(np, "#clock-cells", NULL)) { + ret = stm32_sai_add_mclk_provider(sai); + if (ret < 0) + return ret; + } else { + sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK"); + if (IS_ERR(sai->sai_mclk)) { + if (PTR_ERR(sai->sai_mclk) != -ENOENT) + return PTR_ERR(sai->sai_mclk); + sai->sai_mclk = NULL; + } + } + return 0; } -- cgit v1.2.3-58-ga151 From 5e8d63a726f8f2fef089515e9a501785cc67dcdc Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 15 Oct 2018 16:03:36 +0200 Subject: ASoC: cs42l51: add mclk support Add MCLK dapm to allow configuration of cirrus CS42l51 codec as a master clock consumer. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 5080d7a3c279..eb40bff54cec 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -237,6 +237,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { + SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") +}; + static const struct snd_soc_dapm_route cs42l51_routes[] = { {"HPL", NULL, "Left DAC"}, {"HPR", NULL, "Right DAC"}, @@ -487,6 +491,10 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_component_probe(struct snd_soc_component *component) { int ret, reg; + struct snd_soc_dapm_context *dapm; + + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); /* * DAC configuration -- cgit v1.2.3-58-ga151 From 2a2aefa41ce48ace8e1e963cb10c3f5ff43aa994 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 19 Oct 2018 13:25:15 +0100 Subject: ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- cgit v1.2.3-58-ga151 From e3a360b8cdede74d25807fc405e5d8bfb025692f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 19 Oct 2018 13:25:16 +0100 Subject: ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Increase the address value width in the debug log from 4 digits to 8 digits to allow for DSP cores with larger memory address ranges. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7ae10c632614..a53dc174bbf0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; adsp_dbg(buf->dsp, - "region=%d type=%d base=%04x off=%04x size=%04x\n", + "region=%d type=%d base=%08x off=%08x size=%08x\n", i, region->mem_type, region->base_addr, region->offset, region->cumulative_size); } -- cgit v1.2.3-58-ga151 From 318e741ee13b5a72f3051d9bb6852b1f4d02d0bb Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 19 Oct 2018 17:56:35 +0200 Subject: ASoC: cs42l51: fix mclk support The MCLK clock is made optional for cs42l51 codec. However, ASoC DAPM clock supply widget, expects the clock to be defined unconditionally. Register MCLK DAPM conditionally in codec driver, depending on clock presence in DT. Fixes: 5e8d63a726f8 ("ASoC: cs42l51: add mclk support") Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index eb40bff54cec..fd2bd74024c1 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -21,6 +21,7 @@ * - master mode *NOT* supported */ +#include #include #include #include @@ -41,6 +42,7 @@ enum master_slave_mode { struct cs42l51_private { unsigned int mclk; + struct clk *mclk_handle; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; }; @@ -492,9 +494,13 @@ static int cs42l51_component_probe(struct snd_soc_component *component) { int ret, reg; struct snd_soc_dapm_context *dapm; + struct cs42l51_private *cs42l51; + cs42l51 = snd_soc_component_get_drvdata(component); dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); + + if (cs42l51->mclk_handle) + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); /* * DAC configuration @@ -548,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, cs42l51); + cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); + if (IS_ERR(cs42l51->mclk_handle)) { + if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT) + return PTR_ERR(cs42l51->mclk_handle); + cs42l51->mclk_handle = NULL; + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { -- cgit v1.2.3-58-ga151 From c5d09485def41cab9e75ba23abbf87080183183c Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Fri, 19 Oct 2018 17:44:22 +0100 Subject: ASoC: wm2000: Remove wm2000_read helper function The return type "unsigned int" was used by the wm2000_read() function despite of the aspect that it will eventually return a negative error code. The resulting function doesn't add much to the code, so replace wm2000_read with regmap_read. Signed-off-by: Lucas Tanure Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 54 ++++++++++++++++++++++++++--------------------- 1 file changed, 30 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c5ae07234a00..bba330e30162 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg, return regmap_write(wm2000->regmap, reg, value); } -static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) -{ - struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); - unsigned int val; - int ret; - - ret = regmap_read(wm2000->regmap, r, &val); - if (ret < 0) - return -1; - - return val; -} - static void wm2000_reset(struct wm2000_priv *wm2000) { struct i2c_client *i2c = wm2000->i2c; @@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000) static int wm2000_poll_bit(struct i2c_client *i2c, unsigned int reg, u8 mask) { + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); int timeout = 4000; - int val; + unsigned int val; - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); while (!(val & mask) && --timeout) { msleep(1); - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); } if (timeout == 0) @@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); unsigned long rate; + unsigned int val; int ret; if (WARN_ON(wm2000->anc_mode != ANC_OFF)) @@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) WM2000_MODE_THERMAL_ENABLE); } - ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret); + regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies); + return ret; + } if (wm2000->speech_clarity) - ret |= WM2000_SPEECH_CLARITY; + val |= WM2000_SPEECH_CLARITY; else - ret &= ~WM2000_SPEECH_CLARITY; - wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + val &= ~WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); @@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, const char *filename; const struct firmware *fw = NULL; int ret, i; - int reg; + unsigned int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL); @@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, } /* Verify that this is a WM2000 */ - reg = wm2000_read(i2c, WM2000_REG_ID1); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret); + return ret; + } id = reg << 8; - reg = wm2000_read(i2c, WM2000_REG_ID2); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret); + return ret; + } id |= reg & 0xff; if (id != 0x2000) { @@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, goto err_supplies; } - reg = wm2000_read(i2c, WM2000_REG_REVISON); + ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret); + return ret; + } dev_info(&i2c->dev, "revision %c\n", reg + 'A'); wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK"); -- cgit v1.2.3-58-ga151 From 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Sun, 21 Oct 2018 08:39:11 -0700 Subject: ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++++++-------------------------- 1 file changed, 25 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index c63d226e2436..d5ec1a20499d 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- cgit v1.2.3-58-ga151 From e6d7942ce602e99d506a7c08f1935a274645dfda Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 22 Oct 2018 17:10:45 +0200 Subject: ASoC: stm32: add clock dependency for sai Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Add COMMON_CLK dependency for STM32 SAI, as it is required by clock provider. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 9b2681397dba..c66ffa72057e 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support" config SND_SOC_STM32_SAI tristate "STM32 SAI interface (Serial Audio Interface) support" depends on (ARCH_STM32 && OF) || COMPILE_TEST + depends on COMMON_CLK depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO -- cgit v1.2.3-58-ga151 From 6be0f96d799f487f05eb412d362d5a1747d665c2 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 22 Oct 2018 17:10:46 +0200 Subject: ASoC: stm32: sai: fix master clock naming Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Fix warning issued by strncat when bound equals to source length. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 31d22abd3204..ea05cc91aa05 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -404,12 +404,11 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) * String after "_" char is stripped in parent name. */ p = mclk_name; - while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { + while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 7))) { *p++ = *s++; i++; } - STM_SAI_IS_SUB_A(sai) ? - strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); + STM_SAI_IS_SUB_A(sai) ? strcat(p, "a_mclk") : strcat(p, "b_mclk"); mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); mclk->sai_data = sai; -- cgit v1.2.3-58-ga151