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authorTakashi Iwai <tiwai@suse.de>2018-08-13 12:12:31 +0200
committerTakashi Iwai <tiwai@suse.de>2018-08-13 12:12:31 +0200
commitf5b6c1fcb42fe7d6f2f6eb2220512e2a5f875133 (patch)
tree325f29d9788e80a0dd66d907ce38650834060e4b /sound/soc
parent73b383141d296c55bfbc0ce336a4a946627e7780 (diff)
parent4aa5db22d35588e1a5d2ee88472348ea73d9fb23 (diff)
Merge tag 'asoc-v4.19' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.19 A fairly big update, including quite a bit of core activity this time around (which is good to see) along with a fairly large set of new drivers. - A new snd_pcm_stop_xrun() helper which is now used in several drivers. - Support for providing name prefixes to generic component nodes. - Quite a few fixes for DPCM as it gains a bit wider use and more robust testing. - Generalization of the DIO2125 support to a simple amplifier driver. - Accessory detection support for the audio graph card. - DT support for PXA AC'97 devices. - Quirks for a number of new x86 systems. - Support for AM Logic Meson, Everest ES7154, Intel systems with RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707.
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/amd/Kconfig1
-rw-r--r--sound/soc/amd/acp-da7219-max98357a.c109
-rw-r--r--sound/soc/amd/acp-pcm-dma.c214
-rw-r--r--sound/soc/amd/acp.h13
-rw-r--r--sound/soc/atmel/atmel-i2s.c46
-rw-r--r--sound/soc/codecs/Kconfig25
-rw-r--r--sound/soc/codecs/Makefile8
-rw-r--r--sound/soc/codecs/adau17x1.c1
-rw-r--r--sound/soc/codecs/adav80x.c1
-rw-r--r--sound/soc/codecs/ak4458.c2
-rw-r--r--sound/soc/codecs/ak4554.c17
-rw-r--r--sound/soc/codecs/ak4613.c26
-rw-r--r--sound/soc/codecs/ak4642.c26
-rw-r--r--sound/soc/codecs/ak5558.c4
-rw-r--r--sound/soc/codecs/cs4270.c2
-rw-r--r--sound/soc/codecs/cs47l24.c11
-rw-r--r--sound/soc/codecs/cx20442.c23
-rw-r--r--sound/soc/codecs/da7210.c27
-rw-r--r--sound/soc/codecs/da7213.c4
-rw-r--r--sound/soc/codecs/da7219-aad.c5
-rw-r--r--sound/soc/codecs/da7219.c48
-rw-r--r--sound/soc/codecs/da7219.h8
-rw-r--r--sound/soc/codecs/da9055.c4
-rw-r--r--sound/soc/codecs/es7134.c227
-rw-r--r--sound/soc/codecs/es7241.c322
-rw-r--r--sound/soc/codecs/hdac_hdmi.c69
-rw-r--r--sound/soc/codecs/hdmi-codec.c21
-rw-r--r--sound/soc/codecs/max98373.c1
-rw-r--r--sound/soc/codecs/max9850.c4
-rw-r--r--sound/soc/codecs/nau8540.c3
-rw-r--r--sound/soc/codecs/nau8824.c2
-rw-r--r--sound/soc/codecs/nau8825.c2
-rw-r--r--sound/soc/codecs/pcm1789.c3
-rw-r--r--sound/soc/codecs/pcm186x.c2
-rw-r--r--sound/soc/codecs/rt1305.c15
-rw-r--r--sound/soc/codecs/rt5514.c8
-rw-r--r--sound/soc/codecs/rt5631.c12
-rw-r--r--sound/soc/codecs/rt5640.c2
-rw-r--r--sound/soc/codecs/rt5651.c235
-rw-r--r--sound/soc/codecs/rt5651.h8
-rw-r--r--sound/soc/codecs/rt5677.c3
-rw-r--r--sound/soc/codecs/rt5682.c2681
-rw-r--r--sound/soc/codecs/rt5682.h1324
-rw-r--r--sound/soc/codecs/simple-amplifier.c (renamed from sound/soc/codecs/dio2125.c)42
-rw-r--r--sound/soc/codecs/tas571x.c110
-rw-r--r--sound/soc/codecs/tas571x.h16
-rw-r--r--sound/soc/codecs/tda7419.c4
-rw-r--r--sound/soc/codecs/tscs42xx.c37
-rw-r--r--sound/soc/codecs/tscs42xx.h8
-rw-r--r--sound/soc/codecs/twl6040.c2
-rw-r--r--sound/soc/codecs/wm2200.c10
-rw-r--r--sound/soc/codecs/wm5100-tables.c12
-rw-r--r--sound/soc/codecs/wm5102.c10
-rw-r--r--sound/soc/codecs/wm5110.c13
-rw-r--r--sound/soc/codecs/wm8903.c4
-rw-r--r--sound/soc/codecs/wm8904.c1
-rw-r--r--sound/soc/codecs/wm8955.c1
-rw-r--r--sound/soc/codecs/wm8960.c1
-rw-r--r--sound/soc/codecs/wm8961.c1
-rw-r--r--sound/soc/codecs/wm8962.c1
-rw-r--r--sound/soc/codecs/wm8988.c4
-rw-r--r--sound/soc/codecs/wm8990.c4
-rw-r--r--sound/soc/codecs/wm8994.c1
-rw-r--r--sound/soc/codecs/wm8995.c1
-rw-r--r--sound/soc/codecs/wm8996.c9
-rw-r--r--sound/soc/codecs/wm9081.c1
-rw-r--r--sound/soc/codecs/wm_adsp.c216
-rw-r--r--sound/soc/codecs/wm_adsp.h12
-rw-r--r--sound/soc/codecs/wmfw.h1
-rw-r--r--sound/soc/davinci/davinci-i2s.c1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c16
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c20
-rw-r--r--sound/soc/fsl/fsl_asrc.c18
-rw-r--r--sound/soc/fsl/fsl_asrc.h5
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c18
-rw-r--r--sound/soc/fsl/fsl_esai.c1
-rw-r--r--sound/soc/fsl/fsl_spdif.c2
-rw-r--r--sound/soc/fsl/fsl_utils.c18
-rw-r--r--sound/soc/fsl/fsl_utils.h7
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c15
-rw-r--r--sound/soc/generic/audio-graph-card.c41
-rw-r--r--sound/soc/generic/audio-graph-scu-card.c25
-rw-r--r--sound/soc/generic/simple-card-utils.c74
-rw-r--r--sound/soc/generic/simple-card.c106
-rw-r--r--sound/soc/generic/simple-scu-card.c21
-rw-r--r--sound/soc/intel/atom/sst/sst_drv_interface.c29
-rw-r--r--sound/soc/intel/atom/sst/sst_loader.c6
-rw-r--r--sound/soc/intel/boards/Kconfig14
-rw-r--r--sound/soc/intel/boards/Makefile2
-rw-r--r--sound/soc/intel/boards/bdw-rt5677.c4
-rw-r--r--sound/soc/intel/boards/bxt_da7219_max98357a.c20
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c2
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c55
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c364
-rw-r--r--sound/soc/intel/boards/glk_rt5682_max98357a.c643
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98357a.c3
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_max98927.c4
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c4
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_max98357a.c2
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_ssm4567.c2
-rw-r--r--sound/soc/intel/boards/skl_rt286.c2
-rw-r--r--sound/soc/intel/common/Makefile6
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-bxt-match.c59
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-byt-match.c40
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cht-match.c56
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cnl-match.c32
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-glk-match.c41
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c16
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-kbl-match.c91
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-skl-match.c47
-rw-r--r--sound/soc/intel/common/sst-firmware.c6
-rw-r--r--sound/soc/intel/haswell/sst-haswell-dsp.c53
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c7
-rw-r--r--sound/soc/intel/skylake/skl-sst-cldma.c8
-rw-r--r--sound/soc/intel/skylake/skl-sst-cldma.h2
-rw-r--r--sound/soc/intel/skylake/skl-topology.c8
-rw-r--r--sound/soc/intel/skylake/skl-topology.h5
-rw-r--r--sound/soc/intel/skylake/skl.c170
-rw-r--r--sound/soc/mediatek/common/mtk-afe-platform-driver.c64
-rw-r--r--sound/soc/mediatek/common/mtk-base-afe.h6
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-afe-common.h1
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-afe-pcm.c65
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-dai-adda.c20
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-dai-hostless.c16
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-dai-pcm.c19
-rw-r--r--sound/soc/meson/Kconfig65
-rw-r--r--sound/soc/meson/Makefile21
-rw-r--r--sound/soc/meson/axg-card.c671
-rw-r--r--sound/soc/meson/axg-fifo.c341
-rw-r--r--sound/soc/meson/axg-fifo.h80
-rw-r--r--sound/soc/meson/axg-frddr.c141
-rw-r--r--sound/soc/meson/axg-spdifout.c456
-rw-r--r--sound/soc/meson/axg-tdm-formatter.c381
-rw-r--r--sound/soc/meson/axg-tdm-formatter.h39
-rw-r--r--sound/soc/meson/axg-tdm-interface.c542
-rw-r--r--sound/soc/meson/axg-tdm.h78
-rw-r--r--sound/soc/meson/axg-tdmin.c229
-rw-r--r--sound/soc/meson/axg-tdmout.c259
-rw-r--r--sound/soc/meson/axg-toddr.c199
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c2
-rw-r--r--sound/soc/omap/omap-dmic.c2
-rw-r--r--sound/soc/omap/omap-mcpdm.c4
-rw-r--r--sound/soc/pxa/Kconfig6
-rw-r--r--sound/soc/pxa/magician.c106
-rw-r--r--sound/soc/pxa/pxa-ssp.c181
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c47
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c9
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c73
-rw-r--r--sound/soc/pxa/zylonite.c9
-rw-r--r--sound/soc/qcom/Kconfig14
-rw-r--r--sound/soc/qcom/Makefile4
-rw-r--r--sound/soc/qcom/apq8096.c188
-rw-r--r--sound/soc/qcom/common.c112
-rw-r--r--sound/soc/qcom/common.h11
-rw-r--r--sound/soc/qcom/lpass-platform.c2
-rw-r--r--sound/soc/qcom/qdsp6/q6adm.c16
-rw-r--r--sound/soc/qcom/qdsp6/q6afe-dai.c225
-rw-r--r--sound/soc/qcom/qdsp6/q6afe.c43
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c42
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c17
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c71
-rw-r--r--sound/soc/qcom/sdm845.c285
-rw-r--r--sound/soc/rockchip/Makefile3
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c3
-rw-r--r--sound/soc/rockchip/rockchip_pcm.c45
-rw-r--r--sound/soc/rockchip/rockchip_pcm.h14
-rw-r--r--sound/soc/rockchip/rockchip_rt5645.c27
-rw-r--r--sound/soc/samsung/i2s.c1
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/sh/dma-sh7760.c26
-rw-r--r--sound/soc/sh/fsi.c22
-rw-r--r--sound/soc/sh/hac.c20
-rw-r--r--sound/soc/sh/migor.c14
-rw-r--r--sound/soc/sh/rcar/Makefile1
-rw-r--r--sound/soc/sh/rcar/adg.c15
-rw-r--r--sound/soc/sh/rcar/cmd.c19
-rw-r--r--sound/soc/sh/rcar/core.c41
-rw-r--r--sound/soc/sh/rcar/ctu.c15
-rw-r--r--sound/soc/sh/rcar/dma.c17
-rw-r--r--sound/soc/sh/rcar/dvc.c16
-rw-r--r--sound/soc/sh/rcar/gen.c16
-rw-r--r--sound/soc/sh/rcar/mix.c14
-rw-r--r--sound/soc/sh/rcar/rsnd.h17
-rw-r--r--sound/soc/sh/rcar/src.c16
-rw-r--r--sound/soc/sh/rcar/ssi.c59
-rw-r--r--sound/soc/sh/rcar/ssiu.c15
-rw-r--r--sound/soc/sh/sh7760-ac97.c14
-rw-r--r--sound/soc/sh/siu.h26
-rw-r--r--sound/soc/sh/siu_dai.c26
-rw-r--r--sound/soc/sh/siu_pcm.c27
-rw-r--r--sound/soc/sh/ssi.c21
-rw-r--r--sound/soc/sirf/sirf-usp.c7
-rw-r--r--sound/soc/soc-ac97.c29
-rw-r--r--sound/soc/soc-acpi.c20
-rw-r--r--sound/soc/soc-compress.c120
-rw-r--r--sound/soc/soc-core.c200
-rw-r--r--sound/soc/soc-dapm.c51
-rw-r--r--sound/soc/soc-devres.c15
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c30
-rw-r--r--sound/soc/soc-io.c19
-rw-r--r--sound/soc/soc-jack.c19
-rw-r--r--sound/soc/soc-ops.c29
-rw-r--r--sound/soc/soc-pcm.c469
-rw-r--r--sound/soc/soc-topology.c91
-rw-r--r--sound/soc/soc-utils.c24
-rw-r--r--sound/soc/sti/uniperif_player.c6
-rw-r--r--sound/soc/sti/uniperif_reader.c2
-rw-r--r--sound/soc/stm/Kconfig1
-rw-r--r--sound/soc/stm/stm32_adfsdm.c10
-rw-r--r--sound/soc/stm/stm32_sai_sub.c146
-rw-r--r--sound/soc/tegra/tegra20_ac97.c2
-rw-r--r--sound/soc/tegra/tegra30_i2s.h2
-rw-r--r--sound/soc/tegra/tegra_alc5632.c17
-rw-r--r--sound/soc/tegra/tegra_rt5677.c17
-rw-r--r--sound/soc/uniphier/aio-core.c84
-rw-r--r--sound/soc/uniphier/aio-cpu.c5
-rw-r--r--sound/soc/uniphier/aio-ld11.c2
-rw-r--r--sound/soc/uniphier/aio-reg.h1
-rw-r--r--sound/soc/uniphier/aio.h6
-rw-r--r--sound/soc/zte/zx-tdm.c4
222 files changed, 12968 insertions, 2457 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 41af6b9cc350..1cf11cf51e1d 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig"
source "sound/soc/img/Kconfig"
source "sound/soc/intel/Kconfig"
source "sound/soc/mediatek/Kconfig"
+source "sound/soc/meson/Kconfig"
source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/qcom/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 06389a5385d7..62a5f87c3cfc 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += img/
obj-$(CONFIG_SND_SOC) += intel/
obj-$(CONFIG_SND_SOC) += mediatek/
+obj-$(CONFIG_SND_SOC) += meson/
obj-$(CONFIG_SND_SOC) += mxs/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 6cbf9cf4d1a4..58c1dcb4d255 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_ADAU7002
+ select REGULATOR
depends on SND_SOC_AMD_ACP && I2C
help
This option enables machine driver for DA7219 and MAX9835.
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index ccddc6650b9c..8e3275a96a82 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -32,6 +32,8 @@
#include <linux/clk.h>
#include <linux/gpio.h>
#include <linux/module.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/driver.h>
#include <linux/i2c.h>
#include <linux/input.h>
#include <linux/acpi.h>
@@ -148,7 +150,8 @@ 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_BT_INSTANCE;
+ machine->i2s_instance = I2S_SP_INSTANCE;
+ machine->capture_channel = CAP_CHANNEL1;
return da7219_clk_enable(substream);
}
@@ -163,7 +166,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
struct snd_soc_card *card = rtd->card;
struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
- machine->i2s_instance = I2S_SP_INSTANCE;
+ machine->i2s_instance = I2S_BT_INSTANCE;
return da7219_clk_enable(substream);
}
@@ -172,13 +175,24 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
da7219_clk_disable();
}
-static int cz_dmic_startup(struct snd_pcm_substream *substream)
+static int cz_dmic0_startup(struct snd_pcm_substream *substream)
+{
+ 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);
+
+ machine->i2s_instance = I2S_BT_INSTANCE;
+ return da7219_clk_enable(substream);
+}
+
+static int cz_dmic1_startup(struct snd_pcm_substream *substream)
{
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);
machine->i2s_instance = I2S_SP_INSTANCE;
+ machine->capture_channel = CAP_CHANNEL0;
return da7219_clk_enable(substream);
}
@@ -197,23 +211,39 @@ static const struct snd_soc_ops cz_max_play_ops = {
.shutdown = cz_max_shutdown,
};
-static const struct snd_soc_ops cz_dmic_cap_ops = {
- .startup = cz_dmic_startup,
+static const struct snd_soc_ops cz_dmic0_cap_ops = {
+ .startup = cz_dmic0_startup,
+ .shutdown = cz_dmic_shutdown,
+};
+
+static const struct snd_soc_ops cz_dmic1_cap_ops = {
+ .startup = cz_dmic1_startup,
.shutdown = cz_dmic_shutdown,
};
static struct snd_soc_dai_link cz_dai_7219_98357[] = {
{
- .name = "amd-da7219-play-cap",
- .stream_name = "Playback and Capture",
+ .name = "amd-da7219-play",
+ .stream_name = "Playback",
.platform_name = "acp_audio_dma.0.auto",
- .cpu_dai_name = "designware-i2s.3.auto",
+ .cpu_dai_name = "designware-i2s.1.auto",
.codec_dai_name = "da7219-hifi",
.codec_name = "i2c-DLGS7219:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.init = cz_da7219_init,
.dpcm_playback = 1,
+ .ops = &cz_da7219_cap_ops,
+ },
+ {
+ .name = "amd-da7219-cap",
+ .stream_name = "Capture",
+ .platform_name = "acp_audio_dma.0.auto",
+ .cpu_dai_name = "designware-i2s.2.auto",
+ .codec_dai_name = "da7219-hifi",
+ .codec_name = "i2c-DLGS7219:00",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
.ops = &cz_da7219_cap_ops,
},
@@ -221,7 +251,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
.name = "amd-max98357-play",
.stream_name = "HiFi Playback",
.platform_name = "acp_audio_dma.0.auto",
- .cpu_dai_name = "designware-i2s.1.auto",
+ .cpu_dai_name = "designware-i2s.3.auto",
.codec_dai_name = "HiFi",
.codec_name = "MX98357A:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -230,8 +260,22 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
.ops = &cz_max_play_ops,
},
{
- .name = "dmic",
- .stream_name = "DMIC Capture",
+ /* C panel DMIC */
+ .name = "dmic0",
+ .stream_name = "DMIC0 Capture",
+ .platform_name = "acp_audio_dma.0.auto",
+ .cpu_dai_name = "designware-i2s.3.auto",
+ .codec_dai_name = "adau7002-hifi",
+ .codec_name = "ADAU7002:00",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .dpcm_capture = 1,
+ .ops = &cz_dmic0_cap_ops,
+ },
+ {
+ /* A/B panel DMIC */
+ .name = "dmic1",
+ .stream_name = "DMIC1 Capture",
.platform_name = "acp_audio_dma.0.auto",
.cpu_dai_name = "designware-i2s.2.auto",
.codec_dai_name = "adau7002-hifi",
@@ -239,7 +283,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
- .ops = &cz_dmic_cap_ops,
+ .ops = &cz_dmic1_cap_ops,
},
};
@@ -278,11 +322,52 @@ static struct snd_soc_card cz_card = {
.num_controls = ARRAY_SIZE(cz_mc_controls),
};
+static struct regulator_consumer_supply acp_da7219_supplies[] = {
+ REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"),
+ REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"),
+ REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"),
+ REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"),
+};
+
+static struct regulator_init_data acp_da7219_data = {
+ .constraints = {
+ .always_on = 1,
+ },
+ .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies),
+ .consumer_supplies = acp_da7219_supplies,
+};
+
+static struct regulator_config acp_da7219_cfg = {
+ .init_data = &acp_da7219_data,
+};
+
+static struct regulator_ops acp_da7219_ops = {
+};
+
+static struct regulator_desc acp_da7219_desc = {
+ .name = "reg-fixed-1.8V",
+ .type = REGULATOR_VOLTAGE,
+ .owner = THIS_MODULE,
+ .ops = &acp_da7219_ops,
+ .fixed_uV = 1800000, /* 1.8V */
+ .n_voltages = 1,
+};
+
static int cz_probe(struct platform_device *pdev)
{
int ret;
struct snd_soc_card *card;
struct acp_platform_info *machine;
+ struct regulator_dev *rdev;
+
+ acp_da7219_cfg.dev = &pdev->dev;
+ rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc,
+ &acp_da7219_cfg);
+ if (IS_ERR(rdev)) {
+ dev_err(&pdev->dev, "Failed to register regulator: %d\n",
+ (int)PTR_ERR(rdev));
+ return -EINVAL;
+ }
machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info),
GFP_KERNEL);
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77203841c535..e359938e3d7e 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio,
switch (asic_type) {
case CHIP_STONEY:
dmadscr[i].xfer_val |=
- BIT(22) |
(ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) |
(size / 2);
break;
default:
dmadscr[i].xfer_val |=
- BIT(22) |
(ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) |
(size / 2);
}
@@ -322,22 +320,87 @@ static void config_acp_dma(void __iomem *acp_mmio,
struct audio_substream_data *rtd,
u32 asic_type)
{
+ u16 ch_acp_sysmem, ch_acp_i2s;
+
acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages,
rtd->pte_offset);
+
+ if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+ ch_acp_sysmem = rtd->ch1;
+ ch_acp_i2s = rtd->ch2;
+ } else {
+ ch_acp_i2s = rtd->ch1;
+ ch_acp_sysmem = rtd->ch2;
+ }
/* Configure System memory <-> ACP SRAM DMA descriptors */
set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size,
rtd->direction, rtd->pte_offset,
- rtd->ch1, rtd->sram_bank,
+ ch_acp_sysmem, rtd->sram_bank,
rtd->dma_dscr_idx_1, asic_type);
/* Configure ACP SRAM <-> I2S DMA descriptors */
set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size,
rtd->direction, rtd->sram_bank,
- rtd->destination, rtd->ch2,
+ rtd->destination, ch_acp_i2s,
rtd->dma_dscr_idx_2, asic_type);
}
+static void acp_dma_cap_channel_enable(void __iomem *acp_mmio,
+ u16 cap_channel)
+{
+ u32 val, ch_reg, imr_reg, res_reg;
+
+ switch (cap_channel) {
+ case CAP_CHANNEL1:
+ ch_reg = mmACP_I2SMICSP_RER1;
+ res_reg = mmACP_I2SMICSP_RCR1;
+ imr_reg = mmACP_I2SMICSP_IMR1;
+ break;
+ case CAP_CHANNEL0:
+ default:
+ ch_reg = mmACP_I2SMICSP_RER0;
+ res_reg = mmACP_I2SMICSP_RCR0;
+ imr_reg = mmACP_I2SMICSP_IMR0;
+ break;
+ }
+ val = acp_reg_read(acp_mmio,
+ mmACP_I2S_16BIT_RESOLUTION_EN);
+ if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) {
+ acp_reg_write(0x0, acp_mmio, ch_reg);
+ /* Set 16bit resolution on capture */
+ acp_reg_write(0x2, acp_mmio, res_reg);
+ }
+ val = acp_reg_read(acp_mmio, imr_reg);
+ val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+ val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+ acp_reg_write(val, acp_mmio, imr_reg);
+ acp_reg_write(0x1, acp_mmio, ch_reg);
+}
+
+static void acp_dma_cap_channel_disable(void __iomem *acp_mmio,
+ u16 cap_channel)
+{
+ u32 val, ch_reg, imr_reg;
+
+ switch (cap_channel) {
+ case CAP_CHANNEL1:
+ imr_reg = mmACP_I2SMICSP_IMR1;
+ ch_reg = mmACP_I2SMICSP_RER1;
+ break;
+ case CAP_CHANNEL0:
+ default:
+ imr_reg = mmACP_I2SMICSP_IMR0;
+ ch_reg = mmACP_I2SMICSP_RER0;
+ break;
+ }
+ val = acp_reg_read(acp_mmio, imr_reg);
+ val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+ val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+ acp_reg_write(val, acp_mmio, imr_reg);
+ acp_reg_write(0x0, acp_mmio, ch_reg);
+}
+
/* Start a given DMA channel transfer */
-static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
+static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular)
{
u32 dma_ctrl;
@@ -356,10 +419,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
switch (ch_num) {
case ACP_TO_I2S_DMA_CH_NUM:
- case ACP_TO_SYSRAM_CH_NUM:
case I2S_TO_ACP_DMA_CH_NUM:
case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM:
- case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM:
case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM:
dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK;
break;
@@ -368,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
break;
}
- /* circular for both DMA channel */
- dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+ /* enable for ACP to SRAM DMA channel */
+ if (is_circular == true)
+ dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+ else
+ dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num);
}
@@ -613,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio)
/* ACP DMA irq handler routine for playback, capture usecases */
static irqreturn_t dma_irq_handler(int irq, void *arg)
{
+ u16 dscr_idx;
u32 intr_flag, ext_intr_status;
struct audio_drv_data *irq_data;
void __iomem *acp_mmio;
@@ -644,32 +709,39 @@ static irqreturn_t dma_irq_handler(int irq, void *arg)
if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) {
valid_irq = true;
+ if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) ==
+ CAPTURE_START_DMA_DESCR_CH15)
+ dscr_idx = CAPTURE_END_DMA_DESCR_CH14;
+ else
+ dscr_idx = CAPTURE_START_DMA_DESCR_CH14;
+ config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx,
+ 1, 0);
+ acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false);
+
snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream);
acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16,
acp_mmio, mmACP_EXTERNAL_INTR_STAT);
}
- if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) {
- valid_irq = true;
- acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16,
- acp_mmio, mmACP_EXTERNAL_INTR_STAT);
- }
-
if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) {
valid_irq = true;
+ if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) ==
+ CAPTURE_START_DMA_DESCR_CH11)
+ dscr_idx = CAPTURE_END_DMA_DESCR_CH10;
+ else
+ dscr_idx = CAPTURE_START_DMA_DESCR_CH10;
+ config_acp_dma_channel(acp_mmio,
+ ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+ dscr_idx, 1, 0);
+ acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+ false);
+
snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream);
acp_reg_write((intr_flag &
BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16,
acp_mmio, mmACP_EXTERNAL_INTR_STAT);
}
- if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) {
- valid_irq = true;
- acp_reg_write((intr_flag &
- BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16,
- acp_mmio, mmACP_EXTERNAL_INTR_STAT);
- }
-
if (valid_irq)
return IRQ_HANDLED;
else
@@ -773,7 +845,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
if (WARN_ON(!rtd))
return -EINVAL;
- rtd->i2s_instance = pinfo->i2s_instance;
+ if (pinfo) {
+ rtd->i2s_instance = pinfo->i2s_instance;
+ rtd->capture_channel = pinfo->capture_channel;
+ }
if (adata->asic_type == CHIP_STONEY) {
val = acp_reg_read(adata->acp_mmio,
mmACP_I2S_16BIT_RESOLUTION_EN);
@@ -841,8 +916,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
switch (rtd->i2s_instance) {
case I2S_BT_INSTANCE:
rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET;
- rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
- rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+ rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+ rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS;
rtd->destination = FROM_BLUETOOTH;
rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10;
@@ -851,13 +926,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH;
rtd->byte_cnt_low_reg_offset =
mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW;
+ rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11;
adata->capture_i2sbt_stream = substream;
break;
case I2S_SP_INSTANCE:
default:
rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET;
- rtd->ch1 = ACP_TO_SYSRAM_CH_NUM;
- rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM;
+ rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM;
+ rtd->ch2 = ACP_TO_SYSRAM_CH_NUM;
switch (adata->asic_type) {
case CHIP_STONEY:
rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET;
@@ -874,6 +950,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
mmACP_I2S_RECEIVED_BYTE_CNT_HIGH;
rtd->byte_cnt_low_reg_offset =
mmACP_I2S_RECEIVED_BYTE_CNT_LOW;
+ rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15;
adata->capture_i2ssp_stream = substream;
}
}
@@ -927,6 +1004,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
u32 buffersize;
u32 pos = 0;
u64 bytescount = 0;
+ u16 dscr;
+ u32 period_bytes, delay;
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_substream_data *rtd = runtime->private_data;
@@ -934,12 +1013,25 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
if (!rtd)
return -EINVAL;
- buffersize = frames_to_bytes(runtime, runtime->buffer_size);
- bytescount = acp_get_byte_count(rtd);
-
- if (bytescount > rtd->bytescount)
- bytescount -= rtd->bytescount;
- pos = do_div(bytescount, buffersize);
+ 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)
+ bytescount -= rtd->bytescount;
+ 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);
+ if (bytescount > rtd->bytescount)
+ bytescount -= rtd->bytescount;
+ pos = do_div(bytescount, buffersize);
+ }
return bytes_to_frames(runtime, pos);
}
@@ -953,16 +1045,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_substream_data *rtd = runtime->private_data;
+ u16 ch_acp_sysmem, ch_acp_i2s;
if (!rtd)
return -EINVAL;
+ if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+ ch_acp_sysmem = rtd->ch1;
+ ch_acp_i2s = rtd->ch2;
+ } else {
+ ch_acp_i2s = rtd->ch1;
+ ch_acp_sysmem = rtd->ch2;
+ }
config_acp_dma_channel(rtd->acp_mmio,
- rtd->ch1,
+ ch_acp_sysmem,
rtd->dma_dscr_idx_1,
NUM_DSCRS_PER_CHANNEL, 0);
config_acp_dma_channel(rtd->acp_mmio,
- rtd->ch2,
+ ch_acp_i2s,
rtd->dma_dscr_idx_2,
NUM_DSCRS_PER_CHANNEL, 0);
return 0;
@@ -971,7 +1071,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret;
- u64 bytescount = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_substream_data *rtd = runtime->private_data;
@@ -982,37 +1081,32 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
- bytescount = acp_get_byte_count(rtd);
- if (rtd->bytescount == 0)
- rtd->bytescount = bytescount;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- acp_dma_start(rtd->acp_mmio, rtd->ch1);
- acp_dma_start(rtd->acp_mmio, rtd->ch2);
+ rtd->bytescount = acp_get_byte_count(rtd);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (rtd->capture_channel == CAP_CHANNEL0) {
+ acp_dma_cap_channel_disable(rtd->acp_mmio,
+ CAP_CHANNEL1);
+ acp_dma_cap_channel_enable(rtd->acp_mmio,
+ CAP_CHANNEL0);
+ }
+ if (rtd->capture_channel == CAP_CHANNEL1) {
+ acp_dma_cap_channel_disable(rtd->acp_mmio,
+ CAP_CHANNEL0);
+ acp_dma_cap_channel_enable(rtd->acp_mmio,
+ CAP_CHANNEL1);
+ }
+ acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
} else {
- acp_dma_start(rtd->acp_mmio, rtd->ch2);
- acp_dma_start(rtd->acp_mmio, rtd->ch1);
+ acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
+ acp_dma_start(rtd->acp_mmio, rtd->ch2, true);
}
ret = 0;
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
- /* For playback, non circular dma should be stopped first
- * i.e Sysram to acp dma transfer channel(rtd->ch1) should be
- * stopped before stopping cirular dma which is acp sram to i2s
- * fifo dma transfer channel(rtd->ch2). Where as in Capture
- * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped
- * first before stopping acp sram to sysram which is circular
- * dma(rtd->ch1).
- */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- acp_dma_stop(rtd->acp_mmio, rtd->ch1);
- ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2);
- } else {
- acp_dma_stop(rtd->acp_mmio, rtd->ch2);
- ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
- }
- rtd->bytescount = 0;
+ acp_dma_stop(rtd->acp_mmio, rtd->ch2);
+ ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
break;
default:
ret = -EINVAL;
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index 9cd3e96c84d4..be3963e8f4fa 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -55,6 +55,8 @@
#define I2S_SP_INSTANCE 0x01
#define I2S_BT_INSTANCE 0x02
+#define CAP_CHANNEL0 0x00
+#define CAP_CHANNEL1 0x01
#define ACP_TILE_ON_MASK 0x03
#define ACP_TILE_OFF_MASK 0x02
@@ -72,16 +74,16 @@
#define ACP_TO_I2S_DMA_CH_NUM 13
/* Capture DMA channels */
-#define ACP_TO_SYSRAM_CH_NUM 14
-#define I2S_TO_ACP_DMA_CH_NUM 15
+#define I2S_TO_ACP_DMA_CH_NUM 14
+#define ACP_TO_SYSRAM_CH_NUM 15
/* Playback DMA Channels for I2S BT instance */
#define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8
#define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9
/* Capture DMA Channels for I2S BT Instance */
-#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10
-#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11
+#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10
+#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11
#define NUM_DSCRS_PER_CHANNEL 2
@@ -125,6 +127,7 @@ struct audio_substream_data {
unsigned int order;
u16 num_of_pages;
u16 i2s_instance;
+ u16 capture_channel;
u16 direction;
u16 ch1;
u16 ch2;
@@ -135,6 +138,7 @@ struct audio_substream_data {
u32 sram_bank;
u32 byte_cnt_high_reg_offset;
u32 byte_cnt_low_reg_offset;
+ u32 dma_curr_dscr;
uint64_t size;
u64 bytescount;
void __iomem *acp_mmio;
@@ -155,6 +159,7 @@ struct audio_drv_data {
*/
struct acp_platform_info {
u16 i2s_instance;
+ u16 capture_channel;
};
union acp_dma_count {
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c
index 5d3b5af9fd92..d88c1d995036 100644
--- a/sound/soc/atmel/atmel-i2s.c
+++ b/sound/soc/atmel/atmel-i2s.c
@@ -206,7 +206,6 @@ struct atmel_i2s_dev {
struct regmap *regmap;
struct clk *pclk;
struct clk *gclk;
- struct clk *aclk;
struct snd_dmaengine_dai_dma_data playback;
struct snd_dmaengine_dai_dma_data capture;
unsigned int fmt;
@@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs)
{
int i, best;
- if (!dev->gclk || !dev->aclk) {
+ if (!dev->gclk) {
dev_err(dev->dev, "cannot generate the I2S Master Clock\n");
return -EINVAL;
}
@@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
bool enabled)
{
unsigned int mr, mr_mask;
- unsigned long aclk_rate;
+ unsigned long gclk_rate;
int ret;
mr = 0;
@@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
/* Disable/unprepare the PMC generated clock. */
clk_disable_unprepare(dev->gclk);
- /* Disable/unprepare the PLL audio clock. */
- clk_disable_unprepare(dev->aclk);
return 0;
}
if (!dev->gck_param)
return -EINVAL;
- aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
+ gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
- /* Fist change the PLL audio clock frequency ... */
- ret = clk_set_rate(dev->aclk, aclk_rate);
+ ret = clk_set_rate(dev->gclk, gclk_rate);
if (ret)
return ret;
- /*
- * ... then set the PMC generated clock rate to the very same frequency
- * to set the gclk parent to aclk.
- */
- ret = clk_set_rate(dev->gclk, aclk_rate);
- if (ret)
- return ret;
-
- /* Prepare and enable the PLL audio clock first ... */
- ret = clk_prepare_enable(dev->aclk);
- if (ret)
- return ret;
-
- /* ... then prepare and enable the PMC generated clock. */
ret = clk_prepare_enable(dev->gclk);
if (ret)
return ret;
@@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev)
return err;
}
- /* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */
- dev->aclk = devm_clk_get(&pdev->dev, "aclk");
+ /* Get audio clock to generate the I2S Master Clock (I2S_MCK) */
dev->gclk = devm_clk_get(&pdev->dev, "gclk");
- if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) {
- if (PTR_ERR(dev->aclk) == -EPROBE_DEFER ||
- PTR_ERR(dev->gclk) == -EPROBE_DEFER)
+ if (IS_ERR(dev->gclk)) {
+ if (PTR_ERR(dev->gclk) == -EPROBE_DEFER)
return -EPROBE_DEFER;
/* Master Mode not supported */
- dev->aclk = NULL;
dev->gclk = NULL;
- } else if (IS_ERR(dev->gclk)) {
- err = PTR_ERR(dev->gclk);
- dev_err(&pdev->dev,
- "failed to get the PMC generated clock: %d\n", err);
- return err;
- } else if (IS_ERR(dev->aclk)) {
- err = PTR_ERR(dev->aclk);
- dev_err(&pdev->dev,
- "failed to get the PLL audio clock: %d\n", err);
- return err;
}
-
dev->dev = &pdev->dev;
dev->regmap = regmap;
platform_set_drvdata(pdev, dev);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 63cf62e9c9aa..efb095dbcd71 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -74,12 +74,12 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DA7219 if I2C
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
- select SND_SOC_DIO2125
select SND_SOC_DMIC if GPIOLIB
select SND_SOC_ES8316 if I2C
select SND_SOC_ES8328_SPI if SPI_MASTER
select SND_SOC_ES8328_I2C if I2C
select SND_SOC_ES7134
+ select SND_SOC_ES7241
select SND_SOC_GTM601
select SND_SOC_HDAC_HDMI
select SND_SOC_ICS43432
@@ -141,8 +141,10 @@ config SND_SOC_ALL_CODECS
select SND_SOC_RT5668 if I2C
select SND_SOC_RT5670 if I2C
select SND_SOC_RT5677 if I2C && SPI_MASTER
+ select SND_SOC_RT5682 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
+ select SND_SOC_SIMPLE_AMPLIFIER
select SND_SOC_SIRF_AUDIO_CODEC
select SND_SOC_SPDIF
select SND_SOC_SSM2305
@@ -572,10 +574,6 @@ config SND_SOC_DA732X
config SND_SOC_DA9055
tristate
-config SND_SOC_DIO2125
- tristate "Dioo DIO2125 Amplifier"
- select GPIOLIB
-
config SND_SOC_DMIC
tristate
@@ -588,6 +586,9 @@ config SND_SOC_HDMI_CODEC
config SND_SOC_ES7134
tristate "Everest Semi ES7134 CODEC"
+config SND_SOC_ES7241
+ tristate "Everest Semi ES7241 CODEC"
+
config SND_SOC_ES8316
tristate "Everest Semi ES8316 CODEC"
depends on I2C
@@ -778,6 +779,7 @@ config SND_SOC_RL6231
default y if SND_SOC_RT5668=y
default y if SND_SOC_RT5670=y
default y if SND_SOC_RT5677=y
+ default y if SND_SOC_RT5682=y
default y if SND_SOC_RT1305=y
default m if SND_SOC_RT5514=m
default m if SND_SOC_RT5616=m
@@ -791,6 +793,7 @@ config SND_SOC_RL6231
default m if SND_SOC_RT5668=m
default m if SND_SOC_RT5670=m
default m if SND_SOC_RT5677=m
+ default m if SND_SOC_RT5682=m
default m if SND_SOC_RT1305=m
config SND_SOC_RL6347A
@@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI
tristate
default SND_SOC_RT5677 && SPI
+config SND_SOC_RT5682
+ tristate
+
#Freescale sgtl5000 codec
config SND_SOC_SGTL5000
tristate "Freescale SGTL5000 CODEC"
@@ -891,6 +897,10 @@ config SND_SOC_SIGMADSP_REGMAP
tristate
select SND_SOC_SIGMADSP
+config SND_SOC_SIMPLE_AMPLIFIER
+ tristate "Simple Audio Amplifier"
+ select GPIOLIB
+
config SND_SOC_SIRF_AUDIO_CODEC
tristate "SiRF SoC internal audio codec"
select REGMAP_MMIO
@@ -953,8 +963,11 @@ config SND_SOC_TAS5086
depends on I2C
config SND_SOC_TAS571X
- tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
+ tristate "Texas Instruments TAS571x power amplifiers"
depends on I2C
+ help
+ Enable support for Texas Instruments TAS5707, TAS5711, TAS5717,
+ TAS5719 and TAS5721 power amplifiers
config SND_SOC_TAS5720
tristate "Texas Instruments TAS5720 Mono Audio amplifier"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index e023fdf85221..7ae7c85e8219 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
snd-soc-dmic-objs := dmic.o
snd-soc-es7134-objs := es7134.o
+snd-soc-es7241-objs := es7241.o
snd-soc-es8316-objs := es8316.o
snd-soc-es8328-objs := es8328.o
snd-soc-es8328-i2c-objs := es8328-i2c.o
@@ -146,6 +147,7 @@ snd-soc-rt5668-objs := rt5668.o
snd-soc-rt5670-objs := rt5670.o
snd-soc-rt5677-objs := rt5677.o
snd-soc-rt5677-spi-objs := rt5677-spi.o
+snd-soc-rt5682-objs := rt5682.o
snd-soc-sgtl5000-objs := sgtl5000.o
snd-soc-alc5623-objs := alc5623.o
snd-soc-alc5632-objs := alc5632.o
@@ -249,9 +251,9 @@ snd-soc-wm9713-objs := wm9713.o
snd-soc-wm-hubs-objs := wm_hubs.o
snd-soc-zx-aud96p22-objs := zx_aud96p22.o
# Amp
-snd-soc-dio2125-objs := dio2125.o
snd-soc-max9877-objs := max9877.o
snd-soc-max98504-objs := max98504.o
+snd-soc-simple-amplifier-objs := simple-amplifier.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
snd-soc-tas2552-objs := tas2552.o
@@ -329,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o
+obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o
obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o
obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o
obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
@@ -405,6 +408,7 @@ obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o
obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o
obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o
obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o
+obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o
obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o
@@ -507,7 +511,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o
obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o
# Amp
-obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o
obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o
obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o
+obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o
obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index ae41edd1c406..57169b8ff14e 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = {
{ "DSP", NULL, "Left Decimator" },
{ "DSP", NULL, "Right Decimator" },
+ { "DSP", NULL, "Playback" },
};
static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index db21ecbe0762..8b9ca7e7a682 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id,
pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
break;
}
+ /* fall through */
default:
return -EINVAL;
}
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 31ec0ba2e639..299ada4dfaa0 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev)
}
#endif /* CONFIG_PM */
-struct snd_soc_component_driver soc_codec_dev_ak4458 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak4458 = {
.probe = ak4458_probe,
.remove = ak4458_remove,
.controls = ak4458_snd_controls,
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index b7ee13406d93..2fa83a1a84cf 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -1,13 +1,8 @@
-/*
- * ak4554.c
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+// ak4554.c
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
#include <linux/module.h>
#include <sound/soc.h>
@@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = {
};
module_platform_driver(ak4554_driver);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("SoC AK4554 driver");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 8523ff9351cf..c1181a20714d 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -1,18 +1,14 @@
-/*
- * ak4613.c -- Asahi Kasei ALSA Soc Audio driver
- *
- * Copyright (C) 2015 Renesas Electronics Corporation
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on ak4642.c by Kuninori Morimoto
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4613.c -- Asahi Kasei ALSA Soc Audio driver
+//
+// Copyright (C) 2015 Renesas Electronics Corporation
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on ak4642.c by Kuninori Morimoto
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
#include <linux/clk.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 605055964529..353237025514 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -1,17 +1,13 @@
-/*
- * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
/* ** CAUTION **
*
@@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver);
MODULE_DESCRIPTION("Soc AK4642 driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index f4ed5cc40661..448bb90c9c8e 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev)
return regcache_sync(ak5558->regmap);
}
-const struct dev_pm_ops ak5558_pm = {
+static const struct dev_pm_ops ak5558_pm = {
SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL)
SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
pm_runtime_force_resume)
};
-struct snd_soc_component_driver soc_codec_dev_ak5558 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak5558 = {
.probe = ak5558_probe,
.remove = ak5558_remove,
.controls = ak5558_snd_controls,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2a7a4168c072..3c266eeb89bf 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg)
{
/* Unreadable registers are considered volatile */
if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
- return 1;
+ return true;
return reg == CS4270_CHIPID;
}
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 196e9c343aeb..45e50fe3bf25 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
};
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -1283,6 +1286,12 @@ static int cs47l24_probe(struct platform_device *pdev)
return ret;
}
+ ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+ if (ret != 0)
+ dev_warn(&pdev->dev,
+ "Failed to set compressed IRQ as a wake source: %d\n",
+ ret);
+
arizona_init_common(arizona);
ret = arizona_init_vol_limit(arizona);
@@ -1306,6 +1315,7 @@ static int cs47l24_probe(struct platform_device *pdev)
err_spk_irqs:
arizona_free_spk_irqs(arizona);
err_dsp_irq:
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
return ret;
@@ -1323,6 +1333,7 @@ static int cs47l24_remove(struct platform_device *pdev)
arizona_free_spk_irqs(arizona);
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
return 0;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 07dd33b09596..ab174b5114dc 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component)
return -ENOMEM;
cx20442->por = regulator_get(component->dev, "POR");
- if (IS_ERR(cx20442->por))
- dev_warn(component->dev, "failed to get the regulator");
+ if (IS_ERR(cx20442->por)) {
+ int err = PTR_ERR(cx20442->por);
+
+ dev_warn(component->dev, "failed to get POR supply (%d)", err);
+ /*
+ * When running on a non-dt platform and requested regulator
+ * is not available, regulator_get() never returns
+ * -EPROBE_DEFER as it is not able to justify if the regulator
+ * may still appear later. On the other hand, the board can
+ * still set full constraints flag at late_initcall in order
+ * to instruct regulator_get() to return a dummy one if
+ * sufficient. Hence, if we get -ENODEV here, let's convert
+ * it to -EPROBE_DEFER and wait for the board to decide or
+ * let Deferred Probe infrastructure handle this error.
+ */
+ if (err == -ENODEV)
+ err = -EPROBE_DEFER;
+ kfree(cx20442);
+ return err;
+ }
+
cx20442->tty = NULL;
snd_soc_component_set_drvdata(component, cx20442);
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index a664111b7184..e172913d04a4 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1,19 +1,14 @@
-/*
- * DA7210 ALSA Soc codec driver
- *
- * Copyright (c) 2009 Dialog Semiconductor
- * Written by David Chen <Dajun.chen@diasemi.com>
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// DA7210 ALSA Soc codec driver
+//
+// Copyright (c) 2009 Dialog Semiconductor
+// Written by David Chen <Dajun.chen@diasemi.com>
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
#include <linux/delay.h>
#include <linux/i2c.h>
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 54cb5f24969f..92d006a5283e 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg)
case DA7213_ALC_OFFSET_AUTO_M_R:
case DA7213_ALC_OFFSET_AUTO_U_R:
case DA7213_ALC_CIC_OP_LVL_DATA:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index a49ab751a036..2c7d5088e6f2 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
container_of(work, struct da7219_aad_priv, btn_det_work);
struct snd_soc_component *component = da7219_aad->component;
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+ struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
u8 statusa, micbias_ctrl;
bool micbias_up = false;
int retries = 0;
@@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
dev_warn(component->dev, "Mic bias status check timed out");
+ da7219->micbias_on_event = true;
+
/*
* Mic bias pulse required to enable mic, must be done before enabling
* button detection to prevent erroneous button readings.
@@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1,
DA7219_BUTTON_CONFIG_MASK, 0);
+ da7219->micbias_on_event = false;
+
/* Disable mic bias */
snd_soc_dapm_disable_pin(dapm, "Mic Bias");
snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 980a6a8bf56d..e46e9f4bc994 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
* DAPM Events
*/
+static int da7219_mic_pga_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);
+ struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (da7219->micbias_on_event) {
+ /*
+ * Delay only for first capture after bias enabled to
+ * avoid possible DC offset related noise.
+ */
+ da7219->micbias_on_event = false;
+ msleep(da7219->mic_pga_delay);
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
static int da7219_dai_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("MIC"),
/* Input PGAs */
- SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
- DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
- NULL, 0),
- SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
- DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
- NULL, 0),
+ SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU),
/* Input Filters */
SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
@@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component)
snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl);
+ /*
+ * Calculate delay required to compensate for DC offset in
+ * Mic PGA, based on Mic Bias voltage.
+ */
+ da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY +
+ (pdata->micbias_lvl *
+ DA7219_MIC_PGA_OFFSET_DELAY);
+
/* Mic */
switch (pdata->mic_amp_in_sel) {
case DA7219_MIC_AMP_IN_SEL_DIFF:
@@ -2143,9 +2175,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg)
case DA7219_ACCDET_IRQ_EVENT_B:
case DA7219_ACCDET_CONFIG_8:
case DA7219_SYSTEM_STATUS:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 1b00023e33cd..3a006862f0e7 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -781,8 +781,10 @@
#define DA7219_SYS_STAT_CHECK_DELAY 50
/* Power up/down Delays */
-#define DA7219_SETTLING_DELAY 40
-#define DA7219_MIN_GAIN_DELAY 30
+#define DA7219_SETTLING_DELAY 40
+#define DA7219_MIN_GAIN_DELAY 30
+#define DA7219_MIC_PGA_BASE_DELAY 100
+#define DA7219_MIC_PGA_OFFSET_DELAY 40
enum da7219_clk_src {
DA7219_CLKSRC_MCLK = 0,
@@ -828,6 +830,8 @@ struct da7219_priv {
bool master;
bool alc_en;
+ bool micbias_on_event;
+ unsigned int mic_pga_delay;
u8 gain_ramp_ctrl;
};
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index afdf90c78884..f6a7bf9560e7 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev,
case DA9055_HP_R_GAIN_STATUS:
case DA9055_LINE_GAIN_STATUS:
case DA9055_ALC_CIC_OP_LVL_DATA:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c
index 58515bb1a303..6d7bca7b78ca 100644
--- a/sound/soc/codecs/es7134.c
+++ b/sound/soc/codecs/es7134.c
@@ -17,6 +17,7 @@
* in the file called COPYING.
*/
+#include <linux/of_platform.h>
#include <linux/module.h>
#include <sound/soc.h>
@@ -24,6 +25,82 @@
* The everest 7134 is a very simple DA converter with no register
*/
+struct es7134_clock_mode {
+ unsigned int rate_min;
+ unsigned int rate_max;
+ unsigned int *mclk_fs;
+ unsigned int mclk_fs_num;
+};
+
+struct es7134_chip {
+ struct snd_soc_dai_driver *dai_drv;
+ const struct es7134_clock_mode *modes;
+ unsigned int mode_num;
+ const struct snd_soc_dapm_widget *extra_widgets;
+ unsigned int extra_widget_num;
+ const struct snd_soc_dapm_route *extra_routes;
+ unsigned int extra_route_num;
+};
+
+struct es7134_data {
+ unsigned int mclk;
+ const struct es7134_chip *chip;
+};
+
+static int es7134_check_mclk(struct snd_soc_dai *dai,
+ struct es7134_data *priv,
+ unsigned int rate)
+{
+ unsigned int mfs = priv->mclk / rate;
+ int i, j;
+
+ for (i = 0; i < priv->chip->mode_num; i++) {
+ const struct es7134_clock_mode *mode = &priv->chip->modes[i];
+
+ if (rate < mode->rate_min || rate > mode->rate_max)
+ continue;
+
+ for (j = 0; j < mode->mclk_fs_num; j++) {
+ if (mode->mclk_fs[j] == mfs)
+ return 0;
+ }
+
+ dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n",
+ mfs, rate);
+ return -EINVAL;
+ }
+
+ /* should not happen */
+ dev_err(dai->dev, "unsupported rate: %u\n", rate);
+ return -EINVAL;
+}
+
+static int es7134_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+ /* mclk has not been provided, assume it is OK */
+ if (!priv->mclk)
+ return 0;
+
+ return es7134_check_mclk(dai, priv, params_rate(params));
+}
+
+static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+ if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+ priv->mclk = freq;
+ return 0;
+ }
+
+ return -ENOTSUPP;
+}
+
static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK |
@@ -38,8 +115,38 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return 0;
}
+static int es7134_component_probe(struct snd_soc_component *c)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c);
+ struct es7134_data *priv = snd_soc_component_get_drvdata(c);
+ const struct es7134_chip *chip = priv->chip;
+ int ret;
+
+ if (chip->extra_widget_num) {
+ ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets,
+ chip->extra_widget_num);
+ if (ret) {
+ dev_err(c->dev, "failed to add extra widgets\n");
+ return ret;
+ }
+ }
+
+ if (chip->extra_route_num) {
+ ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes,
+ chip->extra_route_num);
+ if (ret) {
+ dev_err(c->dev, "failed to add extra routes\n");
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dai_ops es7134_dai_ops = {
.set_fmt = es7134_set_fmt,
+ .hw_params = es7134_hw_params,
+ .set_sysclk = es7134_set_sysclk,
};
static struct snd_soc_dai_driver es7134_dai = {
@@ -48,7 +155,11 @@ static struct snd_soc_dai_driver es7134_dai = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = (SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S18_3LE |
SNDRV_PCM_FMTBIT_S20_3LE |
@@ -58,18 +169,56 @@ static struct snd_soc_dai_driver es7134_dai = {
.ops = &es7134_dai_ops,
};
+static const struct es7134_clock_mode es7134_modes[] = {
+ {
+ /* Single speed mode */
+ .rate_min = 8000,
+ .rate_max = 50000,
+ .mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+ .mclk_fs_num = 5,
+ }, {
+ /* Double speed mode */
+ .rate_min = 84000,
+ .rate_max = 100000,
+ .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 },
+ .mclk_fs_num = 5,
+ }, {
+ /* Quad speed mode */
+ .rate_min = 167000,
+ .rate_max = 192000,
+ .mclk_fs = (unsigned int[]) { 128, 192, 256 },
+ .mclk_fs_num = 3,
+ },
+};
+
+/* Digital I/O are also supplied by VDD on the es7134 */
+static const struct snd_soc_dapm_route es7134_extra_routes[] = {
+ { "Playback", NULL, "VDD", }
+};
+
+static const struct es7134_chip es7134_chip = {
+ .dai_drv = &es7134_dai,
+ .modes = es7134_modes,
+ .mode_num = ARRAY_SIZE(es7134_modes),
+ .extra_routes = es7134_extra_routes,
+ .extra_route_num = ARRAY_SIZE(es7134_extra_routes),
+};
+
static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("AOUTL"),
SND_SOC_DAPM_OUTPUT("AOUTR"),
SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0),
};
static const struct snd_soc_dapm_route es7134_dapm_routes[] = {
{ "AOUTL", NULL, "DAC" },
{ "AOUTR", NULL, "DAC" },
+ { "DAC", NULL, "VDD" },
};
static const struct snd_soc_component_driver es7134_component_driver = {
+ .probe = es7134_component_probe,
.dapm_widgets = es7134_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets),
.dapm_routes = es7134_dapm_routes,
@@ -80,17 +229,87 @@ static const struct snd_soc_component_driver es7134_component_driver = {
.non_legacy_dai_naming = 1,
};
+static struct snd_soc_dai_driver es7154_dai = {
+ .name = "es7154-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S18_3LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &es7134_dai_ops,
+};
+
+static const struct es7134_clock_mode es7154_modes[] = {
+ {
+ /* Single speed mode */
+ .rate_min = 8000,
+ .rate_max = 50000,
+ .mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256,
+ 384, 512, 768, 1024 },
+ .mclk_fs_num = 9,
+ }, {
+ /* Double speed mode */
+ .rate_min = 84000,
+ .rate_max = 100000,
+ .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512,
+ 768, 1024},
+ .mclk_fs_num = 7,
+ }
+};
+
+/* Es7154 has a separate supply for digital I/O */
+static const struct snd_soc_dapm_widget es7154_extra_widgets[] = {
+ SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7154_extra_routes[] = {
+ { "Playback", NULL, "PVDD", }
+};
+
+static const struct es7134_chip es7154_chip = {
+ .dai_drv = &es7154_dai,
+ .modes = es7154_modes,
+ .mode_num = ARRAY_SIZE(es7154_modes),
+ .extra_routes = es7154_extra_routes,
+ .extra_route_num = ARRAY_SIZE(es7154_extra_routes),
+ .extra_widgets = es7154_extra_widgets,
+ .extra_widget_num = ARRAY_SIZE(es7154_extra_widgets),
+};
+
static int es7134_probe(struct platform_device *pdev)
{
+ struct device *dev = &pdev->dev;
+ struct es7134_data *priv;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, priv);
+
+ priv->chip = of_device_get_match_data(dev);
+ if (!priv->chip) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
return devm_snd_soc_register_component(&pdev->dev,
&es7134_component_driver,
- &es7134_dai, 1);
+ priv->chip->dai_drv, 1);
}
#ifdef CONFIG_OF
static const struct of_device_id es7134_ids[] = {
- { .compatible = "everest,es7134", },
- { .compatible = "everest,es7144", },
+ { .compatible = "everest,es7134", .data = &es7134_chip },
+ { .compatible = "everest,es7144", .data = &es7134_chip },
+ { .compatible = "everest,es7154", .data = &es7154_chip },
{ }
};
MODULE_DEVICE_TABLE(of, es7134_ids);
diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c
new file mode 100644
index 000000000000..87991bd4acef
--- /dev/null
+++ b/sound/soc/codecs/es7241.c
@@ -0,0 +1,322 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/gpio/consumer.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+struct es7241_clock_mode {
+ unsigned int rate_min;
+ unsigned int rate_max;
+ unsigned int *slv_mfs;
+ unsigned int slv_mfs_num;
+ unsigned int mst_mfs;
+ unsigned int mst_m0:1;
+ unsigned int mst_m1:1;
+};
+
+struct es7241_chip {
+ const struct es7241_clock_mode *modes;
+ unsigned int mode_num;
+};
+
+struct es7241_data {
+ struct gpio_desc *reset;
+ struct gpio_desc *m0;
+ struct gpio_desc *m1;
+ unsigned int fmt;
+ unsigned int mclk;
+ bool is_slave;
+ const struct es7241_chip *chip;
+};
+
+static void es7241_set_mode(struct es7241_data *priv, int m0, int m1)
+{
+ /* put the device in reset */
+ gpiod_set_value_cansleep(priv->reset, 0);
+
+ /* set the mode */
+ gpiod_set_value_cansleep(priv->m0, m0);
+ gpiod_set_value_cansleep(priv->m1, m1);
+
+ /* take the device out of reset - datasheet does not specify a delay */
+ gpiod_set_value_cansleep(priv->reset, 1);
+}
+
+static int es7241_set_slave_mode(struct es7241_data *priv,
+ const struct es7241_clock_mode *mode,
+ unsigned int mfs)
+{
+ int j;
+
+ if (!mfs)
+ goto out_ok;
+
+ for (j = 0; j < mode->slv_mfs_num; j++) {
+ if (mode->slv_mfs[j] == mfs)
+ goto out_ok;
+ }
+
+ return -EINVAL;
+
+out_ok:
+ es7241_set_mode(priv, 1, 1);
+ return 0;
+}
+
+static int es7241_set_master_mode(struct es7241_data *priv,
+ const struct es7241_clock_mode *mode,
+ unsigned int mfs)
+{
+ /*
+ * We can't really set clock ratio, if the mclk/lrclk is different
+ * from what we provide, then error out
+ */
+ if (mfs && mfs != mode->mst_mfs)
+ return -EINVAL;
+
+ es7241_set_mode(priv, mode->mst_m0, mode->mst_m1);
+
+ return 0;
+}
+
+static int es7241_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned int rate = params_rate(params);
+ unsigned int mfs = priv->mclk / rate;
+ int i;
+
+ for (i = 0; i < priv->chip->mode_num; i++) {
+ const struct es7241_clock_mode *mode = &priv->chip->modes[i];
+
+ if (rate < mode->rate_min || rate >= mode->rate_max)
+ continue;
+
+ if (priv->is_slave)
+ return es7241_set_slave_mode(priv, mode, mfs);
+ else
+ return es7241_set_master_mode(priv, mode, mfs);
+ }
+
+ /* should not happen */
+ dev_err(dai->dev, "unsupported rate: %u\n", rate);
+ return -EINVAL;
+}
+
+static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+ if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+ priv->mclk = freq;
+ return 0;
+ }
+
+ return -ENOTSUPP;
+}
+
+static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+ dev_err(dai->dev, "Unsupported dai clock inversion\n");
+ return -EINVAL;
+ }
+
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) {
+ dev_err(dai->dev, "Invalid dai format\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ priv->is_slave = true;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ priv->is_slave = false;
+ break;
+
+ default:
+ dev_err(dai->dev, "Unsupported clock configuration\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops es7241_dai_ops = {
+ .set_fmt = es7241_set_fmt,
+ .hw_params = es7241_hw_params,
+ .set_sysclk = es7241_set_sysclk,
+};
+
+static struct snd_soc_dai_driver es7241_dai = {
+ .name = "es7241-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &es7241_dai_ops,
+};
+
+static const struct es7241_clock_mode es7241_modes[] = {
+ {
+ /* Single speed mode */
+ .rate_min = 8000,
+ .rate_max = 50000,
+ .slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+ .slv_mfs_num = 5,
+ .mst_mfs = 256,
+ .mst_m0 = 0,
+ .mst_m1 = 0,
+ }, {
+ /* Double speed mode */
+ .rate_min = 50000,
+ .rate_max = 100000,
+ .slv_mfs = (unsigned int[]) { 128, 192 },
+ .slv_mfs_num = 2,
+ .mst_mfs = 128,
+ .mst_m0 = 1,
+ .mst_m1 = 0,
+ }, {
+ /* Quad speed mode */
+ .rate_min = 100000,
+ .rate_max = 200000,
+ .slv_mfs = (unsigned int[]) { 64 },
+ .slv_mfs_num = 1,
+ .mst_mfs = 64,
+ .mst_m0 = 0,
+ .mst_m1 = 1,
+ },
+};
+
+static const struct es7241_chip es7241_chip = {
+ .modes = es7241_modes,
+ .mode_num = ARRAY_SIZE(es7241_modes),
+};
+
+static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("AINL"),
+ SND_SOC_DAPM_INPUT("AINR"),
+ SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7241_dapm_routes[] = {
+ { "ADC", NULL, "AINL", },
+ { "ADC", NULL, "AINR", },
+ { "ADC", NULL, "VDDA", },
+ { "Capture", NULL, "VDDP", },
+ { "Capture", NULL, "VDDD", },
+};
+
+static const struct snd_soc_component_driver es7241_component_driver = {
+ .dapm_widgets = es7241_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(es7241_dapm_widgets),
+ .dapm_routes = es7241_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes),
+ .idle_bias_on = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv)
+{
+ bool is_leftj;
+
+ /*
+ * The format is given by a pull resistor on the SDOUT pin:
+ * pull-up for i2s, pull-down for left justified.
+ */
+ is_leftj = of_property_read_bool(dev->of_node,
+ "everest,sdout-pull-down");
+ if (is_leftj)
+ priv->fmt = SND_SOC_DAIFMT_LEFT_J;
+ else
+ priv->fmt = SND_SOC_DAIFMT_I2S;
+}
+
+static int es7241_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct es7241_data *priv;
+ int err;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, priv);
+
+ priv->chip = of_device_get_match_data(dev);
+ if (!priv->chip) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
+ es7241_parse_fmt(dev, priv);
+
+ priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW);
+ if (IS_ERR(priv->reset)) {
+ err = PTR_ERR(priv->reset);
+ if (err != -EPROBE_DEFER)
+ dev_err(dev, "Failed to get 'reset' gpio: %d", err);
+ return err;
+ }
+
+ priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW);
+ if (IS_ERR(priv->m0)) {
+ err = PTR_ERR(priv->m0);
+ if (err != -EPROBE_DEFER)
+ dev_err(dev, "Failed to get 'm0' gpio: %d", err);
+ return err;
+ }
+
+ priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW);
+ if (IS_ERR(priv->m1)) {
+ err = PTR_ERR(priv->m1);
+ if (err != -EPROBE_DEFER)
+ dev_err(dev, "Failed to get 'm1' gpio: %d", err);
+ return err;
+ }
+
+ return devm_snd_soc_register_component(&pdev->dev,
+ &es7241_component_driver,
+ &es7241_dai, 1);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id es7241_ids[] = {
+ { .compatible = "everest,es7241", .data = &es7241_chip },
+ { }
+};
+MODULE_DEVICE_TABLE(of, es7241_ids);
+#endif
+
+static struct platform_driver es7241_driver = {
+ .driver = {
+ .name = "es7241",
+ .of_match_table = of_match_ptr(es7241_ids),
+ },
+ .probe = es7241_probe,
+};
+
+module_platform_driver(es7241_driver);
+
+MODULE_DESCRIPTION("ASoC ES7241 audio codec driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 4748a9d5de3b..7b8533abf637 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -2093,6 +2093,75 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
}
#ifdef CONFIG_PM
+/*
+ * Power management sequences
+ * ==========================
+ *
+ * The following explains the PM handling of HDAC HDMI with its parent
+ * device SKL and display power usage
+ *
+ * Probe
+ * -----
+ * In SKL probe,
+ * 1. skl_probe_work() powers up the display (refcount++ -> 1)
+ * 2. enumerates the codecs on the link
+ * 3. powers down the display (refcount-- -> 0)
+ *
+ * In HDAC HDMI probe,
+ * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1)
+ * 2. probe the codec
+ * 3. put the HDAC HDMI device to runtime suspend
+ * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Once children are runtime suspended, SKL device also goes to runtime
+ * suspend
+ *
+ * HDMI Playback
+ * -------------
+ * Open HDMI device,
+ * 1. skl_runtime_resume() invoked
+ * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ *
+ * Close HDMI device,
+ * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 2. skl_runtime_suspend() invoked
+ *
+ * S0/S3 Cycle with playback in progress
+ * -------------------------------------
+ * When the device is opened for playback, the device is runtime active
+ * already and the display refcount is 1 as explained above.
+ *
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just
+ * increments the PM runtime usage count of the codec since the device
+ * is in use already
+ * 2. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just
+ * decrements the PM runtime usage count of the codec since the device
+ * is in use already
+ *
+ * Once playback is stopped, the display refcount is set to 0 as explained
+ * above in the HDMI playback sequence. The PM handlings are designed in
+ * such way that to balance the refcount of display power when the codec
+ * device put to S3 while playback is going on.
+ *
+ * S0/S3 Cycle without playback in progress
+ * ----------------------------------------
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec
+ * 2. skl_runtime_resume() invoked
+ * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ * 4. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec
+ * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 4. skl_runtime_suspend() invoked
+ */
static int hdac_hdmi_runtime_suspend(struct device *dev)
{
struct hdac_device *hdev = dev_to_hdac_dev(dev);
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 38e4a8515709..d00734d31e04 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = {
SND_SOC_DAPM_OUTPUT("TX"),
};
-static const struct snd_soc_dapm_route hdmi_routes[] = {
- { "TX", NULL, "Playback" },
-};
-
enum {
DAI_ID_I2S = 0,
DAI_ID_SPDIF,
@@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
return snd_ctl_add(rtd->card->snd_card, kctl);
}
+static int hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_dapm_route route = {
+ .sink = "TX",
+ .source = dai->driver->playback.stream_name,
+ };
+
+ dapm = snd_soc_component_get_dapm(dai->component);
+
+ return snd_soc_dapm_add_routes(dapm, &route, 1);
+}
+
static const struct snd_soc_dai_driver hdmi_i2s_dai = {
.name = "i2s-hifi",
.id = DAI_ID_I2S,
+ .probe = hdmi_dai_probe,
.playback = {
.stream_name = "I2S Playback",
.channels_min = 2,
@@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = {
static const struct snd_soc_dai_driver hdmi_spdif_dai = {
.name = "spdif-hifi",
.id = DAI_ID_SPDIF,
+ .probe = hdmi_dai_probe,
.playback = {
.stream_name = "SPDIF Playback",
.channels_min = 2,
@@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
static const struct snd_soc_component_driver hdmi_driver = {
.dapm_widgets = hdmi_widgets,
.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
- .dapm_routes = hdmi_routes,
- .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
.of_xlate_dai_id = hdmi_of_xlate_dai_id,
.idle_bias_on = 1,
.use_pmdown_time = 1,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index a92586106932..92b7125ea169 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
static bool max98373_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
+ case MAX98373_R2000_SW_RESET:
case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3:
case MAX98373_R2010_IRQ_CTRL:
case MAX98373_R2014_THERM_WARN_THRESH
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 74d7f52c7e73..6e6134589588 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg)
switch (reg) {
case MAX9850_STATUSA:
case MAX9850_STATUSB:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index 17104f8dc1a9..e3c8cd17daf2 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = {
static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr)
{
- int osrate;
-
if (osr >= ARRAY_SIZE(osr_adc_sel))
return -EINVAL;
- osrate = osr_adc_sel[osr].osr;
if (rate * osr > CLK_ADC_MAX) {
dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n");
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 6bd14453f06e..468d5143e2c4 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in,
fvco_max = 0;
fvco_sel = ARRAY_SIZE(mclk_src_scaling);
for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
- fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+ fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
fvco_max < fvco) {
fvco_max = fvco;
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index dc6ea4987b7d..b9fed99d8b5e 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs,
fvco_max = 0;
fvco_sel = ARRAY_SIZE(mclk_src_scaling);
for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
- fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+ fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
fvco_max < fvco) {
fvco_max = fvco;
diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c
index 21f15219b3ad..8df6447c76a6 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev)
{
struct pcm1789_private *priv = dev_get_drvdata(dev);
- if (&priv->work)
- flush_work(&priv->work);
+ flush_work(&priv->work);
return 0;
}
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 88fde70b1e9e..690c26e7389e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component);
unsigned int rate = params_rate(params);
- unsigned int format = params_format(params);
+ snd_pcm_format_t format = params_format(params);
unsigned int width = params_width(params);
unsigned int channels = params_channels(params);
unsigned int div_lrck;
diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c
index f4c8c45f4010..c4452efc7970 100644
--- a/sound/soc/codecs/rt1305.c
+++ b/sound/soc/codecs/rt1305.c
@@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
- r0l = 562949953421312;
+ r0l = 562949953421312ULL;
if (rhl != 0)
do_div(r0l, rhl);
pr_debug("Left_r0 = 0x%llx\n", r0l);
@@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
- r0r = 562949953421312;
+ r0r = 562949953421312ULL;
if (rhl != 0)
do_div(r0r, rhl);
pr_debug("Right_r0 = 0x%llx\n", r0r);
@@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c,
rt1305_reset(rt1305->regmap);
rt1305_calibrate(rt1305);
- return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305,
+ return devm_snd_soc_register_component(&i2c->dev,
+ &soc_component_dev_rt1305,
rt1305_dai, ARRAY_SIZE(rt1305_dai));
}
-static int rt1305_i2c_remove(struct i2c_client *i2c)
-{
- snd_soc_unregister_component(&i2c->dev);
-
- return 0;
-}
-
static void rt1305_i2c_shutdown(struct i2c_client *client)
{
struct rt1305_priv *rt1305 = i2c_get_clientdata(client);
@@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = {
#endif
},
.probe = rt1305_i2c_probe,
- .remove = rt1305_i2c_remove,
.shutdown = rt1305_i2c_shutdown,
.id_table = rt1305_i2c_id,
};
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1570b91bf018..dca82dd6e3bf 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = {
{RT5514_ANA_CTRL_LDO10, 0x00028604},
{RT5514_ANA_CTRL_ADCFED, 0x00000800},
{RT5514_ASRC_IN_CTRL1, 0x00000003},
- {RT5514_DOWNFILTER0_CTRL3, 0x10000362},
- {RT5514_DOWNFILTER1_CTRL3, 0x10000362},
+ {RT5514_DOWNFILTER0_CTRL3, 0x10000352},
+ {RT5514_DOWNFILTER1_CTRL3, 0x10000352},
};
static const struct reg_default rt5514_reg[] = {
@@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = {
{RT5514_ASRC_IN_CTRL1, 0x00000003},
{RT5514_DOWNFILTER0_CTRL1, 0x00020c2f},
{RT5514_DOWNFILTER0_CTRL2, 0x00020c2f},
- {RT5514_DOWNFILTER0_CTRL3, 0x10000362},
+ {RT5514_DOWNFILTER0_CTRL3, 0x10000352},
{RT5514_DOWNFILTER1_CTRL1, 0x00020c2f},
{RT5514_DOWNFILTER1_CTRL2, 0x00020c2f},
- {RT5514_DOWNFILTER1_CTRL3, 0x10000362},
+ {RT5514_DOWNFILTER1_CTRL3, 0x10000352},
{RT5514_ANA_CTRL_LDO10, 0x00028604},
{RT5514_ANA_CTRL_LDO18_16, 0x02000345},
{RT5514_ANA_CTRL_ADC12, 0x0000a2a8},
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index cf6dce69eb2a..865f49ac38dd 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
case RT5631_INDEX_ADD:
case RT5631_INDEX_DATA:
case RT5631_EQ_CTRL:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
@@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg)
case RT5631_VENDOR_ID:
case RT5631_VENDOR_ID1:
case RT5631_VENDOR_ID2:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
@@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
static const struct snd_kcontrol_new rt5631_snd_controls[] = {
/* MIC */
SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum),
- SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2,
+ SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2,
RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv),
SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum),
- SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2,
+ SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2,
RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv),
/* MONO IN */
SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum),
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 8bf8d360c25f..27770143ae8f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
break;
case RT5640_IF_113:
ret |= RT5640_U_IF1;
+ /* fall through */
case RT5640_IF_312:
case RT5640_IF_213:
ret |= RT5640_U_IF2;
@@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
break;
case RT5640_IF_223:
ret |= RT5640_U_IF1;
+ /* fall through */
case RT5640_IF_123:
case RT5640_IF_321:
ret |= RT5640_U_IF2;
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 6b5669f3e85d..985852fd9723 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = {
SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL,
RT5651_L_VOL_SFT, RT5651_R_VOL_SFT,
175, 0, dac_vol_tlv),
- /* IN1/IN2 Control */
+ /* IN1/IN2/IN3 Control */
SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2,
RT5651_BST_SFT1, 8, 0, bst_tlv),
SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2,
RT5651_BST_SFT2, 8, 0, bst_tlv),
+ SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3,
+ RT5651_BST_SFT1, 8, 0, bst_tlv),
/* INL/INR Volume Control */
SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL,
RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT,
@@ -1581,6 +1583,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component
snd_soc_dapm_mutex_unlock(dapm);
}
+static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+ struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+ snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+ RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR);
+ rt5651->ovcd_irq_enabled = true;
+}
+
+static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+ struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+ snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+ RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP);
+ rt5651->ovcd_irq_enabled = false;
+}
+
static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component)
{
snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
@@ -1622,10 +1642,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component)
return val == 0;
}
-/* Jack detect timings */
+/* Jack detect and button-press timings */
#define JACK_SETTLE_TIME 100 /* milli seconds */
#define JACK_DETECT_COUNT 5
#define JACK_DETECT_MAXCOUNT 20 /* Aprox. 2 seconds worth of tries */
+#define JACK_UNPLUG_TIME 80 /* milli seconds */
+#define BP_POLL_TIME 10 /* milli seconds */
+#define BP_POLL_MAXCOUNT 200 /* assume something is wrong after this */
+#define BP_THRESHOLD 3
+
+static void rt5651_start_button_press_work(struct snd_soc_component *component)
+{
+ struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+ rt5651->poll_count = 0;
+ rt5651->press_count = 0;
+ rt5651->release_count = 0;
+ rt5651->pressed = false;
+ rt5651->press_reported = false;
+ rt5651_clear_micbias1_ovcd(component);
+ schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
+
+static void rt5651_button_press_work(struct work_struct *work)
+{
+ struct rt5651_priv *rt5651 =
+ container_of(work, struct rt5651_priv, bp_work.work);
+ struct snd_soc_component *component = rt5651->component;
+
+ /* Check the jack was not removed underneath us */
+ if (!rt5651_jack_inserted(component))
+ return;
+
+ if (rt5651_micbias1_ovcd(component)) {
+ rt5651->release_count = 0;
+ rt5651->press_count++;
+ /* Remember till after JACK_UNPLUG_TIME wait */
+ if (rt5651->press_count >= BP_THRESHOLD)
+ rt5651->pressed = true;
+ rt5651_clear_micbias1_ovcd(component);
+ } else {
+ rt5651->press_count = 0;
+ rt5651->release_count++;
+ }
+
+ /*
+ * The pins get temporarily shorted on jack unplug, so we poll for
+ * at least JACK_UNPLUG_TIME milli-seconds before reporting a press.
+ */
+ rt5651->poll_count++;
+ if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) {
+ schedule_delayed_work(&rt5651->bp_work,
+ msecs_to_jiffies(BP_POLL_TIME));
+ return;
+ }
+
+ if (rt5651->pressed && !rt5651->press_reported) {
+ dev_dbg(component->dev, "headset button press\n");
+ snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0,
+ SND_JACK_BTN_0);
+ rt5651->press_reported = true;
+ }
+
+ if (rt5651->release_count >= BP_THRESHOLD) {
+ if (rt5651->press_reported) {
+ dev_dbg(component->dev, "headset button release\n");
+ snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+ }
+ /* Re-enable OVCD IRQ to detect next press */
+ rt5651_enable_micbias1_ovcd_irq(component);
+ return; /* Stop polling */
+ }
+
+ schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
static int rt5651_detect_headset(struct snd_soc_component *component)
{
@@ -1676,15 +1766,58 @@ static void rt5651_jack_detect_work(struct work_struct *work)
{
struct rt5651_priv *rt5651 =
container_of(work, struct rt5651_priv, jack_detect_work);
+ struct snd_soc_component *component = rt5651->component;
int report = 0;
- if (rt5651_jack_inserted(rt5651->component)) {
- rt5651_enable_micbias1_for_ovcd(rt5651->component);
- report = rt5651_detect_headset(rt5651->component);
- rt5651_disable_micbias1_for_ovcd(rt5651->component);
+ if (!rt5651_jack_inserted(component)) {
+ /* Jack removed, or spurious IRQ? */
+ if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) {
+ if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+ cancel_delayed_work_sync(&rt5651->bp_work);
+ rt5651_disable_micbias1_ovcd_irq(component);
+ rt5651_disable_micbias1_for_ovcd(component);
+ }
+ snd_soc_jack_report(rt5651->hp_jack, 0,
+ SND_JACK_HEADSET | SND_JACK_BTN_0);
+ dev_dbg(component->dev, "jack unplugged\n");
+ }
+ } else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) {
+ /* Jack inserted */
+ WARN_ON(rt5651->ovcd_irq_enabled);
+ rt5651_enable_micbias1_for_ovcd(component);
+ report = rt5651_detect_headset(component);
+ if (report == SND_JACK_HEADSET) {
+ /* Enable ovcd IRQ for button press detect. */
+ rt5651_enable_micbias1_ovcd_irq(component);
+ } else {
+ /* No more need for overcurrent detect. */
+ rt5651_disable_micbias1_for_ovcd(component);
+ }
+ dev_dbg(component->dev, "detect report %#02x\n", report);
+ snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
+ } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) {
+ dev_dbg(component->dev, "OVCD IRQ\n");
+
+ /*
+ * The ovcd IRQ keeps firing while the button is pressed, so
+ * we disable it and start polling the button until released.
+ *
+ * The disable will make the IRQ pin 0 again and since we get
+ * IRQs on both edges (so as to detect both jack plugin and
+ * unplug) this means we will immediately get another IRQ.
+ * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP.
+ */
+ rt5651_disable_micbias1_ovcd_irq(component);
+ rt5651_start_button_press_work(component);
+
+ /*
+ * If the jack-detect IRQ flag goes high (unplug) after our
+ * above rt5651_jack_inserted() check and before we have
+ * disabled the OVCD IRQ, the IRQ pin will stay high and as
+ * we react to edges, we miss the unplug event -> recheck.
+ */
+ queue_work(system_long_wq, &rt5651->jack_detect_work);
}
-
- snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
}
static irqreturn_t rt5651_irq(int irq, void *data)
@@ -1696,14 +1829,18 @@ static irqreturn_t rt5651_irq(int irq, void *data)
return IRQ_HANDLED;
}
-static int rt5651_set_jack(struct snd_soc_component *component,
- struct snd_soc_jack *hp_jack, void *data)
+static void rt5651_cancel_work(void *data)
{
- struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
- int ret;
+ struct rt5651_priv *rt5651 = data;
- if (!rt5651->irq)
- return -EINVAL;
+ cancel_work_sync(&rt5651->jack_detect_work);
+ cancel_delayed_work_sync(&rt5651->bp_work);
+}
+
+static void rt5651_enable_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *hp_jack)
+{
+ struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
/* IRQ output on GPIO1 */
snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1,
@@ -1730,10 +1867,10 @@ static int rt5651_set_jack(struct snd_soc_component *component,
RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN);
break;
case RT5651_JD_NULL:
- return 0;
+ return;
default:
dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n");
- return -EINVAL;
+ return;
}
/* Enable jack detect power */
@@ -1767,19 +1904,39 @@ static int rt5651_set_jack(struct snd_soc_component *component,
RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN);
rt5651->hp_jack = hp_jack;
-
- ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL,
- rt5651_irq,
- IRQF_TRIGGER_RISING |
- IRQF_TRIGGER_FALLING |
- IRQF_ONESHOT, "rt5651", rt5651);
- if (ret) {
- dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret);
- return ret;
+ if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+ rt5651_enable_micbias1_for_ovcd(component);
+ rt5651_enable_micbias1_ovcd_irq(component);
}
+ enable_irq(rt5651->irq);
/* sync initial jack state */
queue_work(system_power_efficient_wq, &rt5651->jack_detect_work);
+}
+
+static void rt5651_disable_jack_detect(struct snd_soc_component *component)
+{
+ struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+ disable_irq(rt5651->irq);
+ rt5651_cancel_work(rt5651);
+
+ if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+ rt5651_disable_micbias1_ovcd_irq(component);
+ rt5651_disable_micbias1_for_ovcd(component);
+ snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+ }
+
+ rt5651->hp_jack = NULL;
+}
+
+static int rt5651_set_jack(struct snd_soc_component *component,
+ struct snd_soc_jack *jack, void *data)
+{
+ if (jack)
+ rt5651_enable_jack_detect(component, jack);
+ else
+ rt5651_disable_jack_detect(component);
return 0;
}
@@ -2034,8 +2191,26 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
rt5651->irq = i2c->irq;
rt5651->hp_mute = 1;
+ INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work);
INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work);
+ /* Make sure work is stopped on probe-error / remove */
+ ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651);
+ if (ret)
+ return ret;
+
+ ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq,
+ IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+ | IRQF_ONESHOT, "rt5651", rt5651);
+ if (ret == 0) {
+ /* Gets re-enabled by rt5651_set_jack() */
+ disable_irq(rt5651->irq);
+ } else {
+ dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n",
+ rt5651->irq, ret);
+ rt5651->irq = -ENXIO;
+ }
+
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_rt5651,
rt5651_dai, ARRAY_SIZE(rt5651_dai));
@@ -2043,15 +2218,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
return ret;
}
-static int rt5651_i2c_remove(struct i2c_client *i2c)
-{
- struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c);
-
- cancel_work_sync(&rt5651->jack_detect_work);
-
- return 0;
-}
-
static struct i2c_driver rt5651_i2c_driver = {
.driver = {
.name = "rt5651",
@@ -2059,7 +2225,6 @@ static struct i2c_driver rt5651_i2c_driver = {
.of_match_table = of_match_ptr(rt5651_of_match),
},
.probe = rt5651_i2c_probe,
- .remove = rt5651_i2c_remove,
.id_table = rt5651_i2c_id,
};
module_i2c_driver(rt5651_i2c_driver);
diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h
index 3a0968c53fde..ac6de6fb5414 100644
--- a/sound/soc/codecs/rt5651.h
+++ b/sound/soc/codecs/rt5651.h
@@ -2071,8 +2071,16 @@ struct rt5651_pll_code {
struct rt5651_priv {
struct snd_soc_component *component;
struct regmap *regmap;
+ /* Jack and button detect data */
struct snd_soc_jack *hp_jack;
struct work_struct jack_detect_work;
+ struct delayed_work bp_work;
+ bool ovcd_irq_enabled;
+ bool pressed;
+ bool press_reported;
+ int press_count;
+ int release_count;
+ int poll_count;
unsigned int jd_src;
unsigned int ovcd_th;
unsigned int ovcd_sf;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 8a0181a2db08..9b7a1833d331 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
break;
case 25:
slot_width_25 = 0x8080;
+ /* fall through */
case 24:
val |= (2 << 8);
break;
@@ -5007,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = {
};
static const struct of_device_id rt5677_of_match[] = {
- { .compatible = "realtek,rt5677", RT5677 },
+ { .compatible = "realtek,rt5677", .data = (const void *)RT5677 },
{ }
};
MODULE_DEVICE_TABLE(of, rt5677_of_match);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
new file mode 100644
index 000000000000..640d400ca013
--- /dev/null
+++ b/sound/soc/codecs/rt5682.c
@@ -0,0 +1,2681 @@
+/*
+ * rt5682.c -- RT5682 ALSA SoC audio component driver
+ *
+ * Copyright 2018 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/acpi.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/rt5682.h>
+
+#include "rl6231.h"
+#include "rt5682.h"
+
+#define RT5682_NUM_SUPPLIES 3
+
+static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = {
+ "AVDD",
+ "MICVDD",
+ "VBAT",
+};
+
+struct rt5682_priv {
+ struct snd_soc_component *component;
+ struct rt5682_platform_data pdata;
+ struct regmap *regmap;
+ struct snd_soc_jack *hs_jack;
+ struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
+ struct delayed_work jack_detect_work;
+ struct delayed_work jd_check_work;
+ struct mutex calibrate_mutex;
+
+ int sysclk;
+ int sysclk_src;
+ int lrck[RT5682_AIFS];
+ int bclk[RT5682_AIFS];
+ int master[RT5682_AIFS];
+
+ int pll_src;
+ int pll_in;
+ int pll_out;
+
+ int jack_type;
+};
+
+static const struct reg_sequence patch_list[] = {
+ {0x01c1, 0x1000},
+};
+
+static const struct reg_default rt5682_reg[] = {
+ {0x0002, 0x8080},
+ {0x0003, 0x8000},
+ {0x0005, 0x0000},
+ {0x0006, 0x0000},
+ {0x0008, 0x800f},
+ {0x000b, 0x0000},
+ {0x0010, 0x4040},
+ {0x0011, 0x0000},
+ {0x0012, 0x1404},
+ {0x0013, 0x1000},
+ {0x0014, 0xa00a},
+ {0x0015, 0x0404},
+ {0x0016, 0x0404},
+ {0x0019, 0xafaf},
+ {0x001c, 0x2f2f},
+ {0x001f, 0x0000},
+ {0x0022, 0x5757},
+ {0x0023, 0x0039},
+ {0x0024, 0x000b},
+ {0x0026, 0xc0c4},
+ {0x0029, 0x8080},
+ {0x002a, 0xa0a0},
+ {0x002b, 0x0300},
+ {0x0030, 0x0000},
+ {0x003c, 0x0080},
+ {0x0044, 0x0c0c},
+ {0x0049, 0x0000},
+ {0x0061, 0x0000},
+ {0x0062, 0x0000},
+ {0x0063, 0x003f},
+ {0x0064, 0x0000},
+ {0x0065, 0x0000},
+ {0x0066, 0x0030},
+ {0x0067, 0x0000},
+ {0x006b, 0x0000},
+ {0x006c, 0x0000},
+ {0x006d, 0x2200},
+ {0x006e, 0x0a10},
+ {0x0070, 0x8000},
+ {0x0071, 0x8000},
+ {0x0073, 0x0000},
+ {0x0074, 0x0000},
+ {0x0075, 0x0002},
+ {0x0076, 0x0001},
+ {0x0079, 0x0000},
+ {0x007a, 0x0000},
+ {0x007b, 0x0000},
+ {0x007c, 0x0100},
+ {0x007e, 0x0000},
+ {0x0080, 0x0000},
+ {0x0081, 0x0000},
+ {0x0082, 0x0000},
+ {0x0083, 0x0000},
+ {0x0084, 0x0000},
+ {0x0085, 0x0000},
+ {0x0086, 0x0005},
+ {0x0087, 0x0000},
+ {0x0088, 0x0000},
+ {0x008c, 0x0003},
+ {0x008d, 0x0000},
+ {0x008e, 0x0060},
+ {0x008f, 0x1000},
+ {0x0091, 0x0c26},
+ {0x0092, 0x0073},
+ {0x0093, 0x0000},
+ {0x0094, 0x0080},
+ {0x0098, 0x0000},
+ {0x009a, 0x0000},
+ {0x009b, 0x0000},
+ {0x009c, 0x0000},
+ {0x009d, 0x0000},
+ {0x009e, 0x100c},
+ {0x009f, 0x0000},
+ {0x00a0, 0x0000},
+ {0x00a3, 0x0002},
+ {0x00a4, 0x0001},
+ {0x00ae, 0x2040},
+ {0x00af, 0x0000},
+ {0x00b6, 0x0000},
+ {0x00b7, 0x0000},
+ {0x00b8, 0x0000},
+ {0x00b9, 0x0002},
+ {0x00be, 0x0000},
+ {0x00c0, 0x0160},
+ {0x00c1, 0x82a0},
+ {0x00c2, 0x0000},
+ {0x00d0, 0x0000},
+ {0x00d1, 0x2244},
+ {0x00d2, 0x3300},
+ {0x00d3, 0x2200},
+ {0x00d4, 0x0000},
+ {0x00d9, 0x0009},
+ {0x00da, 0x0000},
+ {0x00db, 0x0000},
+ {0x00dc, 0x00c0},
+ {0x00dd, 0x2220},
+ {0x00de, 0x3131},
+ {0x00df, 0x3131},
+ {0x00e0, 0x3131},
+ {0x00e2, 0x0000},
+ {0x00e3, 0x4000},
+ {0x00e4, 0x0aa0},
+ {0x00e5, 0x3131},
+ {0x00e6, 0x3131},
+ {0x00e7, 0x3131},
+ {0x00e8, 0x3131},
+ {0x00ea, 0xb320},
+ {0x00eb, 0x0000},
+ {0x00f0, 0x0000},
+ {0x00f1, 0x00d0},
+ {0x00f2, 0x00d0},
+ {0x00f6, 0x0000},
+ {0x00fa, 0x0000},
+ {0x00fb, 0x0000},
+ {0x00fc, 0x0000},
+ {0x00fd, 0x0000},
+ {0x00fe, 0x10ec},
+ {0x00ff, 0x6530},
+ {0x0100, 0xa0a0},
+ {0x010b, 0x0000},
+ {0x010c, 0xae00},
+ {0x010d, 0xaaa0},
+ {0x010e, 0x8aa2},
+ {0x010f, 0x02a2},
+ {0x0110, 0xc000},
+ {0x0111, 0x04a2},
+ {0x0112, 0x2800},
+ {0x0113, 0x0000},
+ {0x0117, 0x0100},
+ {0x0125, 0x0410},
+ {0x0132, 0x6026},
+ {0x0136, 0x5555},
+ {0x0138, 0x3700},
+ {0x013a, 0x2000},
+ {0x013b, 0x2000},
+ {0x013c, 0x2005},
+ {0x013f, 0x0000},
+ {0x0142, 0x0000},
+ {0x0145, 0x0002},
+ {0x0146, 0x0000},
+ {0x0147, 0x0000},
+ {0x0148, 0x0000},
+ {0x0149, 0x0000},
+ {0x0150, 0x79a1},
+ {0x0151, 0x0000},
+ {0x0160, 0x4ec0},
+ {0x0161, 0x0080},
+ {0x0162, 0x0200},
+ {0x0163, 0x0800},
+ {0x0164, 0x0000},
+ {0x0165, 0x0000},
+ {0x0166, 0x0000},
+ {0x0167, 0x000f},
+ {0x0168, 0x000f},
+ {0x0169, 0x0021},
+ {0x0190, 0x413d},
+ {0x0194, 0x0000},
+ {0x0195, 0x0000},
+ {0x0197, 0x0022},
+ {0x0198, 0x0000},
+ {0x0199, 0x0000},
+ {0x01af, 0x0000},
+ {0x01b0, 0x0400},
+ {0x01b1, 0x0000},
+ {0x01b2, 0x0000},
+ {0x01b3, 0x0000},
+ {0x01b4, 0x0000},
+ {0x01b5, 0x0000},
+ {0x01b6, 0x01c3},
+ {0x01b7, 0x02a0},
+ {0x01b8, 0x03e9},
+ {0x01b9, 0x1389},
+ {0x01ba, 0xc351},
+ {0x01bb, 0x0009},
+ {0x01bc, 0x0018},
+ {0x01bd, 0x002a},
+ {0x01be, 0x004c},
+ {0x01bf, 0x0097},
+ {0x01c0, 0x433d},
+ {0x01c2, 0x0000},
+ {0x01c3, 0x0000},
+ {0x01c4, 0x0000},
+ {0x01c5, 0x0000},
+ {0x01c6, 0x0000},
+ {0x01c7, 0x0000},
+ {0x01c8, 0x40af},
+ {0x01c9, 0x0702},
+ {0x01ca, 0x0000},
+ {0x01cb, 0x0000},
+ {0x01cc, 0x5757},
+ {0x01cd, 0x5757},
+ {0x01ce, 0x5757},
+ {0x01cf, 0x5757},
+ {0x01d0, 0x5757},
+ {0x01d1, 0x5757},
+ {0x01d2, 0x5757},
+ {0x01d3, 0x5757},
+ {0x01d4, 0x5757},
+ {0x01d5, 0x5757},
+ {0x01d6, 0x0000},
+ {0x01d7, 0x0008},
+ {0x01d8, 0x0029},
+ {0x01d9, 0x3333},
+ {0x01da, 0x0000},
+ {0x01db, 0x0004},
+ {0x01dc, 0x0000},
+ {0x01de, 0x7c00},
+ {0x01df, 0x0320},
+ {0x01e0, 0x06a1},
+ {0x01e1, 0x0000},
+ {0x01e2, 0x0000},
+ {0x01e3, 0x0000},
+ {0x01e4, 0x0000},
+ {0x01e6, 0x0001},
+ {0x01e7, 0x0000},
+ {0x01e8, 0x0000},
+ {0x01ea, 0x0000},
+ {0x01eb, 0x0000},
+ {0x01ec, 0x0000},
+ {0x01ed, 0x0000},
+ {0x01ee, 0x0000},
+ {0x01ef, 0x0000},
+ {0x01f0, 0x0000},
+ {0x01f1, 0x0000},
+ {0x01f2, 0x0000},
+ {0x01f3, 0x0000},
+ {0x01f4, 0x0000},
+ {0x0210, 0x6297},
+ {0x0211, 0xa005},
+ {0x0212, 0x824c},
+ {0x0213, 0xf7ff},
+ {0x0214, 0xf24c},
+ {0x0215, 0x0102},
+ {0x0216, 0x00a3},
+ {0x0217, 0x0048},
+ {0x0218, 0xa2c0},
+ {0x0219, 0x0400},
+ {0x021a, 0x00c8},
+ {0x021b, 0x00c0},
+ {0x021c, 0x0000},
+ {0x0250, 0x4500},
+ {0x0251, 0x40b3},
+ {0x0252, 0x0000},
+ {0x0253, 0x0000},
+ {0x0254, 0x0000},
+ {0x0255, 0x0000},
+ {0x0256, 0x0000},
+ {0x0257, 0x0000},
+ {0x0258, 0x0000},
+ {0x0259, 0x0000},
+ {0x025a, 0x0005},
+ {0x0270, 0x0000},
+ {0x02ff, 0x0110},
+ {0x0300, 0x001f},
+ {0x0301, 0x032c},
+ {0x0302, 0x5f21},
+ {0x0303, 0x4000},
+ {0x0304, 0x4000},
+ {0x0305, 0x06d5},
+ {0x0306, 0x8000},
+ {0x0307, 0x0700},
+ {0x0310, 0x4560},
+ {0x0311, 0xa4a8},
+ {0x0312, 0x7418},
+ {0x0313, 0x0000},
+ {0x0314, 0x0006},
+ {0x0315, 0xffff},
+ {0x0316, 0xc400},
+ {0x0317, 0x0000},
+ {0x03c0, 0x7e00},
+ {0x03c1, 0x8000},
+ {0x03c2, 0x8000},
+ {0x03c3, 0x8000},
+ {0x03c4, 0x8000},
+ {0x03c5, 0x8000},
+ {0x03c6, 0x8000},
+ {0x03c7, 0x8000},
+ {0x03c8, 0x8000},
+ {0x03c9, 0x8000},
+ {0x03ca, 0x8000},
+ {0x03cb, 0x8000},
+ {0x03cc, 0x8000},
+ {0x03d0, 0x0000},
+ {0x03d1, 0x0000},
+ {0x03d2, 0x0000},
+ {0x03d3, 0x0000},
+ {0x03d4, 0x2000},
+ {0x03d5, 0x2000},
+ {0x03d6, 0x0000},
+ {0x03d7, 0x0000},
+ {0x03d8, 0x2000},
+ {0x03d9, 0x2000},
+ {0x03da, 0x2000},
+ {0x03db, 0x2000},
+ {0x03dc, 0x0000},
+ {0x03dd, 0x0000},
+ {0x03de, 0x0000},
+ {0x03df, 0x2000},
+ {0x03e0, 0x0000},
+ {0x03e1, 0x0000},
+ {0x03e2, 0x0000},
+ {0x03e3, 0x0000},
+ {0x03e4, 0x0000},
+ {0x03e5, 0x0000},
+ {0x03e6, 0x0000},
+ {0x03e7, 0x0000},
+ {0x03e8, 0x0000},
+ {0x03e9, 0x0000},
+ {0x03ea, 0x0000},
+ {0x03eb, 0x0000},
+ {0x03ec, 0x0000},
+ {0x03ed, 0x0000},
+ {0x03ee, 0x0000},
+ {0x03ef, 0x0000},
+ {0x03f0, 0x0800},
+ {0x03f1, 0x0800},
+ {0x03f2, 0x0800},
+ {0x03f3, 0x0800},
+};
+
+static bool rt5682_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case RT5682_RESET:
+ case RT5682_CBJ_CTRL_2:
+ case RT5682_INT_ST_1:
+ case RT5682_4BTN_IL_CMD_1:
+ case RT5682_AJD1_CTRL:
+ case RT5682_HP_CALIB_CTRL_1:
+ case RT5682_DEVICE_ID:
+ case RT5682_I2C_MODE:
+ case RT5682_HP_CALIB_CTRL_10:
+ case RT5682_EFUSE_CTRL_2:
+ case RT5682_JD_TOP_VC_VTRL:
+ case RT5682_HP_IMP_SENS_CTRL_19:
+ case RT5682_IL_CMD_1:
+ case RT5682_SAR_IL_CMD_2:
+ case RT5682_SAR_IL_CMD_4:
+ case RT5682_SAR_IL_CMD_10:
+ case RT5682_SAR_IL_CMD_11:
+ case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11:
+ case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rt5682_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case RT5682_RESET:
+ case RT5682_VERSION_ID:
+ case RT5682_VENDOR_ID:
+ case RT5682_DEVICE_ID:
+ case RT5682_HP_CTRL_1:
+ case RT5682_HP_CTRL_2:
+ case RT5682_HPL_GAIN:
+ case RT5682_HPR_GAIN:
+ case RT5682_I2C_CTRL:
+ case RT5682_CBJ_BST_CTRL:
+ case RT5682_CBJ_CTRL_1:
+ case RT5682_CBJ_CTRL_2:
+ case RT5682_CBJ_CTRL_3:
+ case RT5682_CBJ_CTRL_4:
+ case RT5682_CBJ_CTRL_5:
+ case RT5682_CBJ_CTRL_6:
+ case RT5682_CBJ_CTRL_7:
+ case RT5682_DAC1_DIG_VOL:
+ case RT5682_STO1_ADC_DIG_VOL:
+ case RT5682_STO1_ADC_BOOST:
+ case RT5682_HP_IMP_GAIN_1:
+ case RT5682_HP_IMP_GAIN_2:
+ case RT5682_SIDETONE_CTRL:
+ case RT5682_STO1_ADC_MIXER:
+ case RT5682_AD_DA_MIXER:
+ case RT5682_STO1_DAC_MIXER:
+ case RT5682_A_DAC1_MUX:
+ case RT5682_DIG_INF2_DATA:
+ case RT5682_REC_MIXER:
+ case RT5682_CAL_REC:
+ case RT5682_ALC_BACK_GAIN:
+ case RT5682_PWR_DIG_1:
+ case RT5682_PWR_DIG_2:
+ case RT5682_PWR_ANLG_1:
+ case RT5682_PWR_ANLG_2:
+ case RT5682_PWR_ANLG_3:
+ case RT5682_PWR_MIXER:
+ case RT5682_PWR_VOL:
+ case RT5682_CLK_DET:
+ case RT5682_RESET_LPF_CTRL:
+ case RT5682_RESET_HPF_CTRL:
+ case RT5682_DMIC_CTRL_1:
+ case RT5682_I2S1_SDP:
+ case RT5682_I2S2_SDP:
+ case RT5682_ADDA_CLK_1:
+ case RT5682_ADDA_CLK_2:
+ case RT5682_I2S1_F_DIV_CTRL_1:
+ case RT5682_I2S1_F_DIV_CTRL_2:
+ case RT5682_TDM_CTRL:
+ case RT5682_TDM_ADDA_CTRL_1:
+ case RT5682_TDM_ADDA_CTRL_2:
+ case RT5682_DATA_SEL_CTRL_1:
+ case RT5682_TDM_TCON_CTRL:
+ case RT5682_GLB_CLK:
+ case RT5682_PLL_CTRL_1:
+ case RT5682_PLL_CTRL_2:
+ case RT5682_PLL_TRACK_1:
+ case RT5682_PLL_TRACK_2:
+ case RT5682_PLL_TRACK_3:
+ case RT5682_PLL_TRACK_4:
+ case RT5682_PLL_TRACK_5:
+ case RT5682_PLL_TRACK_6:
+ case RT5682_PLL_TRACK_11:
+ case RT5682_SDW_REF_CLK:
+ case RT5682_DEPOP_1:
+ case RT5682_DEPOP_2:
+ case RT5682_HP_CHARGE_PUMP_1:
+ case RT5682_HP_CHARGE_PUMP_2:
+ case RT5682_MICBIAS_1:
+ case RT5682_MICBIAS_2:
+ case RT5682_PLL_TRACK_12:
+ case RT5682_PLL_TRACK_14:
+ case RT5682_PLL2_CTRL_1:
+ case RT5682_PLL2_CTRL_2:
+ case RT5682_PLL2_CTRL_3:
+ case RT5682_PLL2_CTRL_4:
+ case RT5682_RC_CLK_CTRL:
+ case RT5682_I2S_M_CLK_CTRL_1:
+ case RT5682_I2S2_F_DIV_CTRL_1:
+ case RT5682_I2S2_F_DIV_CTRL_2:
+ case RT5682_EQ_CTRL_1:
+ case RT5682_EQ_CTRL_2:
+ case RT5682_IRQ_CTRL_1:
+ case RT5682_IRQ_CTRL_2:
+ case RT5682_IRQ_CTRL_3:
+ case RT5682_IRQ_CTRL_4:
+ case RT5682_INT_ST_1:
+ case RT5682_GPIO_CTRL_1:
+ case RT5682_GPIO_CTRL_2:
+ case RT5682_GPIO_CTRL_3:
+ case RT5682_HP_AMP_DET_CTRL_1:
+ case RT5682_HP_AMP_DET_CTRL_2:
+ case RT5682_MID_HP_AMP_DET:
+ case RT5682_LOW_HP_AMP_DET:
+ case RT5682_DELAY_BUF_CTRL:
+ case RT5682_SV_ZCD_1:
+ case RT5682_SV_ZCD_2:
+ case RT5682_IL_CMD_1:
+ case RT5682_IL_CMD_2:
+ case RT5682_IL_CMD_3:
+ case RT5682_IL_CMD_4:
+ case RT5682_IL_CMD_5:
+ case RT5682_IL_CMD_6:
+ case RT5682_4BTN_IL_CMD_1:
+ case RT5682_4BTN_IL_CMD_2:
+ case RT5682_4BTN_IL_CMD_3:
+ case RT5682_4BTN_IL_CMD_4:
+ case RT5682_4BTN_IL_CMD_5:
+ case RT5682_4BTN_IL_CMD_6:
+ case RT5682_4BTN_IL_CMD_7:
+ case RT5682_ADC_STO1_HP_CTRL_1:
+ case RT5682_ADC_STO1_HP_CTRL_2:
+ case RT5682_AJD1_CTRL:
+ case RT5682_JD1_THD:
+ case RT5682_JD2_THD:
+ case RT5682_JD_CTRL_1:
+ case RT5682_DUMMY_1:
+ case RT5682_DUMMY_2:
+ case RT5682_DUMMY_3:
+ case RT5682_DAC_ADC_DIG_VOL1:
+ case RT5682_BIAS_CUR_CTRL_2:
+ case RT5682_BIAS_CUR_CTRL_3:
+ case RT5682_BIAS_CUR_CTRL_4:
+ case RT5682_BIAS_CUR_CTRL_5:
+ case RT5682_BIAS_CUR_CTRL_6:
+ case RT5682_BIAS_CUR_CTRL_7:
+ case RT5682_BIAS_CUR_CTRL_8:
+ case RT5682_BIAS_CUR_CTRL_9:
+ case RT5682_BIAS_CUR_CTRL_10:
+ case RT5682_VREF_REC_OP_FB_CAP_CTRL:
+ case RT5682_CHARGE_PUMP_1:
+ case RT5682_DIG_IN_CTRL_1:
+ case RT5682_PAD_DRIVING_CTRL:
+ case RT5682_SOFT_RAMP_DEPOP:
+ case RT5682_CHOP_DAC:
+ case RT5682_CHOP_ADC:
+ case RT5682_CALIB_ADC_CTRL:
+ case RT5682_VOL_TEST:
+ case RT5682_SPKVDD_DET_STA:
+ case RT5682_TEST_MODE_CTRL_1:
+ case RT5682_TEST_MODE_CTRL_2:
+ case RT5682_TEST_MODE_CTRL_3:
+ case RT5682_TEST_MODE_CTRL_4:
+ case RT5682_TEST_MODE_CTRL_5:
+ case RT5682_PLL1_INTERNAL:
+ case RT5682_PLL2_INTERNAL:
+ case RT5682_STO_NG2_CTRL_1:
+ case RT5682_STO_NG2_CTRL_2:
+ case RT5682_STO_NG2_CTRL_3:
+ case RT5682_STO_NG2_CTRL_4:
+ case RT5682_STO_NG2_CTRL_5:
+ case RT5682_STO_NG2_CTRL_6:
+ case RT5682_STO_NG2_CTRL_7:
+ case RT5682_STO_NG2_CTRL_8:
+ case RT5682_STO_NG2_CTRL_9:
+ case RT5682_STO_NG2_CTRL_10:
+ case RT5682_STO1_DAC_SIL_DET:
+ case RT5682_SIL_PSV_CTRL1:
+ case RT5682_SIL_PSV_CTRL2:
+ case RT5682_SIL_PSV_CTRL3:
+ case RT5682_SIL_PSV_CTRL4:
+ case RT5682_SIL_PSV_CTRL5:
+ case RT5682_HP_IMP_SENS_CTRL_01:
+ case RT5682_HP_IMP_SENS_CTRL_02:
+ case RT5682_HP_IMP_SENS_CTRL_03:
+ case RT5682_HP_IMP_SENS_CTRL_04:
+ case RT5682_HP_IMP_SENS_CTRL_05:
+ case RT5682_HP_IMP_SENS_CTRL_06:
+ case RT5682_HP_IMP_SENS_CTRL_07:
+ case RT5682_HP_IMP_SENS_CTRL_08:
+ case RT5682_HP_IMP_SENS_CTRL_09:
+ case RT5682_HP_IMP_SENS_CTRL_10:
+ case RT5682_HP_IMP_SENS_CTRL_11:
+ case RT5682_HP_IMP_SENS_CTRL_12:
+ case RT5682_HP_IMP_SENS_CTRL_13:
+ case RT5682_HP_IMP_SENS_CTRL_14:
+ case RT5682_HP_IMP_SENS_CTRL_15:
+ case RT5682_HP_IMP_SENS_CTRL_16:
+ case RT5682_HP_IMP_SENS_CTRL_17:
+ case RT5682_HP_IMP_SENS_CTRL_18:
+ case RT5682_HP_IMP_SENS_CTRL_19:
+ case RT5682_HP_IMP_SENS_CTRL_20:
+ case RT5682_HP_IMP_SENS_CTRL_21:
+ case RT5682_HP_IMP_SENS_CTRL_22:
+ case RT5682_HP_IMP_SENS_CTRL_23:
+ case RT5682_HP_IMP_SENS_CTRL_24:
+ case RT5682_HP_IMP_SENS_CTRL_25:
+ case RT5682_HP_IMP_SENS_CTRL_26:
+ case RT5682_HP_IMP_SENS_CTRL_27:
+ case RT5682_HP_IMP_SENS_CTRL_28:
+ case RT5682_HP_IMP_SENS_CTRL_29:
+ case RT5682_HP_IMP_SENS_CTRL_30:
+ case RT5682_HP_IMP_SENS_CTRL_31:
+ case RT5682_HP_IMP_SENS_CTRL_32:
+ case RT5682_HP_IMP_SENS_CTRL_33:
+ case RT5682_HP_IMP_SENS_CTRL_34:
+ case RT5682_HP_IMP_SENS_CTRL_35:
+ case RT5682_HP_IMP_SENS_CTRL_36:
+ case RT5682_HP_IMP_SENS_CTRL_37:
+ case RT5682_HP_IMP_SENS_CTRL_38:
+ case RT5682_HP_IMP_SENS_CTRL_39:
+ case RT5682_HP_IMP_SENS_CTRL_40:
+ case RT5682_HP_IMP_SENS_CTRL_41:
+ case RT5682_HP_IMP_SENS_CTRL_42:
+ case RT5682_HP_IMP_SENS_CTRL_43:
+ case RT5682_HP_LOGIC_CTRL_1:
+ case RT5682_HP_LOGIC_CTRL_2:
+ case RT5682_HP_LOGIC_CTRL_3:
+ case RT5682_HP_CALIB_CTRL_1:
+ case RT5682_HP_CALIB_CTRL_2:
+ case RT5682_HP_CALIB_CTRL_3:
+ case RT5682_HP_CALIB_CTRL_4:
+ case RT5682_HP_CALIB_CTRL_5:
+ case RT5682_HP_CALIB_CTRL_6:
+ case RT5682_HP_CALIB_CTRL_7:
+ case RT5682_HP_CALIB_CTRL_9:
+ case RT5682_HP_CALIB_CTRL_10:
+ case RT5682_HP_CALIB_CTRL_11:
+ case RT5682_HP_CALIB_STA_1:
+ case RT5682_HP_CALIB_STA_2:
+ case RT5682_HP_CALIB_STA_3:
+ case RT5682_HP_CALIB_STA_4:
+ case RT5682_HP_CALIB_STA_5:
+ case RT5682_HP_CALIB_STA_6:
+ case RT5682_HP_CALIB_STA_7:
+ case RT5682_HP_CALIB_STA_8:
+ case RT5682_HP_CALIB_STA_9:
+ case RT5682_HP_CALIB_STA_10:
+ case RT5682_HP_CALIB_STA_11:
+ case RT5682_SAR_IL_CMD_1:
+ case RT5682_SAR_IL_CMD_2:
+ case RT5682_SAR_IL_CMD_3:
+ case RT5682_SAR_IL_CMD_4:
+ case RT5682_SAR_IL_CMD_5:
+ case RT5682_SAR_IL_CMD_6:
+ case RT5682_SAR_IL_CMD_7:
+ case RT5682_SAR_IL_CMD_8:
+ case RT5682_SAR_IL_CMD_9:
+ case RT5682_SAR_IL_CMD_10:
+ case RT5682_SAR_IL_CMD_11:
+ case RT5682_SAR_IL_CMD_12:
+ case RT5682_SAR_IL_CMD_13:
+ case RT5682_EFUSE_CTRL_1:
+ case RT5682_EFUSE_CTRL_2:
+ case RT5682_EFUSE_CTRL_3:
+ case RT5682_EFUSE_CTRL_4:
+ case RT5682_EFUSE_CTRL_5:
+ case RT5682_EFUSE_CTRL_6:
+ case RT5682_EFUSE_CTRL_7:
+ case RT5682_EFUSE_CTRL_8:
+ case RT5682_EFUSE_CTRL_9:
+ case RT5682_EFUSE_CTRL_10:
+ case RT5682_EFUSE_CTRL_11:
+ case RT5682_JD_TOP_VC_VTRL:
+ case RT5682_DRC1_CTRL_0:
+ case RT5682_DRC1_CTRL_1:
+ case RT5682_DRC1_CTRL_2:
+ case RT5682_DRC1_CTRL_3:
+ case RT5682_DRC1_CTRL_4:
+ case RT5682_DRC1_CTRL_5:
+ case RT5682_DRC1_CTRL_6:
+ case RT5682_DRC1_HARD_LMT_CTRL_1:
+ case RT5682_DRC1_HARD_LMT_CTRL_2:
+ case RT5682_DRC1_PRIV_1:
+ case RT5682_DRC1_PRIV_2:
+ case RT5682_DRC1_PRIV_3:
+ case RT5682_DRC1_PRIV_4:
+ case RT5682_DRC1_PRIV_5:
+ case RT5682_DRC1_PRIV_6:
+ case RT5682_DRC1_PRIV_7:
+ case RT5682_DRC1_PRIV_8:
+ case RT5682_EQ_AUTO_RCV_CTRL1:
+ case RT5682_EQ_AUTO_RCV_CTRL2:
+ case RT5682_EQ_AUTO_RCV_CTRL3:
+ case RT5682_EQ_AUTO_RCV_CTRL4:
+ case RT5682_EQ_AUTO_RCV_CTRL5:
+ case RT5682_EQ_AUTO_RCV_CTRL6:
+ case RT5682_EQ_AUTO_RCV_CTRL7:
+ case RT5682_EQ_AUTO_RCV_CTRL8:
+ case RT5682_EQ_AUTO_RCV_CTRL9:
+ case RT5682_EQ_AUTO_RCV_CTRL10:
+ case RT5682_EQ_AUTO_RCV_CTRL11:
+ case RT5682_EQ_AUTO_RCV_CTRL12:
+ case RT5682_EQ_AUTO_RCV_CTRL13:
+ case RT5682_ADC_L_EQ_LPF1_A1:
+ case RT5682_R_EQ_LPF1_A1:
+ case RT5682_L_EQ_LPF1_H0:
+ case RT5682_R_EQ_LPF1_H0:
+ case RT5682_L_EQ_BPF1_A1:
+ case RT5682_R_EQ_BPF1_A1:
+ case RT5682_L_EQ_BPF1_A2:
+ case RT5682_R_EQ_BPF1_A2:
+ case RT5682_L_EQ_BPF1_H0:
+ case RT5682_R_EQ_BPF1_H0:
+ case RT5682_L_EQ_BPF2_A1:
+ case RT5682_R_EQ_BPF2_A1:
+ case RT5682_L_EQ_BPF2_A2:
+ case RT5682_R_EQ_BPF2_A2:
+ case RT5682_L_EQ_BPF2_H0:
+ case RT5682_R_EQ_BPF2_H0:
+ case RT5682_L_EQ_BPF3_A1:
+ case RT5682_R_EQ_BPF3_A1:
+ case RT5682_L_EQ_BPF3_A2:
+ case RT5682_R_EQ_BPF3_A2:
+ case RT5682_L_EQ_BPF3_H0:
+ case RT5682_R_EQ_BPF3_H0:
+ case RT5682_L_EQ_BPF4_A1:
+ case RT5682_R_EQ_BPF4_A1:
+ case RT5682_L_EQ_BPF4_A2:
+ case RT5682_R_EQ_BPF4_A2:
+ case RT5682_L_EQ_BPF4_H0:
+ case RT5682_R_EQ_BPF4_H0:
+ case RT5682_L_EQ_HPF1_A1:
+ case RT5682_R_EQ_HPF1_A1:
+ case RT5682_L_EQ_HPF1_H0:
+ case RT5682_R_EQ_HPF1_H0:
+ case RT5682_L_EQ_PRE_VOL:
+ case RT5682_R_EQ_PRE_VOL:
+ case RT5682_L_EQ_POST_VOL:
+ case RT5682_R_EQ_POST_VOL:
+ case RT5682_I2C_MODE:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
+
+/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
+static const DECLARE_TLV_DB_RANGE(bst_tlv,
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0),
+ 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0),
+ 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0),
+ 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0)
+);
+
+/* Interface data select */
+static const char * const rt5682_data_select[] = {
+ "L/R", "R/L", "L/L", "R/R"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum,
+ RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum,
+ RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum,
+ RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum,
+ RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum,
+ RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select);
+
+static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux =
+ SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux =
+ SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux =
+ SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux =
+ SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux =
+ SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum);
+
+static void rt5682_reset(struct regmap *regmap)
+{
+ regmap_write(regmap, RT5682_RESET, 0);
+ regmap_write(regmap, RT5682_I2C_MODE, 1);
+}
+/**
+ * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @component: SoC audio component device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the component driver will turn on
+ * ASRC for these filters if ASRC is selected as their clock source.
+ */
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+ unsigned int filter_mask, unsigned int clk_src)
+{
+
+ switch (clk_src) {
+ case RT5682_CLK_SEL_SYS:
+ case RT5682_CLK_SEL_I2S1_ASRC:
+ case RT5682_CLK_SEL_I2S2_ASRC:
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ if (filter_mask & RT5682_DA_STEREO1_FILTER) {
+ snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2,
+ RT5682_FILTER_CLK_SEL_MASK,
+ clk_src << RT5682_FILTER_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5682_AD_STEREO1_FILTER) {
+ snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3,
+ RT5682_FILTER_CLK_SEL_MASK,
+ clk_src << RT5682_FILTER_CLK_SEL_SFT);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src);
+
+static int rt5682_button_detect(struct snd_soc_component *component)
+{
+ int btn_type, val;
+
+ val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1);
+ btn_type = val & 0xfff0;
+ snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val);
+ pr_debug("%s btn_type=%x\n", __func__, btn_type);
+ snd_soc_component_update_bits(component,
+ RT5682_SAR_IL_CMD_2, 0x10, 0x10);
+
+ return btn_type;
+}
+
+static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
+ bool enable)
+{
+ if (enable) {
+ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+ RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN);
+ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+ RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN);
+ snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040);
+ snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+ RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK,
+ RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR);
+ snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+ RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN);
+ } else {
+ snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+ RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS);
+ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+ RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS);
+ snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+ RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS);
+ snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+ RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST);
+ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+ RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE);
+ }
+}
+
+/**
+ * rt5682_headset_detect - Detect headset.
+ * @component: SoC audio component device.
+ * @jack_insert: Jack insert or not.
+ *
+ * Detect whether is headset or not when jack inserted.
+ *
+ * Returns detect status.
+ */
+static int rt5682_headset_detect(struct snd_soc_component *component,
+ int jack_insert)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
+ unsigned int val, count;
+
+ if (jack_insert) {
+ snd_soc_dapm_force_enable_pin(dapm, "CBJ Power");
+ snd_soc_dapm_sync(dapm);
+ snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
+ RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
+
+ count = 0;
+ val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2)
+ & RT5682_JACK_TYPE_MASK;
+ while (val == 0 && count < 50) {
+ usleep_range(10000, 15000);
+ val = snd_soc_component_read32(component,
+ RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK;
+ count++;
+ }
+
+ switch (val) {
+ case 0x1:
+ case 0x2:
+ rt5682->jack_type = SND_JACK_HEADSET;
+ rt5682_enable_push_button_irq(component, true);
+ break;
+ default:
+ rt5682->jack_type = SND_JACK_HEADPHONE;
+ }
+
+ } else {
+ rt5682_enable_push_button_irq(component, false);
+ snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
+ RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
+ snd_soc_dapm_disable_pin(dapm, "CBJ Power");
+ snd_soc_dapm_sync(dapm);
+
+ rt5682->jack_type = 0;
+ }
+
+ dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type);
+ return rt5682->jack_type;
+}
+
+static irqreturn_t rt5682_irq(int irq, void *data)
+{
+ struct rt5682_priv *rt5682 = data;
+
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work, msecs_to_jiffies(250));
+
+ return IRQ_HANDLED;
+}
+
+static void rt5682_jd_check_handler(struct work_struct *work)
+{
+ struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv,
+ jd_check_work.work);
+
+ if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+ & RT5682_JDH_RS_MASK) {
+ /* jack out */
+ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+
+ snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3);
+ } else {
+ schedule_delayed_work(&rt5682->jd_check_work, 500);
+ }
+}
+
+static int rt5682_set_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *hs_jack, void *data)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ switch (rt5682->pdata.jd_src) {
+ case RT5682_JD1:
+ snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
+ RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+ snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042);
+ snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3,
+ RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
+ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+ RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_IRQ | RT5682_POW_JDH |
+ RT5682_POW_ANA, RT5682_POW_IRQ |
+ RT5682_POW_JDH | RT5682_POW_ANA);
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+ RT5682_PWR_JDH | RT5682_PWR_JDL,
+ RT5682_PWR_JDH | RT5682_PWR_JDL);
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
+ RT5682_JD1_EN | RT5682_JD1_POL_NOR);
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work, msecs_to_jiffies(250));
+ break;
+
+ case RT5682_JD_NULL:
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_JDH | RT5682_POW_JDL, 0);
+ break;
+
+ default:
+ dev_warn(component->dev, "Wrong JD source\n");
+ break;
+ }
+
+ rt5682->hs_jack = hs_jack;
+
+ return 0;
+}
+
+static void rt5682_jack_detect_handler(struct work_struct *work)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(work, struct rt5682_priv, jack_detect_work.work);
+ int val, btn_type;
+
+ while (!rt5682->component)
+ usleep_range(10000, 15000);
+
+ while (!rt5682->component->card->instantiated)
+ usleep_range(10000, 15000);
+
+ mutex_lock(&rt5682->calibrate_mutex);
+
+ val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+ & RT5682_JDH_RS_MASK;
+ if (!val) {
+ /* jack in */
+ if (rt5682->jack_type == 0) {
+ /* jack was out, report jack type */
+ rt5682->jack_type =
+ rt5682_headset_detect(rt5682->component, 1);
+ } else {
+ /* jack is already in, report button event */
+ rt5682->jack_type = SND_JACK_HEADSET;
+ btn_type = rt5682_button_detect(rt5682->component);
+ /**
+ * rt5682 can report three kinds of button behavior,
+ * one click, double click and hold. However,
+ * currently we will report button pressed/released
+ * event. So all the three button behaviors are
+ * treated as button pressed.
+ */
+ switch (btn_type) {
+ case 0x8000:
+ case 0x4000:
+ case 0x2000:
+ rt5682->jack_type |= SND_JACK_BTN_0;
+ break;
+ case 0x1000:
+ case 0x0800:
+ case 0x0400:
+ rt5682->jack_type |= SND_JACK_BTN_1;
+ break;
+ case 0x0200:
+ case 0x0100:
+ case 0x0080:
+ rt5682->jack_type |= SND_JACK_BTN_2;
+ break;
+ case 0x0040:
+ case 0x0020:
+ case 0x0010:
+ rt5682->jack_type |= SND_JACK_BTN_3;
+ break;
+ case 0x0000: /* unpressed */
+ break;
+ default:
+ btn_type = 0;
+ dev_err(rt5682->component->dev,
+ "Unexpected button code 0x%04x\n",
+ btn_type);
+ break;
+ }
+ }
+ } else {
+ /* jack out */
+ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+ }
+
+ snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3);
+
+ if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3))
+ schedule_delayed_work(&rt5682->jd_check_work, 0);
+ else
+ cancel_delayed_work_sync(&rt5682->jd_check_work);
+
+ mutex_unlock(&rt5682->calibrate_mutex);
+}
+
+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, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv),
+
+ /* IN Boost Volume */
+ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL,
+ RT5682_BST_CBJ_SFT, 8, 0, bst_tlv),
+
+ /* ADC Digital Volume Control */
+ SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL,
+ RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1),
+ SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL,
+ RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv),
+
+ /* ADC Boost Volume Control */
+ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST,
+ RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT,
+ 3, 0, adc_bst_tlv),
+};
+
+
+static int rt5682_div_sel(struct rt5682_priv *rt5682,
+ int target, const int div[], int size)
+{
+ int i;
+
+ if (rt5682->sysclk < target) {
+ pr_err("sysclk rate %d is too low\n",
+ rt5682->sysclk);
+ return 0;
+ }
+
+ for (i = 0; i < size - 1; i++) {
+ pr_info("div[%d]=%d\n", i, div[i]);
+ if (target * div[i] == rt5682->sysclk)
+ return i;
+ if (target * div[i + 1] > rt5682->sysclk) {
+ pr_err("can't find div for sysclk %d\n",
+ rt5682->sysclk);
+ return i;
+ }
+ }
+
+ if (target * div[i] < rt5682->sysclk)
+ pr_err("sysclk rate %d is too high\n",
+ rt5682->sysclk);
+
+ return size - 1;
+
+}
+
+/**
+ * set_dmic_clk - Set parameter of dmic.
+ *
+ * @w: DAPM widget.
+ * @kcontrol: The kcontrol of this widget.
+ * @event: Event id.
+ *
+ * Choose dmic clock between 1MHz and 3MHz.
+ * It is better for clock to approximate 3MHz.
+ */
+static int set_dmic_clk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ int idx = -EINVAL;
+ static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128};
+
+ idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div));
+
+ snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1,
+ RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT);
+
+ return 0;
+}
+
+static int set_filter_clk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ int ref, val, reg, sft, mask, idx = -EINVAL;
+ static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48};
+ static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48};
+
+ val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) &
+ RT5682_GP4_PIN_MASK;
+ if (w->shift == RT5682_PWR_ADC_S1F_BIT &&
+ val == RT5682_GP4_PIN_ADCDAT2)
+ ref = 256 * rt5682->lrck[RT5682_AIF2];
+ else
+ ref = 256 * rt5682->lrck[RT5682_AIF1];
+
+ idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f));
+
+ if (w->shift == RT5682_PWR_ADC_S1F_BIT) {
+ reg = RT5682_PLL_TRACK_3;
+ sft = RT5682_ADC_OSR_SFT;
+ mask = RT5682_ADC_OSR_MASK;
+ } else {
+ reg = RT5682_PLL_TRACK_2;
+ sft = RT5682_DAC_OSR_SFT;
+ mask = RT5682_DAC_OSR_MASK;
+ }
+
+ snd_soc_component_update_bits(component, reg,
+ RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT);
+
+ /* select over sample rate */
+ for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) {
+ if (rt5682->sysclk <= 12288000 * div_o[idx])
+ break;
+ }
+
+ snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1,
+ mask, idx << sft);
+
+ return 0;
+}
+
+static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int val;
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ val = snd_soc_component_read32(component, RT5682_GLB_CLK);
+ val &= RT5682_SCLK_SRC_MASK;
+ if (val == RT5682_SCLK_SRC_PLL1)
+ return 1;
+ else
+ return 0;
+}
+
+static int is_using_asrc(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int reg, shift, val;
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ switch (w->shift) {
+ case RT5682_ADC_STO1_ASRC_SFT:
+ reg = RT5682_PLL_TRACK_3;
+ shift = RT5682_FILTER_CLK_SEL_SFT;
+ break;
+ case RT5682_DAC_STO1_ASRC_SFT:
+ reg = RT5682_PLL_TRACK_2;
+ shift = RT5682_FILTER_CLK_SEL_SFT;
+ break;
+ default:
+ return 0;
+ }
+
+ val = (snd_soc_component_read32(component, reg) >> shift) & 0xf;
+ switch (val) {
+ case RT5682_CLK_SEL_I2S1_ASRC:
+ case RT5682_CLK_SEL_I2S2_ASRC:
+ return 1;
+ default:
+ return 0;
+ }
+
+}
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+ RT5682_M_STO1_ADC_L1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+ RT5682_M_STO1_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+ RT5682_M_STO1_ADC_R1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+ RT5682_M_STO1_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_l_mix[] = {
+ SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+ RT5682_M_ADCMIX_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+ RT5682_M_DAC1_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_r_mix[] = {
+ SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+ RT5682_M_ADCMIX_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+ RT5682_M_DAC1_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+ RT5682_M_DAC_L1_STO_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+ RT5682_M_DAC_R1_STO_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+ RT5682_M_DAC_L1_STO_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+ RT5682_M_DAC_R1_STO_R_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = {
+ SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER,
+ RT5682_M_CBJ_RM1_L_SFT, 1, 1),
+};
+
+/* STO1 ADC1 Source */
+/* MX-26 [13] [5] */
+static const char * const rt5682_sto1_adc1_src[] = {
+ "DAC MIX", "ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum);
+
+/* STO1 ADC Source */
+/* MX-26 [11:10] [3:2] */
+static const char * const rt5682_sto1_adc_src[] = {
+ "ADC1 L", "ADC1 R"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcl_mux =
+ SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcr_mux =
+ SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum);
+
+/* STO1 ADC2 Source */
+/* MX-26 [12] [4] */
+static const char * const rt5682_sto1_adc2_src[] = {
+ "DAC MIX", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER,
+ RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum);
+
+/* MX-79 [6:4] I2S1 ADC data location */
+static const unsigned int rt5682_if1_adc_slot_values[] = {
+ 0,
+ 2,
+ 4,
+ 6,
+};
+
+static const char * const rt5682_if1_adc_slot_src[] = {
+ "Slot 0", "Slot 2", "Slot 4", "Slot 6"
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum,
+ RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK,
+ rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values);
+
+static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux =
+ SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum);
+
+/* Analog DAC L1 Source, Analog DAC R1 Source*/
+/* MX-2B [4], MX-2B [0]*/
+static const char * const rt5682_alg_dac1_src[] = {
+ "Stereo1 DAC Mixer", "DAC1"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX,
+ RT5682_A_DACL1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux =
+ SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX,
+ RT5682_A_DACR1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux =
+ SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum);
+
+/* Out Switch */
+static const struct snd_kcontrol_new hpol_switch =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
+ RT5682_L_MUTE_SFT, 1, 1);
+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_hp_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_write(component,
+ 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;
+
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_component_update_bits(component,
+ RT5682_DEPOP_1, 0x60, 0x0);
+ snd_soc_component_write(component,
+ RT5682_HP_CTRL_2, 0x0000);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+
+}
+
+static int set_dmic_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /*Add delay to avoid pop noise*/
+ msleep(150);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5655_set_verf(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:
+ switch (w->shift) {
+ case RT5682_PWR_VREF1_BIT:
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0);
+ break;
+
+ case RT5682_PWR_VREF2_BIT:
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
+ break;
+
+ default:
+ break;
+ }
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
+ usleep_range(15000, 20000);
+ switch (w->shift) {
+ case RT5682_PWR_VREF1_BIT:
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV1,
+ RT5682_PWR_FV1);
+ break;
+
+ case RT5682_PWR_VREF2_BIT:
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV2,
+ RT5682_PWR_FV2);
+ break;
+
+ default:
+ break;
+ }
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static const unsigned int rt5682_adcdat_pin_values[] = {
+ 1,
+ 3,
+};
+
+static const char * const rt5682_adcdat_pin_select[] = {
+ "ADCDAT1",
+ "ADCDAT2",
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum,
+ RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK,
+ rt5682_adcdat_pin_select, rt5682_adcdat_pin_values);
+
+static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl =
+ SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum);
+
+static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
+ rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
+ rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+ /* ASRC */
+ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+ RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+ RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1,
+ RT5682_AD_ASRC_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1,
+ RT5682_DA_ASRC_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1,
+ RT5682_DMIC_ASRC_SFT, 0, NULL, 0),
+
+ /* Input Side */
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT,
+ 0, NULL, 0),
+
+ /* Input Lines */
+ SND_SOC_DAPM_INPUT("DMIC L1"),
+ SND_SOC_DAPM_INPUT("DMIC R1"),
+
+ SND_SOC_DAPM_INPUT("IN1P"),
+
+ SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
+ set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1,
+ RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU),
+
+ /* Boost */
+ SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM,
+ 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3,
+ RT5682_PWR_CBJ_BIT, 0, NULL, 0),
+
+ /* REC Mixer */
+ SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix,
+ ARRAY_SIZE(rt5682_rec1_l_mix)),
+ SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2,
+ RT5682_PWR_RM1_L_BIT, 0, NULL, 0),
+
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1,
+ RT5682_PWR_ADC_L1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1,
+ RT5682_PWR_ADC_R1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC,
+ RT5682_CKGEN_ADC1_SFT, 0, NULL, 0),
+
+ /* ADC Mux */
+ SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adc1l_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adc1r_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adc2l_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adc2r_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adcl_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_sto1_adcr_mux),
+ SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if1_adc_slot_mux),
+
+ /* ADC Mixer */
+ SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2,
+ RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk,
+ SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL,
+ RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix,
+ ARRAY_SIZE(rt5682_sto1_adc_l_mix)),
+ SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL,
+ RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix,
+ ARRAY_SIZE(rt5682_sto1_adc_r_mix)),
+ SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1,
+ 14, 1, NULL, 0),
+
+ /* ADC PGA */
+ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Digital Interface */
+ SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT,
+ 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Digital Interface Select */
+ SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if1_01_adc_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if1_23_adc_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if1_45_adc_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if1_67_adc_swap_mux),
+ SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_if2_adc_swap_mux),
+
+ SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_adcdat_pin_ctrl),
+
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0,
+ RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1),
+ SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
+ RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1),
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Output Side */
+ /* DAC mixer before sound effect */
+ SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0,
+ rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)),
+ SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0,
+ rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)),
+
+ /* DAC channel Mux */
+ SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0,
+ &rt5682_alg_dac_l1_mux),
+ SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0,
+ &rt5682_alg_dac_r1_mux),
+
+ /* DAC Mixer */
+ SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2,
+ RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk,
+ SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)),
+ SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1,
+ RT5682_PWR_DAC_L1_BIT, 0),
+ SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1,
+ RT5682_PWR_DAC_R1_BIT, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC,
+ RT5682_CKGEN_DAC1_SFT, 0, NULL, 0),
+
+ /* HPO */
+ SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event,
+ SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU),
+
+ SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1,
+ RT5682_PWR_HA_L_BIT, 0, NULL, 0),
+ 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),
+ SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
+ RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0,
+ &hpol_switch),
+ SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0,
+ &hpor_switch),
+
+ /* CLK DET */
+ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET,
+ RT5682_SYS_CLK_DET_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET,
+ RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET,
+ RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET,
+ RT5682_POW_CLK_DET_SFT, 0, NULL, 0),
+
+ /* Output Lines */
+ SND_SOC_DAPM_OUTPUT("HPOL"),
+ SND_SOC_DAPM_OUTPUT("HPOR"),
+
+};
+
+static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
+ /*PLL*/
+ {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+ {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+
+ /*ASRC*/
+ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
+ {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc},
+ {"ADC STO1 ASRC", NULL, "AD ASRC"},
+ {"ADC STO1 ASRC", NULL, "CLKDET"},
+ {"DAC STO1 ASRC", NULL, "DA ASRC"},
+ {"DAC STO1 ASRC", NULL, "CLKDET"},
+
+ /*Vref*/
+ {"MICBIAS1", NULL, "Vref1"},
+ {"MICBIAS1", NULL, "Vref2"},
+ {"MICBIAS2", NULL, "Vref1"},
+ {"MICBIAS2", NULL, "Vref2"},
+
+ {"CLKDET SYS", NULL, "CLKDET"},
+
+ {"IN1P", NULL, "LDO2"},
+
+ {"BST1 CBJ", NULL, "IN1P"},
+ {"BST1 CBJ", NULL, "CBJ Power"},
+ {"CBJ Power", NULL, "Vref2"},
+
+ {"RECMIX1L", "CBJ Switch", "BST1 CBJ"},
+ {"RECMIX1L", NULL, "RECMIX1L Power"},
+
+ {"ADC1 L", NULL, "RECMIX1L"},
+ {"ADC1 L", NULL, "ADC1 L Power"},
+ {"ADC1 L", NULL, "ADC1 clock"},
+
+ {"DMIC L1", NULL, "DMIC CLK"},
+ {"DMIC L1", NULL, "DMIC1 Power"},
+ {"DMIC R1", NULL, "DMIC CLK"},
+ {"DMIC R1", NULL, "DMIC1 Power"},
+ {"DMIC CLK", NULL, "DMIC ASRC"},
+
+ {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"},
+ {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"},
+ {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"},
+ {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"},
+
+ {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"},
+ {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+ {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"},
+ {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+
+ {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"},
+ {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+ {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"},
+ {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+
+ {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"},
+ {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"},
+ {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"},
+
+ {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"},
+ {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"},
+ {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"},
+
+ {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"},
+
+ {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"},
+ {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"},
+
+ {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+ {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+ {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+ {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+ {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+ {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+ {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+ {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+ {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+ {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+ {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+ {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+ {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+ {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+ {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+ {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+
+ {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"},
+ {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"},
+ {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"},
+ {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"},
+ {"IF1_ADC Mux", NULL, "I2S1"},
+ {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"},
+ {"AIF1TX", NULL, "ADCDAT Mux"},
+ {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+ {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+ {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+ {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+ {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"},
+ {"AIF2TX", NULL, "ADCDAT Mux"},
+
+ {"IF1 DAC1 L", NULL, "AIF1RX"},
+ {"IF1 DAC1 L", NULL, "I2S1"},
+ {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"},
+ {"IF1 DAC1 R", NULL, "AIF1RX"},
+ {"IF1 DAC1 R", NULL, "I2S1"},
+ {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"},
+
+ {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"},
+ {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"},
+ {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"},
+ {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"},
+
+ {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"},
+ {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"},
+
+ {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"},
+ {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"},
+
+ {"DAC L1 Source", "DAC1", "DAC1 MIXL"},
+ {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"},
+ {"DAC R1 Source", "DAC1", "DAC1 MIXR"},
+ {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"},
+
+ {"DAC L1", NULL, "DAC L1 Source"},
+ {"DAC R1", NULL, "DAC R1 Source"},
+
+ {"DAC L1", NULL, "DAC 1 Clock"},
+ {"DAC R1", NULL, "DAC 1 Clock"},
+
+ {"HP Amp", NULL, "DAC L1"},
+ {"HP Amp", NULL, "DAC R1"},
+ {"HP Amp", NULL, "HP Amp L"},
+ {"HP Amp", NULL, "HP Amp R"},
+ {"HP Amp", NULL, "Capless"},
+ {"HP Amp", NULL, "Charge Pump"},
+ {"HP Amp", NULL, "CLKDET SYS"},
+ {"HP Amp", NULL, "CBJ Power"},
+ {"HP Amp", NULL, "Vref2"},
+ {"HPOL Playback", "Switch", "HP Amp"},
+ {"HPOR Playback", "Switch", "HP Amp"},
+ {"HPOL", NULL, "HPOL Playback"},
+ {"HPOR", NULL, "HPOR Playback"},
+};
+
+static int rt5682_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;
+ unsigned int cl, val = 0;
+
+ if (tx_mask || rx_mask)
+ snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+ RT5682_TDM_EN, RT5682_TDM_EN);
+ else
+ snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+ RT5682_TDM_EN, 0);
+
+ switch (slots) {
+ case 4:
+ val |= RT5682_TDM_TX_CH_4;
+ val |= RT5682_TDM_RX_CH_4;
+ break;
+ case 6:
+ val |= RT5682_TDM_TX_CH_6;
+ val |= RT5682_TDM_RX_CH_6;
+ break;
+ case 8:
+ val |= RT5682_TDM_TX_CH_8;
+ val |= RT5682_TDM_RX_CH_8;
+ break;
+ case 2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, RT5682_TDM_CTRL,
+ RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val);
+
+ switch (slot_width) {
+ case 8:
+ if (tx_mask || rx_mask)
+ return -EINVAL;
+ cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8;
+ break;
+ case 16:
+ val = RT5682_TDM_CL_16;
+ cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16;
+ break;
+ case 20:
+ val = RT5682_TDM_CL_20;
+ cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20;
+ break;
+ case 24:
+ val = RT5682_TDM_CL_24;
+ cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24;
+ break;
+ case 32:
+ val = RT5682_TDM_CL_32;
+ cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_CL_MASK, val);
+ snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+ RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl);
+
+ return 0;
+}
+
+
+static int rt5682_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 rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ unsigned int len_1 = 0, len_2 = 0;
+ int pre_div, frame_size;
+
+ rt5682->lrck[dai->id] = params_rate(params);
+ pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]);
+
+ frame_size = snd_soc_params_to_frame_size(params);
+ if (frame_size < 0) {
+ dev_err(component->dev, "Unsupported frame size: %d\n",
+ frame_size);
+ return -EINVAL;
+ }
+
+ dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n",
+ rt5682->lrck[dai->id], pre_div, dai->id);
+
+ switch (params_width(params)) {
+ case 16:
+ break;
+ case 20:
+ len_1 |= RT5682_I2S1_DL_20;
+ len_2 |= RT5682_I2S2_DL_20;
+ break;
+ case 24:
+ len_1 |= RT5682_I2S1_DL_24;
+ len_2 |= RT5682_I2S2_DL_24;
+ break;
+ case 32:
+ len_1 |= RT5682_I2S1_DL_32;
+ len_2 |= RT5682_I2S2_DL_24;
+ break;
+ case 8:
+ len_1 |= RT5682_I2S2_DL_8;
+ len_2 |= RT5682_I2S2_DL_8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (dai->id) {
+ case RT5682_AIF1:
+ snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+ RT5682_I2S1_DL_MASK, len_1);
+ if (rt5682->master[RT5682_AIF1]) {
+ snd_soc_component_update_bits(component,
+ RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK,
+ pre_div << RT5682_I2S_M_DIV_SFT);
+ }
+ if (params_channels(params) == 1) /* mono mode */
+ snd_soc_component_update_bits(component,
+ RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+ RT5682_I2S1_MONO_EN);
+ else
+ snd_soc_component_update_bits(component,
+ RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+ RT5682_I2S1_MONO_DIS);
+ break;
+ case RT5682_AIF2:
+ snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+ RT5682_I2S2_DL_MASK, len_2);
+ if (rt5682->master[RT5682_AIF2]) {
+ snd_soc_component_update_bits(component,
+ RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK,
+ pre_div << RT5682_I2S2_M_PD_SFT);
+ }
+ if (params_channels(params) == 1) /* mono mode */
+ snd_soc_component_update_bits(component,
+ RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+ RT5682_I2S2_MONO_EN);
+ else
+ snd_soc_component_update_bits(component,
+ RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+ RT5682_I2S2_MONO_DIS);
+ break;
+ default:
+ dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ unsigned int reg_val = 0, tdm_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ rt5682->master[dai->id] = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ rt5682->master[dai->id] = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ reg_val |= RT5682_I2S_BP_INV;
+ tdm_ctrl |= RT5682_TDM_S_BP_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ if (dai->id == RT5682_AIF1)
+ tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV;
+ else
+ return -EINVAL;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ if (dai->id == RT5682_AIF1)
+ tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV |
+ RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV;
+ else
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ reg_val |= RT5682_I2S_DF_LEFT;
+ tdm_ctrl |= RT5682_TDM_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ reg_val |= RT5682_I2S_DF_PCM_A;
+ tdm_ctrl |= RT5682_TDM_DF_PCM_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ reg_val |= RT5682_I2S_DF_PCM_B;
+ tdm_ctrl |= RT5682_TDM_DF_PCM_B;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (dai->id) {
+ case RT5682_AIF1:
+ snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+ RT5682_I2S_DF_MASK, reg_val);
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK |
+ RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK |
+ RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK,
+ tdm_ctrl | rt5682->master[dai->id]);
+ break;
+ case RT5682_AIF2:
+ if (rt5682->master[dai->id] == 0)
+ reg_val |= RT5682_I2S2_MS_S;
+ snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+ RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK |
+ RT5682_I2S_DF_MASK, reg_val);
+ break;
+ default:
+ dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int rt5682_set_component_sysclk(struct snd_soc_component *component,
+ int clk_id, int source, unsigned int freq, int dir)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ unsigned int reg_val = 0, src = 0;
+
+ if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src)
+ return 0;
+
+ switch (clk_id) {
+ case RT5682_SCLK_S_MCLK:
+ reg_val |= RT5682_SCLK_SRC_MCLK;
+ src = RT5682_CLK_SRC_MCLK;
+ break;
+ case RT5682_SCLK_S_PLL1:
+ reg_val |= RT5682_SCLK_SRC_PLL1;
+ src = RT5682_CLK_SRC_PLL1;
+ break;
+ case RT5682_SCLK_S_PLL2:
+ reg_val |= RT5682_SCLK_SRC_PLL2;
+ src = RT5682_CLK_SRC_PLL2;
+ break;
+ case RT5682_SCLK_S_RCCLK:
+ reg_val |= RT5682_SCLK_SRC_RCCLK;
+ src = RT5682_CLK_SRC_RCCLK;
+ break;
+ default:
+ dev_err(component->dev, "Invalid clock id (%d)\n", clk_id);
+ return -EINVAL;
+ }
+ snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+ RT5682_SCLK_SRC_MASK, reg_val);
+
+ if (rt5682->master[RT5682_AIF2]) {
+ snd_soc_component_update_bits(component,
+ RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK,
+ src << RT5682_I2S2_SRC_SFT);
+ }
+
+ rt5682->sysclk = freq;
+ rt5682->sysclk_src = clk_id;
+
+ dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n",
+ freq, clk_id);
+
+ return 0;
+}
+
+static int rt5682_set_component_pll(struct snd_soc_component *component,
+ int pll_id, int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct rl6231_pll_code pll_code;
+ int ret;
+
+ if (source == rt5682->pll_src && freq_in == rt5682->pll_in &&
+ freq_out == rt5682->pll_out)
+ return 0;
+
+ if (!freq_in || !freq_out) {
+ dev_dbg(component->dev, "PLL disabled\n");
+
+ rt5682->pll_in = 0;
+ rt5682->pll_out = 0;
+ snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+ RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK);
+ return 0;
+ }
+
+ switch (source) {
+ case RT5682_PLL1_S_MCLK:
+ snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+ RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK);
+ break;
+ case RT5682_PLL1_S_BCLK1:
+ snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+ RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1);
+ break;
+ default:
+ dev_err(component->dev, "Unknown PLL Source %d\n", source);
+ return -EINVAL;
+ }
+
+ ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+ if (ret < 0) {
+ dev_err(component->dev, "Unsupport input clock %d\n", freq_in);
+ return ret;
+ }
+
+ dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
+ pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+ pll_code.n_code, pll_code.k_code);
+
+ snd_soc_component_write(component, RT5682_PLL_CTRL_1,
+ pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
+ snd_soc_component_write(component, RT5682_PLL_CTRL_2,
+ (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
+ pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
+
+ rt5682->pll_in = freq_in;
+ rt5682->pll_out = freq_out;
+ rt5682->pll_src = source;
+
+ return 0;
+}
+
+static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ rt5682->bclk[dai->id] = ratio;
+
+ switch (ratio) {
+ case 64:
+ snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+ RT5682_I2S2_BCLK_MS2_MASK,
+ RT5682_I2S2_BCLK_MS2_64);
+ break;
+ case 32:
+ snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+ RT5682_I2S2_BCLK_MS2_MASK,
+ RT5682_I2S2_BCLK_MS2_32);
+ break;
+ default:
+ dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int rt5682_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+ RT5682_PWR_MB | RT5682_PWR_BG,
+ RT5682_PWR_MB | RT5682_PWR_BG);
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+ RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO,
+ RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+ RT5682_PWR_MB, RT5682_PWR_MB);
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+ RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL);
+ break;
+ case SND_SOC_BIAS_OFF:
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+ RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0);
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+ RT5682_PWR_MB | RT5682_PWR_BG, 0);
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int rt5682_probe(struct snd_soc_component *component)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ rt5682->component = component;
+
+ return 0;
+}
+
+static void rt5682_remove(struct snd_soc_component *component)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_PM
+static int rt5682_suspend(struct snd_soc_component *component)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ regcache_cache_only(rt5682->regmap, true);
+ regcache_mark_dirty(rt5682->regmap);
+ return 0;
+}
+
+static int rt5682_resume(struct snd_soc_component *component)
+{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ regcache_cache_only(rt5682->regmap, false);
+ regcache_sync(rt5682->regmap);
+
+ return 0;
+}
+#else
+#define rt5682_suspend NULL
+#define rt5682_resume NULL
+#endif
+
+#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000
+#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = {
+ .hw_params = rt5682_hw_params,
+ .set_fmt = rt5682_set_dai_fmt,
+ .set_tdm_slot = rt5682_set_tdm_slot,
+};
+
+static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = {
+ .hw_params = rt5682_hw_params,
+ .set_fmt = rt5682_set_dai_fmt,
+ .set_bclk_ratio = rt5682_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt5682_dai[] = {
+ {
+ .name = "rt5682-aif1",
+ .id = RT5682_AIF1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5682_STEREO_RATES,
+ .formats = RT5682_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5682_STEREO_RATES,
+ .formats = RT5682_FORMATS,
+ },
+ .ops = &rt5682_aif1_dai_ops,
+ },
+ {
+ .name = "rt5682-aif2",
+ .id = RT5682_AIF2,
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5682_STEREO_RATES,
+ .formats = RT5682_FORMATS,
+ },
+ .ops = &rt5682_aif2_dai_ops,
+ },
+};
+
+static const struct snd_soc_component_driver soc_component_dev_rt5682 = {
+ .probe = rt5682_probe,
+ .remove = rt5682_remove,
+ .suspend = rt5682_suspend,
+ .resume = rt5682_resume,
+ .set_bias_level = rt5682_set_bias_level,
+ .controls = rt5682_snd_controls,
+ .num_controls = ARRAY_SIZE(rt5682_snd_controls),
+ .dapm_widgets = rt5682_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets),
+ .dapm_routes = rt5682_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes),
+ .set_sysclk = rt5682_set_component_sysclk,
+ .set_pll = rt5682_set_component_pll,
+ .set_jack = rt5682_set_jack_detect,
+ .use_pmdown_time = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static const struct regmap_config rt5682_regmap = {
+ .reg_bits = 16,
+ .val_bits = 16,
+ .max_register = RT5682_I2C_MODE,
+ .volatile_reg = rt5682_volatile_register,
+ .readable_reg = rt5682_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = rt5682_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt5682_reg),
+ .use_single_rw = true,
+};
+
+static const struct i2c_device_id rt5682_i2c_id[] = {
+ {"rt5682", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id);
+
+static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev)
+{
+
+ device_property_read_u32(dev, "realtek,dmic1-data-pin",
+ &rt5682->pdata.dmic1_data_pin);
+ device_property_read_u32(dev, "realtek,dmic1-clk-pin",
+ &rt5682->pdata.dmic1_clk_pin);
+ device_property_read_u32(dev, "realtek,jd-src",
+ &rt5682->pdata.jd_src);
+
+ rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node,
+ "realtek,ldo1-en-gpios", 0);
+
+ return 0;
+}
+
+static void rt5682_calibrate(struct rt5682_priv *rt5682)
+{
+ int value, count;
+
+ mutex_lock(&rt5682->calibrate_mutex);
+
+ rt5682_reset(rt5682->regmap);
+ 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_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, 0xfc00);
+
+ for (count = 0; count < 60; count++) {
+ regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value);
+ if (!(value & 0x8000))
+ break;
+
+ usleep_range(10000, 10005);
+ }
+
+ if (count >= 60)
+ pr_err("HP Calibration Failure\n");
+
+ /* restore settings */
+ regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+ regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+
+ mutex_unlock(&rt5682->calibrate_mutex);
+
+}
+
+static int rt5682_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev);
+ struct rt5682_priv *rt5682;
+ int i, ret;
+ unsigned int val;
+
+ rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv),
+ GFP_KERNEL);
+
+ if (rt5682 == NULL)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, rt5682);
+
+ if (pdata)
+ rt5682->pdata = *pdata;
+ else
+ rt5682_parse_dt(rt5682, &i2c->dev);
+
+ rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap);
+ if (IS_ERR(rt5682->regmap)) {
+ ret = PTR_ERR(rt5682->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++)
+ rt5682->supplies[i].supply = rt5682_supply_names[i];
+
+ ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies),
+ rt5682->supplies);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies),
+ rt5682->supplies);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ if (gpio_is_valid(rt5682->pdata.ldo1_en)) {
+ if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en,
+ GPIOF_OUT_INIT_HIGH, "rt5682"))
+ dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n");
+ }
+
+ /* Sleep for 300 ms miniumum */
+ usleep_range(300000, 350000);
+
+ regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1);
+ usleep_range(10000, 15000);
+
+ regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val);
+ if (val != DEVICE_ID) {
+ pr_err("Device with ID register %x is not rt5682\n", val);
+ return -ENODEV;
+ }
+
+ rt5682_reset(rt5682->regmap);
+
+ rt5682_calibrate(rt5682);
+
+ ret = regmap_register_patch(rt5682->regmap, patch_list,
+ ARRAY_SIZE(patch_list));
+ if (ret != 0)
+ dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
+ regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000);
+
+ /* DMIC pin*/
+ if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) {
+ switch (rt5682->pdata.dmic1_data_pin) {
+ case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */
+ regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+ RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2);
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA);
+ break;
+
+ case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */
+ regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+ RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5);
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA);
+ break;
+
+ default:
+ dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n");
+ break;
+ }
+
+ switch (rt5682->pdata.dmic1_clk_pin) {
+ case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK);
+ break;
+
+ case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK);
+ break;
+
+ default:
+ dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n");
+ break;
+ }
+ }
+
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+ RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK,
+ RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X);
+ regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ 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);
+
+ INIT_DELAYED_WORK(&rt5682->jack_detect_work,
+ rt5682_jack_detect_handler);
+ INIT_DELAYED_WORK(&rt5682->jd_check_work,
+ rt5682_jd_check_handler);
+
+ mutex_init(&rt5682->calibrate_mutex);
+
+ if (i2c->irq) {
+ ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+ rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+ | IRQF_ONESHOT, "rt5682", rt5682);
+ if (ret)
+ dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
+
+ }
+
+ return devm_snd_soc_register_component(&i2c->dev,
+ &soc_component_dev_rt5682,
+ rt5682_dai, ARRAY_SIZE(rt5682_dai));
+}
+
+static void rt5682_i2c_shutdown(struct i2c_client *client)
+{
+ struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
+
+ rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id rt5682_of_match[] = {
+ {.compatible = "realtek,rt5682i"},
+ {},
+};
+MODULE_DEVICE_TABLE(of, rt5682_of_match);
+#endif
+
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id rt5682_acpi_match[] = {
+ {"10EC5682", 0,},
+ {},
+};
+MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match);
+#endif
+
+static struct i2c_driver rt5682_i2c_driver = {
+ .driver = {
+ .name = "rt5682",
+ .of_match_table = of_match_ptr(rt5682_of_match),
+ .acpi_match_table = ACPI_PTR(rt5682_acpi_match),
+ },
+ .probe = rt5682_i2c_probe,
+ .shutdown = rt5682_i2c_shutdown,
+ .id_table = rt5682_i2c_id,
+};
+module_i2c_driver(rt5682_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT5682 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
new file mode 100644
index 000000000000..8068140ebe3f
--- /dev/null
+++ b/sound/soc/codecs/rt5682.h
@@ -0,0 +1,1324 @@
+/*
+ * rt5682.h -- RT5682/RT5658 ALSA SoC audio driver
+ *
+ * Copyright 2018 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * 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 __RT5682_H__
+#define __RT5682_H__
+
+#include <sound/rt5682.h>
+
+#define DEVICE_ID 0x6530
+
+/* Info */
+#define RT5682_RESET 0x0000
+#define RT5682_VERSION_ID 0x00fd
+#define RT5682_VENDOR_ID 0x00fe
+#define RT5682_DEVICE_ID 0x00ff
+/* I/O - Output */
+#define RT5682_HP_CTRL_1 0x0002
+#define RT5682_HP_CTRL_2 0x0003
+#define RT5682_HPL_GAIN 0x0005
+#define RT5682_HPR_GAIN 0x0006
+
+#define RT5682_I2C_CTRL 0x0008
+
+/* I/O - Input */
+#define RT5682_CBJ_BST_CTRL 0x000b
+#define RT5682_CBJ_CTRL_1 0x0010
+#define RT5682_CBJ_CTRL_2 0x0011
+#define RT5682_CBJ_CTRL_3 0x0012
+#define RT5682_CBJ_CTRL_4 0x0013
+#define RT5682_CBJ_CTRL_5 0x0014
+#define RT5682_CBJ_CTRL_6 0x0015
+#define RT5682_CBJ_CTRL_7 0x0016
+/* I/O - ADC/DAC/DMIC */
+#define RT5682_DAC1_DIG_VOL 0x0019
+#define RT5682_STO1_ADC_DIG_VOL 0x001c
+#define RT5682_STO1_ADC_BOOST 0x001f
+#define RT5682_HP_IMP_GAIN_1 0x0022
+#define RT5682_HP_IMP_GAIN_2 0x0023
+/* Mixer - D-D */
+#define RT5682_SIDETONE_CTRL 0x0024
+#define RT5682_STO1_ADC_MIXER 0x0026
+#define RT5682_AD_DA_MIXER 0x0029
+#define RT5682_STO1_DAC_MIXER 0x002a
+#define RT5682_A_DAC1_MUX 0x002b
+#define RT5682_DIG_INF2_DATA 0x0030
+/* Mixer - ADC */
+#define RT5682_REC_MIXER 0x003c
+#define RT5682_CAL_REC 0x0044
+#define RT5682_ALC_BACK_GAIN 0x0049
+/* Power */
+#define RT5682_PWR_DIG_1 0x0061
+#define RT5682_PWR_DIG_2 0x0062
+#define RT5682_PWR_ANLG_1 0x0063
+#define RT5682_PWR_ANLG_2 0x0064
+#define RT5682_PWR_ANLG_3 0x0065
+#define RT5682_PWR_MIXER 0x0066
+#define RT5682_PWR_VOL 0x0067
+/* Clock Detect */
+#define RT5682_CLK_DET 0x006b
+/* Filter Auto Reset */
+#define RT5682_RESET_LPF_CTRL 0x006c
+#define RT5682_RESET_HPF_CTRL 0x006d
+/* DMIC */
+#define RT5682_DMIC_CTRL_1 0x006e
+/* Format - ADC/DAC */
+#define RT5682_I2S1_SDP 0x0070
+#define RT5682_I2S2_SDP 0x0071
+#define RT5682_ADDA_CLK_1 0x0073
+#define RT5682_ADDA_CLK_2 0x0074
+#define RT5682_I2S1_F_DIV_CTRL_1 0x0075
+#define RT5682_I2S1_F_DIV_CTRL_2 0x0076
+/* Format - TDM Control */
+#define RT5682_TDM_CTRL 0x0079
+#define RT5682_TDM_ADDA_CTRL_1 0x007a
+#define RT5682_TDM_ADDA_CTRL_2 0x007b
+#define RT5682_DATA_SEL_CTRL_1 0x007c
+#define RT5682_TDM_TCON_CTRL 0x007e
+/* Function - Analog */
+#define RT5682_GLB_CLK 0x0080
+#define RT5682_PLL_CTRL_1 0x0081
+#define RT5682_PLL_CTRL_2 0x0082
+#define RT5682_PLL_TRACK_1 0x0083
+#define RT5682_PLL_TRACK_2 0x0084
+#define RT5682_PLL_TRACK_3 0x0085
+#define RT5682_PLL_TRACK_4 0x0086
+#define RT5682_PLL_TRACK_5 0x0087
+#define RT5682_PLL_TRACK_6 0x0088
+#define RT5682_PLL_TRACK_11 0x008c
+#define RT5682_SDW_REF_CLK 0x008d
+#define RT5682_DEPOP_1 0x008e
+#define RT5682_DEPOP_2 0x008f
+#define RT5682_HP_CHARGE_PUMP_1 0x0091
+#define RT5682_HP_CHARGE_PUMP_2 0x0092
+#define RT5682_MICBIAS_1 0x0093
+#define RT5682_MICBIAS_2 0x0094
+#define RT5682_PLL_TRACK_12 0x0098
+#define RT5682_PLL_TRACK_14 0x009a
+#define RT5682_PLL2_CTRL_1 0x009b
+#define RT5682_PLL2_CTRL_2 0x009c
+#define RT5682_PLL2_CTRL_3 0x009d
+#define RT5682_PLL2_CTRL_4 0x009e
+#define RT5682_RC_CLK_CTRL 0x009f
+#define RT5682_I2S_M_CLK_CTRL_1 0x00a0
+#define RT5682_I2S2_F_DIV_CTRL_1 0x00a3
+#define RT5682_I2S2_F_DIV_CTRL_2 0x00a4
+/* Function - Digital */
+#define RT5682_EQ_CTRL_1 0x00ae
+#define RT5682_EQ_CTRL_2 0x00af
+#define RT5682_IRQ_CTRL_1 0x00b6
+#define RT5682_IRQ_CTRL_2 0x00b7
+#define RT5682_IRQ_CTRL_3 0x00b8
+#define RT5682_IRQ_CTRL_4 0x00b9
+#define RT5682_INT_ST_1 0x00be
+#define RT5682_GPIO_CTRL_1 0x00c0
+#define RT5682_GPIO_CTRL_2 0x00c1
+#define RT5682_GPIO_CTRL_3 0x00c2
+#define RT5682_HP_AMP_DET_CTRL_1 0x00d0
+#define RT5682_HP_AMP_DET_CTRL_2 0x00d1
+#define RT5682_MID_HP_AMP_DET 0x00d2
+#define RT5682_LOW_HP_AMP_DET 0x00d3
+#define RT5682_DELAY_BUF_CTRL 0x00d4
+#define RT5682_SV_ZCD_1 0x00d9
+#define RT5682_SV_ZCD_2 0x00da
+#define RT5682_IL_CMD_1 0x00db
+#define RT5682_IL_CMD_2 0x00dc
+#define RT5682_IL_CMD_3 0x00dd
+#define RT5682_IL_CMD_4 0x00de
+#define RT5682_IL_CMD_5 0x00df
+#define RT5682_IL_CMD_6 0x00e0
+#define RT5682_4BTN_IL_CMD_1 0x00e2
+#define RT5682_4BTN_IL_CMD_2 0x00e3
+#define RT5682_4BTN_IL_CMD_3 0x00e4
+#define RT5682_4BTN_IL_CMD_4 0x00e5
+#define RT5682_4BTN_IL_CMD_5 0x00e6
+#define RT5682_4BTN_IL_CMD_6 0x00e7
+#define RT5682_4BTN_IL_CMD_7 0x00e8
+
+#define RT5682_ADC_STO1_HP_CTRL_1 0x00ea
+#define RT5682_ADC_STO1_HP_CTRL_2 0x00eb
+#define RT5682_AJD1_CTRL 0x00f0
+#define RT5682_JD1_THD 0x00f1
+#define RT5682_JD2_THD 0x00f2
+#define RT5682_JD_CTRL_1 0x00f6
+/* General Control */
+#define RT5682_DUMMY_1 0x00fa
+#define RT5682_DUMMY_2 0x00fb
+#define RT5682_DUMMY_3 0x00fc
+
+#define RT5682_DAC_ADC_DIG_VOL1 0x0100
+#define RT5682_BIAS_CUR_CTRL_2 0x010b
+#define RT5682_BIAS_CUR_CTRL_3 0x010c
+#define RT5682_BIAS_CUR_CTRL_4 0x010d
+#define RT5682_BIAS_CUR_CTRL_5 0x010e
+#define RT5682_BIAS_CUR_CTRL_6 0x010f
+#define RT5682_BIAS_CUR_CTRL_7 0x0110
+#define RT5682_BIAS_CUR_CTRL_8 0x0111
+#define RT5682_BIAS_CUR_CTRL_9 0x0112
+#define RT5682_BIAS_CUR_CTRL_10 0x0113
+#define RT5682_VREF_REC_OP_FB_CAP_CTRL 0x0117
+#define RT5682_CHARGE_PUMP_1 0x0125
+#define RT5682_DIG_IN_CTRL_1 0x0132
+#define RT5682_PAD_DRIVING_CTRL 0x0136
+#define RT5682_SOFT_RAMP_DEPOP 0x0138
+#define RT5682_CHOP_DAC 0x013a
+#define RT5682_CHOP_ADC 0x013b
+#define RT5682_CALIB_ADC_CTRL 0x013c
+#define RT5682_VOL_TEST 0x013f
+#define RT5682_SPKVDD_DET_STA 0x0142
+#define RT5682_TEST_MODE_CTRL_1 0x0145
+#define RT5682_TEST_MODE_CTRL_2 0x0146
+#define RT5682_TEST_MODE_CTRL_3 0x0147
+#define RT5682_TEST_MODE_CTRL_4 0x0148
+#define RT5682_TEST_MODE_CTRL_5 0x0149
+#define RT5682_PLL1_INTERNAL 0x0150
+#define RT5682_PLL2_INTERNAL 0x0151
+#define RT5682_STO_NG2_CTRL_1 0x0160
+#define RT5682_STO_NG2_CTRL_2 0x0161
+#define RT5682_STO_NG2_CTRL_3 0x0162
+#define RT5682_STO_NG2_CTRL_4 0x0163
+#define RT5682_STO_NG2_CTRL_5 0x0164
+#define RT5682_STO_NG2_CTRL_6 0x0165
+#define RT5682_STO_NG2_CTRL_7 0x0166
+#define RT5682_STO_NG2_CTRL_8 0x0167
+#define RT5682_STO_NG2_CTRL_9 0x0168
+#define RT5682_STO_NG2_CTRL_10 0x0169
+#define RT5682_STO1_DAC_SIL_DET 0x0190
+#define RT5682_SIL_PSV_CTRL1 0x0194
+#define RT5682_SIL_PSV_CTRL2 0x0195
+#define RT5682_SIL_PSV_CTRL3 0x0197
+#define RT5682_SIL_PSV_CTRL4 0x0198
+#define RT5682_SIL_PSV_CTRL5 0x0199
+#define RT5682_HP_IMP_SENS_CTRL_01 0x01af
+#define RT5682_HP_IMP_SENS_CTRL_02 0x01b0
+#define RT5682_HP_IMP_SENS_CTRL_03 0x01b1
+#define RT5682_HP_IMP_SENS_CTRL_04 0x01b2
+#define RT5682_HP_IMP_SENS_CTRL_05 0x01b3
+#define RT5682_HP_IMP_SENS_CTRL_06 0x01b4
+#define RT5682_HP_IMP_SENS_CTRL_07 0x01b5
+#define RT5682_HP_IMP_SENS_CTRL_08 0x01b6
+#define RT5682_HP_IMP_SENS_CTRL_09 0x01b7
+#define RT5682_HP_IMP_SENS_CTRL_10 0x01b8
+#define RT5682_HP_IMP_SENS_CTRL_11 0x01b9
+#define RT5682_HP_IMP_SENS_CTRL_12 0x01ba
+#define RT5682_HP_IMP_SENS_CTRL_13 0x01bb
+#define RT5682_HP_IMP_SENS_CTRL_14 0x01bc
+#define RT5682_HP_IMP_SENS_CTRL_15 0x01bd
+#define RT5682_HP_IMP_SENS_CTRL_16 0x01be
+#define RT5682_HP_IMP_SENS_CTRL_17 0x01bf
+#define RT5682_HP_IMP_SENS_CTRL_18 0x01c0
+#define RT5682_HP_IMP_SENS_CTRL_19 0x01c1
+#define RT5682_HP_IMP_SENS_CTRL_20 0x01c2
+#define RT5682_HP_IMP_SENS_CTRL_21 0x01c3
+#define RT5682_HP_IMP_SENS_CTRL_22 0x01c4
+#define RT5682_HP_IMP_SENS_CTRL_23 0x01c5
+#define RT5682_HP_IMP_SENS_CTRL_24 0x01c6
+#define RT5682_HP_IMP_SENS_CTRL_25 0x01c7
+#define RT5682_HP_IMP_SENS_CTRL_26 0x01c8
+#define RT5682_HP_IMP_SENS_CTRL_27 0x01c9
+#define RT5682_HP_IMP_SENS_CTRL_28 0x01ca
+#define RT5682_HP_IMP_SENS_CTRL_29 0x01cb
+#define RT5682_HP_IMP_SENS_CTRL_30 0x01cc
+#define RT5682_HP_IMP_SENS_CTRL_31 0x01cd
+#define RT5682_HP_IMP_SENS_CTRL_32 0x01ce
+#define RT5682_HP_IMP_SENS_CTRL_33 0x01cf
+#define RT5682_HP_IMP_SENS_CTRL_34 0x01d0
+#define RT5682_HP_IMP_SENS_CTRL_35 0x01d1
+#define RT5682_HP_IMP_SENS_CTRL_36 0x01d2
+#define RT5682_HP_IMP_SENS_CTRL_37 0x01d3
+#define RT5682_HP_IMP_SENS_CTRL_38 0x01d4
+#define RT5682_HP_IMP_SENS_CTRL_39 0x01d5
+#define RT5682_HP_IMP_SENS_CTRL_40 0x01d6
+#define RT5682_HP_IMP_SENS_CTRL_41 0x01d7
+#define RT5682_HP_IMP_SENS_CTRL_42 0x01d8
+#define RT5682_HP_IMP_SENS_CTRL_43 0x01d9
+#define RT5682_HP_LOGIC_CTRL_1 0x01da
+#define RT5682_HP_LOGIC_CTRL_2 0x01db
+#define RT5682_HP_LOGIC_CTRL_3 0x01dc
+#define RT5682_HP_CALIB_CTRL_1 0x01de
+#define RT5682_HP_CALIB_CTRL_2 0x01df
+#define RT5682_HP_CALIB_CTRL_3 0x01e0
+#define RT5682_HP_CALIB_CTRL_4 0x01e1
+#define RT5682_HP_CALIB_CTRL_5 0x01e2
+#define RT5682_HP_CALIB_CTRL_6 0x01e3
+#define RT5682_HP_CALIB_CTRL_7 0x01e4
+#define RT5682_HP_CALIB_CTRL_9 0x01e6
+#define RT5682_HP_CALIB_CTRL_10 0x01e7
+#define RT5682_HP_CALIB_CTRL_11 0x01e8
+#define RT5682_HP_CALIB_STA_1 0x01ea
+#define RT5682_HP_CALIB_STA_2 0x01eb
+#define RT5682_HP_CALIB_STA_3 0x01ec
+#define RT5682_HP_CALIB_STA_4 0x01ed
+#define RT5682_HP_CALIB_STA_5 0x01ee
+#define RT5682_HP_CALIB_STA_6 0x01ef
+#define RT5682_HP_CALIB_STA_7 0x01f0
+#define RT5682_HP_CALIB_STA_8 0x01f1
+#define RT5682_HP_CALIB_STA_9 0x01f2
+#define RT5682_HP_CALIB_STA_10 0x01f3
+#define RT5682_HP_CALIB_STA_11 0x01f4
+#define RT5682_SAR_IL_CMD_1 0x0210
+#define RT5682_SAR_IL_CMD_2 0x0211
+#define RT5682_SAR_IL_CMD_3 0x0212
+#define RT5682_SAR_IL_CMD_4 0x0213
+#define RT5682_SAR_IL_CMD_5 0x0214
+#define RT5682_SAR_IL_CMD_6 0x0215
+#define RT5682_SAR_IL_CMD_7 0x0216
+#define RT5682_SAR_IL_CMD_8 0x0217
+#define RT5682_SAR_IL_CMD_9 0x0218
+#define RT5682_SAR_IL_CMD_10 0x0219
+#define RT5682_SAR_IL_CMD_11 0x021a
+#define RT5682_SAR_IL_CMD_12 0x021b
+#define RT5682_SAR_IL_CMD_13 0x021c
+#define RT5682_EFUSE_CTRL_1 0x0250
+#define RT5682_EFUSE_CTRL_2 0x0251
+#define RT5682_EFUSE_CTRL_3 0x0252
+#define RT5682_EFUSE_CTRL_4 0x0253
+#define RT5682_EFUSE_CTRL_5 0x0254
+#define RT5682_EFUSE_CTRL_6 0x0255
+#define RT5682_EFUSE_CTRL_7 0x0256
+#define RT5682_EFUSE_CTRL_8 0x0257
+#define RT5682_EFUSE_CTRL_9 0x0258
+#define RT5682_EFUSE_CTRL_10 0x0259
+#define RT5682_EFUSE_CTRL_11 0x025a
+#define RT5682_JD_TOP_VC_VTRL 0x0270
+#define RT5682_DRC1_CTRL_0 0x02ff
+#define RT5682_DRC1_CTRL_1 0x0300
+#define RT5682_DRC1_CTRL_2 0x0301
+#define RT5682_DRC1_CTRL_3 0x0302
+#define RT5682_DRC1_CTRL_4 0x0303
+#define RT5682_DRC1_CTRL_5 0x0304
+#define RT5682_DRC1_CTRL_6 0x0305
+#define RT5682_DRC1_HARD_LMT_CTRL_1 0x0306
+#define RT5682_DRC1_HARD_LMT_CTRL_2 0x0307
+#define RT5682_DRC1_PRIV_1 0x0310
+#define RT5682_DRC1_PRIV_2 0x0311
+#define RT5682_DRC1_PRIV_3 0x0312
+#define RT5682_DRC1_PRIV_4 0x0313
+#define RT5682_DRC1_PRIV_5 0x0314
+#define RT5682_DRC1_PRIV_6 0x0315
+#define RT5682_DRC1_PRIV_7 0x0316
+#define RT5682_DRC1_PRIV_8 0x0317
+#define RT5682_EQ_AUTO_RCV_CTRL1 0x03c0
+#define RT5682_EQ_AUTO_RCV_CTRL2 0x03c1
+#define RT5682_EQ_AUTO_RCV_CTRL3 0x03c2
+#define RT5682_EQ_AUTO_RCV_CTRL4 0x03c3
+#define RT5682_EQ_AUTO_RCV_CTRL5 0x03c4
+#define RT5682_EQ_AUTO_RCV_CTRL6 0x03c5
+#define RT5682_EQ_AUTO_RCV_CTRL7 0x03c6
+#define RT5682_EQ_AUTO_RCV_CTRL8 0x03c7
+#define RT5682_EQ_AUTO_RCV_CTRL9 0x03c8
+#define RT5682_EQ_AUTO_RCV_CTRL10 0x03c9
+#define RT5682_EQ_AUTO_RCV_CTRL11 0x03ca
+#define RT5682_EQ_AUTO_RCV_CTRL12 0x03cb
+#define RT5682_EQ_AUTO_RCV_CTRL13 0x03cc
+#define RT5682_ADC_L_EQ_LPF1_A1 0x03d0
+#define RT5682_R_EQ_LPF1_A1 0x03d1
+#define RT5682_L_EQ_LPF1_H0 0x03d2
+#define RT5682_R_EQ_LPF1_H0 0x03d3
+#define RT5682_L_EQ_BPF1_A1 0x03d4
+#define RT5682_R_EQ_BPF1_A1 0x03d5
+#define RT5682_L_EQ_BPF1_A2 0x03d6
+#define RT5682_R_EQ_BPF1_A2 0x03d7
+#define RT5682_L_EQ_BPF1_H0 0x03d8
+#define RT5682_R_EQ_BPF1_H0 0x03d9
+#define RT5682_L_EQ_BPF2_A1 0x03da
+#define RT5682_R_EQ_BPF2_A1 0x03db
+#define RT5682_L_EQ_BPF2_A2 0x03dc
+#define RT5682_R_EQ_BPF2_A2 0x03dd
+#define RT5682_L_EQ_BPF2_H0 0x03de
+#define RT5682_R_EQ_BPF2_H0 0x03df
+#define RT5682_L_EQ_BPF3_A1 0x03e0
+#define RT5682_R_EQ_BPF3_A1 0x03e1
+#define RT5682_L_EQ_BPF3_A2 0x03e2
+#define RT5682_R_EQ_BPF3_A2 0x03e3
+#define RT5682_L_EQ_BPF3_H0 0x03e4
+#define RT5682_R_EQ_BPF3_H0 0x03e5
+#define RT5682_L_EQ_BPF4_A1 0x03e6
+#define RT5682_R_EQ_BPF4_A1 0x03e7
+#define RT5682_L_EQ_BPF4_A2 0x03e8
+#define RT5682_R_EQ_BPF4_A2 0x03e9
+#define RT5682_L_EQ_BPF4_H0 0x03ea
+#define RT5682_R_EQ_BPF4_H0 0x03eb
+#define RT5682_L_EQ_HPF1_A1 0x03ec
+#define RT5682_R_EQ_HPF1_A1 0x03ed
+#define RT5682_L_EQ_HPF1_H0 0x03ee
+#define RT5682_R_EQ_HPF1_H0 0x03ef
+#define RT5682_L_EQ_PRE_VOL 0x03f0
+#define RT5682_R_EQ_PRE_VOL 0x03f1
+#define RT5682_L_EQ_POST_VOL 0x03f2
+#define RT5682_R_EQ_POST_VOL 0x03f3
+#define RT5682_I2C_MODE 0xffff
+
+
+/* global definition */
+#define RT5682_L_MUTE (0x1 << 15)
+#define RT5682_L_MUTE_SFT 15
+#define RT5682_VOL_L_MUTE (0x1 << 14)
+#define RT5682_VOL_L_SFT 14
+#define RT5682_R_MUTE (0x1 << 7)
+#define RT5682_R_MUTE_SFT 7
+#define RT5682_VOL_R_MUTE (0x1 << 6)
+#define RT5682_VOL_R_SFT 6
+#define RT5682_L_VOL_MASK (0x3f << 8)
+#define RT5682_L_VOL_SFT 8
+#define RT5682_R_VOL_MASK (0x3f)
+#define RT5682_R_VOL_SFT 0
+
+/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/
+#define RT5682_G_HP (0xf << 8)
+#define RT5682_G_HP_SFT 8
+#define RT5682_G_STO_DA_DMIX (0xf)
+#define RT5682_G_STO_DA_SFT 0
+
+/* CBJ Control (0x000b) */
+#define RT5682_BST_CBJ_MASK (0xf << 8)
+#define RT5682_BST_CBJ_SFT 8
+
+/* Embeeded Jack and Type Detection Control 1 (0x0010) */
+#define RT5682_EMB_JD_EN (0x1 << 15)
+#define RT5682_EMB_JD_EN_SFT 15
+#define RT5682_EMB_JD_RST (0x1 << 14)
+#define RT5682_JD_MODE (0x1 << 13)
+#define RT5682_JD_MODE_SFT 13
+#define RT5682_DET_TYPE (0x1 << 12)
+#define RT5682_DET_TYPE_SFT 12
+#define RT5682_POLA_EXT_JD_MASK (0x1 << 11)
+#define RT5682_POLA_EXT_JD_LOW (0x1 << 11)
+#define RT5682_POLA_EXT_JD_HIGH (0x0 << 11)
+#define RT5682_EXT_JD_DIG (0x1 << 9)
+#define RT5682_POL_FAST_OFF_MASK (0x1 << 8)
+#define RT5682_POL_FAST_OFF_HIGH (0x1 << 8)
+#define RT5682_POL_FAST_OFF_LOW (0x0 << 8)
+#define RT5682_FAST_OFF_MASK (0x1 << 7)
+#define RT5682_FAST_OFF_EN (0x1 << 7)
+#define RT5682_FAST_OFF_DIS (0x0 << 7)
+#define RT5682_VREF_POW_MASK (0x1 << 6)
+#define RT5682_VREF_POW_FSM (0x0 << 6)
+#define RT5682_VREF_POW_REG (0x1 << 6)
+#define RT5682_MB1_PATH_MASK (0x1 << 5)
+#define RT5682_CTRL_MB1_REG (0x1 << 5)
+#define RT5682_CTRL_MB1_FSM (0x0 << 5)
+#define RT5682_MB2_PATH_MASK (0x1 << 4)
+#define RT5682_CTRL_MB2_REG (0x1 << 4)
+#define RT5682_CTRL_MB2_FSM (0x0 << 4)
+#define RT5682_TRIG_JD_MASK (0x1 << 3)
+#define RT5682_TRIG_JD_HIGH (0x1 << 3)
+#define RT5682_TRIG_JD_LOW (0x0 << 3)
+#define RT5682_MIC_CAP_MASK (0x1 << 1)
+#define RT5682_MIC_CAP_HS (0x1 << 1)
+#define RT5682_MIC_CAP_HP (0x0 << 1)
+#define RT5682_MIC_CAP_SRC_MASK (0x1)
+#define RT5682_MIC_CAP_SRC_REG (0x1)
+#define RT5682_MIC_CAP_SRC_ANA (0x0)
+
+/* Embeeded Jack and Type Detection Control 2 (0x0011) */
+#define RT5682_EXT_JD_SRC (0x7 << 4)
+#define RT5682_EXT_JD_SRC_SFT 4
+#define RT5682_EXT_JD_SRC_GPIO_JD1 (0x0 << 4)
+#define RT5682_EXT_JD_SRC_GPIO_JD2 (0x1 << 4)
+#define RT5682_EXT_JD_SRC_JDH (0x2 << 4)
+#define RT5682_EXT_JD_SRC_JDL (0x3 << 4)
+#define RT5682_EXT_JD_SRC_MANUAL (0x4 << 4)
+#define RT5682_JACK_TYPE_MASK (0x3)
+
+/* Combo Jack and Type Detection Control 3 (0x0012) */
+#define RT5682_CBJ_IN_BUF_EN (0x1 << 7)
+
+/* Combo Jack and Type Detection Control 4 (0x0013) */
+#define RT5682_SEL_SHT_MID_TON_MASK (0x3 << 12)
+#define RT5682_SEL_SHT_MID_TON_2 (0x0 << 12)
+#define RT5682_SEL_SHT_MID_TON_3 (0x1 << 12)
+#define RT5682_CBJ_JD_TEST_MASK (0x1 << 6)
+#define RT5682_CBJ_JD_TEST_NORM (0x0 << 6)
+#define RT5682_CBJ_JD_TEST_MODE (0x1 << 6)
+
+/* DAC1 Digital Volume (0x0019) */
+#define RT5682_DAC_L1_VOL_MASK (0xff << 8)
+#define RT5682_DAC_L1_VOL_SFT 8
+#define RT5682_DAC_R1_VOL_MASK (0xff)
+#define RT5682_DAC_R1_VOL_SFT 0
+
+/* ADC Digital Volume Control (0x001c) */
+#define RT5682_ADC_L_VOL_MASK (0x7f << 8)
+#define RT5682_ADC_L_VOL_SFT 8
+#define RT5682_ADC_R_VOL_MASK (0x7f)
+#define RT5682_ADC_R_VOL_SFT 0
+
+/* Stereo1 ADC Boost Gain Control (0x001f) */
+#define RT5682_STO1_ADC_L_BST_MASK (0x3 << 14)
+#define RT5682_STO1_ADC_L_BST_SFT 14
+#define RT5682_STO1_ADC_R_BST_MASK (0x3 << 12)
+#define RT5682_STO1_ADC_R_BST_SFT 12
+
+/* Sidetone Control (0x0024) */
+#define RT5682_ST_SRC_SEL (0x1 << 8)
+#define RT5682_ST_SRC_SFT 8
+#define RT5682_ST_EN_MASK (0x1 << 6)
+#define RT5682_ST_DIS (0x0 << 6)
+#define RT5682_ST_EN (0x1 << 6)
+#define RT5682_ST_EN_SFT 6
+
+/* Stereo1 ADC Mixer Control (0x0026) */
+#define RT5682_M_STO1_ADC_L1 (0x1 << 15)
+#define RT5682_M_STO1_ADC_L1_SFT 15
+#define RT5682_M_STO1_ADC_L2 (0x1 << 14)
+#define RT5682_M_STO1_ADC_L2_SFT 14
+#define RT5682_STO1_ADC1L_SRC_MASK (0x1 << 13)
+#define RT5682_STO1_ADC1L_SRC_SFT 13
+#define RT5682_STO1_ADC1_SRC_ADC (0x1 << 13)
+#define RT5682_STO1_ADC1_SRC_DACMIX (0x0 << 13)
+#define RT5682_STO1_ADC2L_SRC_MASK (0x1 << 12)
+#define RT5682_STO1_ADC2L_SRC_SFT 12
+#define RT5682_STO1_ADCL_SRC_MASK (0x3 << 10)
+#define RT5682_STO1_ADCL_SRC_SFT 10
+#define RT5682_STO1_DD_L_SRC_MASK (0x1 << 9)
+#define RT5682_STO1_DD_L_SRC_SFT 9
+#define RT5682_STO1_DMIC_SRC_MASK (0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_SFT 8
+#define RT5682_STO1_DMIC_SRC_DMIC2 (0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_DMIC1 (0x0 << 8)
+#define RT5682_M_STO1_ADC_R1 (0x1 << 7)
+#define RT5682_M_STO1_ADC_R1_SFT 7
+#define RT5682_M_STO1_ADC_R2 (0x1 << 6)
+#define RT5682_M_STO1_ADC_R2_SFT 6
+#define RT5682_STO1_ADC1R_SRC_MASK (0x1 << 5)
+#define RT5682_STO1_ADC1R_SRC_SFT 5
+#define RT5682_STO1_ADC2R_SRC_MASK (0x1 << 4)
+#define RT5682_STO1_ADC2R_SRC_SFT 4
+#define RT5682_STO1_ADCR_SRC_MASK (0x3 << 2)
+#define RT5682_STO1_ADCR_SRC_SFT 2
+
+/* ADC Mixer to DAC Mixer Control (0x0029) */
+#define RT5682_M_ADCMIX_L (0x1 << 15)
+#define RT5682_M_ADCMIX_L_SFT 15
+#define RT5682_M_DAC1_L (0x1 << 14)
+#define RT5682_M_DAC1_L_SFT 14
+#define RT5682_DAC1_R_SEL_MASK (0x1 << 10)
+#define RT5682_DAC1_R_SEL_SFT 10
+#define RT5682_DAC1_L_SEL_MASK (0x1 << 8)
+#define RT5682_DAC1_L_SEL_SFT 8
+#define RT5682_M_ADCMIX_R (0x1 << 7)
+#define RT5682_M_ADCMIX_R_SFT 7
+#define RT5682_M_DAC1_R (0x1 << 6)
+#define RT5682_M_DAC1_R_SFT 6
+
+/* Stereo1 DAC Mixer Control (0x002a) */
+#define RT5682_M_DAC_L1_STO_L (0x1 << 15)
+#define RT5682_M_DAC_L1_STO_L_SFT 15
+#define RT5682_G_DAC_L1_STO_L_MASK (0x1 << 14)
+#define RT5682_G_DAC_L1_STO_L_SFT 14
+#define RT5682_M_DAC_R1_STO_L (0x1 << 13)
+#define RT5682_M_DAC_R1_STO_L_SFT 13
+#define RT5682_G_DAC_R1_STO_L_MASK (0x1 << 12)
+#define RT5682_G_DAC_R1_STO_L_SFT 12
+#define RT5682_M_DAC_L1_STO_R (0x1 << 7)
+#define RT5682_M_DAC_L1_STO_R_SFT 7
+#define RT5682_G_DAC_L1_STO_R_MASK (0x1 << 6)
+#define RT5682_G_DAC_L1_STO_R_SFT 6
+#define RT5682_M_DAC_R1_STO_R (0x1 << 5)
+#define RT5682_M_DAC_R1_STO_R_SFT 5
+#define RT5682_G_DAC_R1_STO_R_MASK (0x1 << 4)
+#define RT5682_G_DAC_R1_STO_R_SFT 4
+
+/* Analog DAC1 Input Source Control (0x002b) */
+#define RT5682_M_ST_STO_L (0x1 << 9)
+#define RT5682_M_ST_STO_L_SFT 9
+#define RT5682_M_ST_STO_R (0x1 << 8)
+#define RT5682_M_ST_STO_R_SFT 8
+#define RT5682_DAC_L1_SRC_MASK (0x3 << 4)
+#define RT5682_A_DACL1_SFT 4
+#define RT5682_DAC_R1_SRC_MASK (0x3)
+#define RT5682_A_DACR1_SFT 0
+
+/* Digital Interface Data Control (0x0030) */
+#define RT5682_IF2_ADC_SEL_MASK (0x3 << 0)
+#define RT5682_IF2_ADC_SEL_SFT 0
+
+/* REC Left Mixer Control 2 (0x003c) */
+#define RT5682_G_CBJ_RM1_L (0x7 << 10)
+#define RT5682_G_CBJ_RM1_L_SFT 10
+#define RT5682_M_CBJ_RM1_L (0x1 << 7)
+#define RT5682_M_CBJ_RM1_L_SFT 7
+
+/* Power Management for Digital 1 (0x0061) */
+#define RT5682_PWR_I2S1 (0x1 << 15)
+#define RT5682_PWR_I2S1_BIT 15
+#define RT5682_PWR_I2S2 (0x1 << 14)
+#define RT5682_PWR_I2S2_BIT 14
+#define RT5682_PWR_DAC_L1 (0x1 << 11)
+#define RT5682_PWR_DAC_L1_BIT 11
+#define RT5682_PWR_DAC_R1 (0x1 << 10)
+#define RT5682_PWR_DAC_R1_BIT 10
+#define RT5682_PWR_LDO (0x1 << 8)
+#define RT5682_PWR_LDO_BIT 8
+#define RT5682_PWR_ADC_L1 (0x1 << 4)
+#define RT5682_PWR_ADC_L1_BIT 4
+#define RT5682_PWR_ADC_R1 (0x1 << 3)
+#define RT5682_PWR_ADC_R1_BIT 3
+#define RT5682_DIG_GATE_CTRL (0x1 << 0)
+#define RT5682_DIG_GATE_CTRL_SFT 0
+
+
+/* Power Management for Digital 2 (0x0062) */
+#define RT5682_PWR_ADC_S1F (0x1 << 15)
+#define RT5682_PWR_ADC_S1F_BIT 15
+#define RT5682_PWR_DAC_S1F (0x1 << 10)
+#define RT5682_PWR_DAC_S1F_BIT 10
+
+/* Power Management for Analog 1 (0x0063) */
+#define RT5682_PWR_VREF1 (0x1 << 15)
+#define RT5682_PWR_VREF1_BIT 15
+#define RT5682_PWR_FV1 (0x1 << 14)
+#define RT5682_PWR_FV1_BIT 14
+#define RT5682_PWR_VREF2 (0x1 << 13)
+#define RT5682_PWR_VREF2_BIT 13
+#define RT5682_PWR_FV2 (0x1 << 12)
+#define RT5682_PWR_FV2_BIT 12
+#define RT5682_LDO1_DBG_MASK (0x3 << 10)
+#define RT5682_PWR_MB (0x1 << 9)
+#define RT5682_PWR_MB_BIT 9
+#define RT5682_PWR_BG (0x1 << 7)
+#define RT5682_PWR_BG_BIT 7
+#define RT5682_LDO1_BYPASS_MASK (0x1 << 6)
+#define RT5682_LDO1_BYPASS (0x1 << 6)
+#define RT5682_LDO1_NOT_BYPASS (0x0 << 6)
+#define RT5682_PWR_MA_BIT 6
+#define RT5682_LDO1_DVO_MASK (0x3 << 4)
+#define RT5682_LDO1_DVO_09 (0x0 << 4)
+#define RT5682_LDO1_DVO_10 (0x1 << 4)
+#define RT5682_LDO1_DVO_12 (0x2 << 4)
+#define RT5682_LDO1_DVO_14 (0x3 << 4)
+#define RT5682_HP_DRIVER_MASK (0x3 << 2)
+#define RT5682_HP_DRIVER_1X (0x0 << 2)
+#define RT5682_HP_DRIVER_3X (0x1 << 2)
+#define RT5682_HP_DRIVER_5X (0x3 << 2)
+#define RT5682_PWR_HA_L (0x1 << 1)
+#define RT5682_PWR_HA_L_BIT 1
+#define RT5682_PWR_HA_R (0x1 << 0)
+#define RT5682_PWR_HA_R_BIT 0
+
+/* Power Management for Analog 2 (0x0064) */
+#define RT5682_PWR_MB1 (0x1 << 11)
+#define RT5682_PWR_MB1_PWR_DOWN (0x0 << 11)
+#define RT5682_PWR_MB1_BIT 11
+#define RT5682_PWR_MB2 (0x1 << 10)
+#define RT5682_PWR_MB2_PWR_DOWN (0x0 << 10)
+#define RT5682_PWR_MB2_BIT 10
+#define RT5682_PWR_JDH (0x1 << 3)
+#define RT5682_PWR_JDH_BIT 3
+#define RT5682_PWR_JDL (0x1 << 2)
+#define RT5682_PWR_JDL_BIT 2
+#define RT5682_PWR_RM1_L (0x1 << 1)
+#define RT5682_PWR_RM1_L_BIT 1
+
+/* Power Management for Analog 3 (0x0065) */
+#define RT5682_PWR_CBJ (0x1 << 9)
+#define RT5682_PWR_CBJ_BIT 9
+#define RT5682_PWR_PLL (0x1 << 6)
+#define RT5682_PWR_PLL_BIT 6
+#define RT5682_PWR_PLL2B (0x1 << 5)
+#define RT5682_PWR_PLL2B_BIT 5
+#define RT5682_PWR_PLL2F (0x1 << 4)
+#define RT5682_PWR_PLL2F_BIT 4
+#define RT5682_PWR_LDO2 (0x1 << 2)
+#define RT5682_PWR_LDO2_BIT 2
+#define RT5682_PWR_DET_SPKVDD (0x1 << 1)
+#define RT5682_PWR_DET_SPKVDD_BIT 1
+
+/* Power Management for Mixer (0x0066) */
+#define RT5682_PWR_STO1_DAC_L (0x1 << 5)
+#define RT5682_PWR_STO1_DAC_L_BIT 5
+#define RT5682_PWR_STO1_DAC_R (0x1 << 4)
+#define RT5682_PWR_STO1_DAC_R_BIT 4
+
+/* MCLK and System Clock Detection Control (0x006b) */
+#define RT5682_SYS_CLK_DET (0x1 << 15)
+#define RT5682_SYS_CLK_DET_SFT 15
+#define RT5682_PLL1_CLK_DET (0x1 << 14)
+#define RT5682_PLL1_CLK_DET_SFT 14
+#define RT5682_PLL2_CLK_DET (0x1 << 13)
+#define RT5682_PLL2_CLK_DET_SFT 13
+#define RT5682_POW_CLK_DET2_SFT 8
+#define RT5682_POW_CLK_DET_SFT 0
+
+/* Digital Microphone Control 1 (0x006e) */
+#define RT5682_DMIC_1_EN_MASK (0x1 << 15)
+#define RT5682_DMIC_1_EN_SFT 15
+#define RT5682_DMIC_1_DIS (0x0 << 15)
+#define RT5682_DMIC_1_EN (0x1 << 15)
+#define RT5682_DMIC_1_DP_MASK (0x3 << 4)
+#define RT5682_DMIC_1_DP_SFT 4
+#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4)
+#define RT5682_DMIC_1_DP_GPIO5 (0x1 << 4)
+#define RT5682_DMIC_CLK_MASK (0xf << 0)
+#define RT5682_DMIC_CLK_SFT 0
+
+/* I2S1 Audio Serial Data Port Control (0x0070) */
+#define RT5682_SEL_ADCDAT_MASK (0x1 << 15)
+#define RT5682_SEL_ADCDAT_OUT (0x0 << 15)
+#define RT5682_SEL_ADCDAT_IN (0x1 << 15)
+#define RT5682_SEL_ADCDAT_SFT 15
+#define RT5682_I2S1_TX_CHL_MASK (0x7 << 12)
+#define RT5682_I2S1_TX_CHL_SFT 12
+#define RT5682_I2S1_TX_CHL_16 (0x0 << 12)
+#define RT5682_I2S1_TX_CHL_20 (0x1 << 12)
+#define RT5682_I2S1_TX_CHL_24 (0x2 << 12)
+#define RT5682_I2S1_TX_CHL_32 (0x3 << 12)
+#define RT5682_I2S1_TX_CHL_8 (0x4 << 12)
+#define RT5682_I2S1_RX_CHL_MASK (0x7 << 8)
+#define RT5682_I2S1_RX_CHL_SFT 8
+#define RT5682_I2S1_RX_CHL_16 (0x0 << 8)
+#define RT5682_I2S1_RX_CHL_20 (0x1 << 8)
+#define RT5682_I2S1_RX_CHL_24 (0x2 << 8)
+#define RT5682_I2S1_RX_CHL_32 (0x3 << 8)
+#define RT5682_I2S1_RX_CHL_8 (0x4 << 8)
+#define RT5682_I2S1_MONO_MASK (0x1 << 7)
+#define RT5682_I2S1_MONO_EN (0x1 << 7)
+#define RT5682_I2S1_MONO_DIS (0x0 << 7)
+#define RT5682_I2S2_MONO_MASK (0x1 << 6)
+#define RT5682_I2S2_MONO_EN (0x1 << 6)
+#define RT5682_I2S2_MONO_DIS (0x0 << 6)
+#define RT5682_I2S1_DL_MASK (0x7 << 4)
+#define RT5682_I2S1_DL_SFT 4
+#define RT5682_I2S1_DL_16 (0x0 << 4)
+#define RT5682_I2S1_DL_20 (0x1 << 4)
+#define RT5682_I2S1_DL_24 (0x2 << 4)
+#define RT5682_I2S1_DL_32 (0x3 << 4)
+#define RT5682_I2S1_DL_8 (0x4 << 4)
+
+/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */
+#define RT5682_I2S2_MS_MASK (0x1 << 15)
+#define RT5682_I2S2_MS_SFT 15
+#define RT5682_I2S2_MS_M (0x0 << 15)
+#define RT5682_I2S2_MS_S (0x1 << 15)
+#define RT5682_I2S2_PIN_CFG_MASK (0x1 << 14)
+#define RT5682_I2S2_PIN_CFG_SFT 14
+#define RT5682_I2S2_CLK_SEL_MASK (0x1 << 11)
+#define RT5682_I2S2_CLK_SEL_SFT 11
+#define RT5682_I2S2_OUT_MASK (0x1 << 9)
+#define RT5682_I2S2_OUT_SFT 9
+#define RT5682_I2S2_OUT_UM (0x0 << 9)
+#define RT5682_I2S2_OUT_M (0x1 << 9)
+#define RT5682_I2S_BP_MASK (0x1 << 8)
+#define RT5682_I2S_BP_SFT 8
+#define RT5682_I2S_BP_NOR (0x0 << 8)
+#define RT5682_I2S_BP_INV (0x1 << 8)
+#define RT5682_I2S2_MONO_EN (0x1 << 6)
+#define RT5682_I2S2_MONO_DIS (0x0 << 6)
+#define RT5682_I2S2_DL_MASK (0x3 << 4)
+#define RT5682_I2S2_DL_SFT 4
+#define RT5682_I2S2_DL_16 (0x0 << 4)
+#define RT5682_I2S2_DL_20 (0x1 << 4)
+#define RT5682_I2S2_DL_24 (0x2 << 4)
+#define RT5682_I2S2_DL_8 (0x3 << 4)
+#define RT5682_I2S_DF_MASK (0x7)
+#define RT5682_I2S_DF_SFT 0
+#define RT5682_I2S_DF_I2S (0x0)
+#define RT5682_I2S_DF_LEFT (0x1)
+#define RT5682_I2S_DF_PCM_A (0x2)
+#define RT5682_I2S_DF_PCM_B (0x3)
+#define RT5682_I2S_DF_PCM_A_N (0x6)
+#define RT5682_I2S_DF_PCM_B_N (0x7)
+
+/* ADC/DAC Clock Control 1 (0x0073) */
+#define RT5682_ADC_OSR_MASK (0xf << 12)
+#define RT5682_ADC_OSR_SFT 12
+#define RT5682_ADC_OSR_D_1 (0x0 << 12)
+#define RT5682_ADC_OSR_D_2 (0x1 << 12)
+#define RT5682_ADC_OSR_D_4 (0x2 << 12)
+#define RT5682_ADC_OSR_D_6 (0x3 << 12)
+#define RT5682_ADC_OSR_D_8 (0x4 << 12)
+#define RT5682_ADC_OSR_D_12 (0x5 << 12)
+#define RT5682_ADC_OSR_D_16 (0x6 << 12)
+#define RT5682_ADC_OSR_D_24 (0x7 << 12)
+#define RT5682_ADC_OSR_D_32 (0x8 << 12)
+#define RT5682_ADC_OSR_D_48 (0x9 << 12)
+#define RT5682_I2S_M_DIV_MASK (0xf << 12)
+#define RT5682_I2S_M_DIV_SFT 8
+#define RT5682_I2S_M_D_1 (0x0 << 8)
+#define RT5682_I2S_M_D_2 (0x1 << 8)
+#define RT5682_I2S_M_D_3 (0x2 << 8)
+#define RT5682_I2S_M_D_4 (0x3 << 8)
+#define RT5682_I2S_M_D_6 (0x4 << 8)
+#define RT5682_I2S_M_D_8 (0x5 << 8)
+#define RT5682_I2S_M_D_12 (0x6 << 8)
+#define RT5682_I2S_M_D_16 (0x7 << 8)
+#define RT5682_I2S_M_D_24 (0x8 << 8)
+#define RT5682_I2S_M_D_32 (0x9 << 8)
+#define RT5682_I2S_M_D_48 (0x10 << 8)
+#define RT5682_I2S_CLK_SRC_MASK (0x7 << 4)
+#define RT5682_I2S_CLK_SRC_SFT 4
+#define RT5682_I2S_CLK_SRC_MCLK (0x0 << 4)
+#define RT5682_I2S_CLK_SRC_PLL1 (0x1 << 4)
+#define RT5682_I2S_CLK_SRC_PLL2 (0x2 << 4)
+#define RT5682_I2S_CLK_SRC_SDW (0x3 << 4)
+#define RT5682_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */
+#define RT5682_DAC_OSR_MASK (0xf << 0)
+#define RT5682_DAC_OSR_SFT 0
+#define RT5682_DAC_OSR_D_1 (0x0 << 0)
+#define RT5682_DAC_OSR_D_2 (0x1 << 0)
+#define RT5682_DAC_OSR_D_4 (0x2 << 0)
+#define RT5682_DAC_OSR_D_6 (0x3 << 0)
+#define RT5682_DAC_OSR_D_8 (0x4 << 0)
+#define RT5682_DAC_OSR_D_12 (0x5 << 0)
+#define RT5682_DAC_OSR_D_16 (0x6 << 0)
+#define RT5682_DAC_OSR_D_24 (0x7 << 0)
+#define RT5682_DAC_OSR_D_32 (0x8 << 0)
+#define RT5682_DAC_OSR_D_48 (0x9 << 0)
+
+/* ADC/DAC Clock Control 2 (0x0074) */
+#define RT5682_I2S2_BCLK_MS2_MASK (0x1 << 11)
+#define RT5682_I2S2_BCLK_MS2_SFT 11
+#define RT5682_I2S2_BCLK_MS2_32 (0x0 << 11)
+#define RT5682_I2S2_BCLK_MS2_64 (0x1 << 11)
+
+
+/* TDM control 1 (0x0079) */
+#define RT5682_TDM_TX_CH_MASK (0x3 << 12)
+#define RT5682_TDM_TX_CH_2 (0x0 << 12)
+#define RT5682_TDM_TX_CH_4 (0x1 << 12)
+#define RT5682_TDM_TX_CH_6 (0x2 << 12)
+#define RT5682_TDM_TX_CH_8 (0x3 << 12)
+#define RT5682_TDM_RX_CH_MASK (0x3 << 8)
+#define RT5682_TDM_RX_CH_2 (0x0 << 8)
+#define RT5682_TDM_RX_CH_4 (0x1 << 8)
+#define RT5682_TDM_RX_CH_6 (0x2 << 8)
+#define RT5682_TDM_RX_CH_8 (0x3 << 8)
+#define RT5682_TDM_ADC_LCA_MASK (0xf << 4)
+#define RT5682_TDM_ADC_LCA_SFT 4
+#define RT5682_TDM_ADC_DL_SFT 0
+
+/* TDM control 2 (0x007a) */
+#define RT5682_IF1_ADC1_SEL_SFT 14
+#define RT5682_IF1_ADC2_SEL_SFT 12
+#define RT5682_IF1_ADC3_SEL_SFT 10
+#define RT5682_IF1_ADC4_SEL_SFT 8
+#define RT5682_TDM_ADC_SEL_SFT 4
+
+/* TDM control 3 (0x007b) */
+#define RT5682_TDM_EN (0x1 << 7)
+
+/* TDM/I2S control (0x007e) */
+#define RT5682_TDM_S_BP_MASK (0x1 << 15)
+#define RT5682_TDM_S_BP_SFT 15
+#define RT5682_TDM_S_BP_NOR (0x0 << 15)
+#define RT5682_TDM_S_BP_INV (0x1 << 15)
+#define RT5682_TDM_S_LP_MASK (0x1 << 14)
+#define RT5682_TDM_S_LP_SFT 14
+#define RT5682_TDM_S_LP_NOR (0x0 << 14)
+#define RT5682_TDM_S_LP_INV (0x1 << 14)
+#define RT5682_TDM_DF_MASK (0x7 << 11)
+#define RT5682_TDM_DF_SFT 11
+#define RT5682_TDM_DF_I2S (0x0 << 11)
+#define RT5682_TDM_DF_LEFT (0x1 << 11)
+#define RT5682_TDM_DF_PCM_A (0x2 << 11)
+#define RT5682_TDM_DF_PCM_B (0x3 << 11)
+#define RT5682_TDM_DF_PCM_A_N (0x6 << 11)
+#define RT5682_TDM_DF_PCM_B_N (0x7 << 11)
+#define RT5682_TDM_CL_MASK (0x3 << 4)
+#define RT5682_TDM_CL_16 (0x0 << 4)
+#define RT5682_TDM_CL_20 (0x1 << 4)
+#define RT5682_TDM_CL_24 (0x2 << 4)
+#define RT5682_TDM_CL_32 (0x3 << 4)
+#define RT5682_TDM_M_BP_MASK (0x1 << 2)
+#define RT5682_TDM_M_BP_SFT 2
+#define RT5682_TDM_M_BP_NOR (0x0 << 2)
+#define RT5682_TDM_M_BP_INV (0x1 << 2)
+#define RT5682_TDM_M_LP_MASK (0x1 << 1)
+#define RT5682_TDM_M_LP_SFT 1
+#define RT5682_TDM_M_LP_NOR (0x0 << 1)
+#define RT5682_TDM_M_LP_INV (0x1 << 1)
+#define RT5682_TDM_MS_MASK (0x1 << 0)
+#define RT5682_TDM_MS_SFT 0
+#define RT5682_TDM_MS_M (0x0 << 0)
+#define RT5682_TDM_MS_S (0x1 << 0)
+
+/* Global Clock Control (0x0080) */
+#define RT5682_SCLK_SRC_MASK (0x7 << 13)
+#define RT5682_SCLK_SRC_SFT 13
+#define RT5682_SCLK_SRC_MCLK (0x0 << 13)
+#define RT5682_SCLK_SRC_PLL1 (0x1 << 13)
+#define RT5682_SCLK_SRC_PLL2 (0x2 << 13)
+#define RT5682_SCLK_SRC_SDW (0x3 << 13)
+#define RT5682_SCLK_SRC_RCCLK (0x4 << 13)
+#define RT5682_PLL1_SRC_MASK (0x3 << 10)
+#define RT5682_PLL1_SRC_SFT 10
+#define RT5682_PLL1_SRC_MCLK (0x0 << 10)
+#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10)
+#define RT5682_PLL1_SRC_SDW (0x2 << 10)
+#define RT5682_PLL1_SRC_RC (0x3 << 10)
+#define RT5682_PLL2_SRC_MASK (0x3 << 8)
+#define RT5682_PLL2_SRC_SFT 8
+#define RT5682_PLL2_SRC_MCLK (0x0 << 8)
+#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8)
+#define RT5682_PLL2_SRC_SDW (0x2 << 8)
+#define RT5682_PLL2_SRC_RC (0x3 << 8)
+
+
+
+#define RT5682_PLL_INP_MAX 40000000
+#define RT5682_PLL_INP_MIN 256000
+/* PLL M/N/K Code Control 1 (0x0081) */
+#define RT5682_PLL_N_MAX 0x001ff
+#define RT5682_PLL_N_MASK (RT5682_PLL_N_MAX << 7)
+#define RT5682_PLL_N_SFT 7
+#define RT5682_PLL_K_MAX 0x001f
+#define RT5682_PLL_K_MASK (RT5682_PLL_K_MAX)
+#define RT5682_PLL_K_SFT 0
+
+/* PLL M/N/K Code Control 2 (0x0082) */
+#define RT5682_PLL_M_MAX 0x00f
+#define RT5682_PLL_M_MASK (RT5682_PLL_M_MAX << 12)
+#define RT5682_PLL_M_SFT 12
+#define RT5682_PLL_M_BP (0x1 << 11)
+#define RT5682_PLL_M_BP_SFT 11
+#define RT5682_PLL_K_BP (0x1 << 10)
+#define RT5682_PLL_K_BP_SFT 10
+#define RT5682_PLL_RST (0x1 << 1)
+
+/* PLL tracking mode 1 (0x0083) */
+#define RT5682_DA_ASRC_MASK (0x1 << 13)
+#define RT5682_DA_ASRC_SFT 13
+#define RT5682_DAC_STO1_ASRC_MASK (0x1 << 12)
+#define RT5682_DAC_STO1_ASRC_SFT 12
+#define RT5682_AD_ASRC_MASK (0x1 << 8)
+#define RT5682_AD_ASRC_SFT 8
+#define RT5682_AD_ASRC_SEL_MASK (0x1 << 4)
+#define RT5682_AD_ASRC_SEL_SFT 4
+#define RT5682_DMIC_ASRC_MASK (0x1 << 3)
+#define RT5682_DMIC_ASRC_SFT 3
+#define RT5682_ADC_STO1_ASRC_MASK (0x1 << 2)
+#define RT5682_ADC_STO1_ASRC_SFT 2
+#define RT5682_DA_ASRC_SEL_MASK (0x1 << 0)
+#define RT5682_DA_ASRC_SEL_SFT 0
+
+/* PLL tracking mode 2 3 (0x0084)(0x0085)*/
+#define RT5682_FILTER_CLK_SEL_MASK (0x7 << 12)
+#define RT5682_FILTER_CLK_SEL_SFT 12
+#define RT5682_FILTER_CLK_DIV_MASK (0xf << 8)
+#define RT5682_FILTER_CLK_DIV_SFT 8
+
+/* ASRC Control 4 (0x0086) */
+#define RT5682_ASRCIN_FTK_N1_MASK (0x3 << 14)
+#define RT5682_ASRCIN_FTK_N1_SFT 14
+#define RT5682_ASRCIN_FTK_N2_MASK (0x3 << 12)
+#define RT5682_ASRCIN_FTK_N2_SFT 12
+#define RT5682_ASRCIN_FTK_M1_MASK (0x7 << 8)
+#define RT5682_ASRCIN_FTK_M1_SFT 8
+#define RT5682_ASRCIN_FTK_M2_MASK (0x7 << 4)
+#define RT5682_ASRCIN_FTK_M2_SFT 4
+
+/* SoundWire reference clk (0x008d) */
+#define RT5682_PLL2_OUT_MASK (0x1 << 8)
+#define RT5682_PLL2_OUT_98M (0x0 << 8)
+#define RT5682_PLL2_OUT_49M (0x1 << 8)
+#define RT5682_SDW_REF_2_MASK (0xf << 4)
+#define RT5682_SDW_REF_2_SFT 4
+#define RT5682_SDW_REF_2_48K (0x0 << 4)
+#define RT5682_SDW_REF_2_96K (0x1 << 4)
+#define RT5682_SDW_REF_2_192K (0x2 << 4)
+#define RT5682_SDW_REF_2_32K (0x3 << 4)
+#define RT5682_SDW_REF_2_24K (0x4 << 4)
+#define RT5682_SDW_REF_2_16K (0x5 << 4)
+#define RT5682_SDW_REF_2_12K (0x6 << 4)
+#define RT5682_SDW_REF_2_8K (0x7 << 4)
+#define RT5682_SDW_REF_2_44K (0x8 << 4)
+#define RT5682_SDW_REF_2_88K (0x9 << 4)
+#define RT5682_SDW_REF_2_176K (0xa << 4)
+#define RT5682_SDW_REF_2_353K (0xb << 4)
+#define RT5682_SDW_REF_2_22K (0xc << 4)
+#define RT5682_SDW_REF_2_384K (0xd << 4)
+#define RT5682_SDW_REF_2_11K (0xe << 4)
+#define RT5682_SDW_REF_1_MASK (0xf << 0)
+#define RT5682_SDW_REF_1_SFT 0
+#define RT5682_SDW_REF_1_48K (0x0 << 0)
+#define RT5682_SDW_REF_1_96K (0x1 << 0)
+#define RT5682_SDW_REF_1_192K (0x2 << 0)
+#define RT5682_SDW_REF_1_32K (0x3 << 0)
+#define RT5682_SDW_REF_1_24K (0x4 << 0)
+#define RT5682_SDW_REF_1_16K (0x5 << 0)
+#define RT5682_SDW_REF_1_12K (0x6 << 0)
+#define RT5682_SDW_REF_1_8K (0x7 << 0)
+#define RT5682_SDW_REF_1_44K (0x8 << 0)
+#define RT5682_SDW_REF_1_88K (0x9 << 0)
+#define RT5682_SDW_REF_1_176K (0xa << 0)
+#define RT5682_SDW_REF_1_353K (0xb << 0)
+#define RT5682_SDW_REF_1_22K (0xc << 0)
+#define RT5682_SDW_REF_1_384K (0xd << 0)
+#define RT5682_SDW_REF_1_11K (0xe << 0)
+
+/* Depop Mode Control 1 (0x008e) */
+#define RT5682_PUMP_EN (0x1 << 3)
+#define RT5682_PUMP_EN_SFT 3
+#define RT5682_CAPLESS_EN (0x1 << 0)
+#define RT5682_CAPLESS_EN_SFT 0
+
+/* Depop Mode Control 2 (0x8f) */
+#define RT5682_RAMP_MASK (0x1 << 12)
+#define RT5682_RAMP_SFT 12
+#define RT5682_RAMP_DIS (0x0 << 12)
+#define RT5682_RAMP_EN (0x1 << 12)
+#define RT5682_BPS_MASK (0x1 << 11)
+#define RT5682_BPS_SFT 11
+#define RT5682_BPS_DIS (0x0 << 11)
+#define RT5682_BPS_EN (0x1 << 11)
+#define RT5682_FAST_UPDN_MASK (0x1 << 10)
+#define RT5682_FAST_UPDN_SFT 10
+#define RT5682_FAST_UPDN_DIS (0x0 << 10)
+#define RT5682_FAST_UPDN_EN (0x1 << 10)
+#define RT5682_VLO_MASK (0x1 << 7)
+#define RT5682_VLO_SFT 7
+#define RT5682_VLO_3V (0x0 << 7)
+#define RT5682_VLO_33V (0x1 << 7)
+
+/* HPOUT charge pump 1 (0x0091) */
+#define RT5682_OSW_L_MASK (0x1 << 11)
+#define RT5682_OSW_L_SFT 11
+#define RT5682_OSW_L_DIS (0x0 << 11)
+#define RT5682_OSW_L_EN (0x1 << 11)
+#define RT5682_OSW_R_MASK (0x1 << 10)
+#define RT5682_OSW_R_SFT 10
+#define RT5682_OSW_R_DIS (0x0 << 10)
+#define RT5682_OSW_R_EN (0x1 << 10)
+#define RT5682_PM_HP_MASK (0x3 << 8)
+#define RT5682_PM_HP_SFT 8
+#define RT5682_PM_HP_LV (0x0 << 8)
+#define RT5682_PM_HP_MV (0x1 << 8)
+#define RT5682_PM_HP_HV (0x2 << 8)
+#define RT5682_IB_HP_MASK (0x3 << 6)
+#define RT5682_IB_HP_SFT 6
+#define RT5682_IB_HP_125IL (0x0 << 6)
+#define RT5682_IB_HP_25IL (0x1 << 6)
+#define RT5682_IB_HP_5IL (0x2 << 6)
+#define RT5682_IB_HP_1IL (0x3 << 6)
+
+/* Micbias Control1 (0x93) */
+#define RT5682_MIC1_OV_MASK (0x3 << 14)
+#define RT5682_MIC1_OV_SFT 14
+#define RT5682_MIC1_OV_2V7 (0x0 << 14)
+#define RT5682_MIC1_OV_2V4 (0x1 << 14)
+#define RT5682_MIC1_OV_2V25 (0x3 << 14)
+#define RT5682_MIC1_OV_1V8 (0x4 << 14)
+#define RT5682_MIC1_CLK_MASK (0x1 << 13)
+#define RT5682_MIC1_CLK_SFT 13
+#define RT5682_MIC1_CLK_DIS (0x0 << 13)
+#define RT5682_MIC1_CLK_EN (0x1 << 13)
+#define RT5682_MIC1_OVCD_MASK (0x1 << 12)
+#define RT5682_MIC1_OVCD_SFT 12
+#define RT5682_MIC1_OVCD_DIS (0x0 << 12)
+#define RT5682_MIC1_OVCD_EN (0x1 << 12)
+#define RT5682_MIC1_OVTH_MASK (0x3 << 10)
+#define RT5682_MIC1_OVTH_SFT 10
+#define RT5682_MIC1_OVTH_768UA (0x0 << 10)
+#define RT5682_MIC1_OVTH_960UA (0x1 << 10)
+#define RT5682_MIC1_OVTH_1152UA (0x2 << 10)
+#define RT5682_MIC1_OVTH_1960UA (0x3 << 10)
+#define RT5682_MIC2_OV_MASK (0x3 << 8)
+#define RT5682_MIC2_OV_SFT 8
+#define RT5682_MIC2_OV_2V7 (0x0 << 8)
+#define RT5682_MIC2_OV_2V4 (0x1 << 8)
+#define RT5682_MIC2_OV_2V25 (0x3 << 8)
+#define RT5682_MIC2_OV_1V8 (0x4 << 8)
+#define RT5682_MIC2_CLK_MASK (0x1 << 7)
+#define RT5682_MIC2_CLK_SFT 7
+#define RT5682_MIC2_CLK_DIS (0x0 << 7)
+#define RT5682_MIC2_CLK_EN (0x1 << 7)
+#define RT5682_MIC2_OVTH_MASK (0x3 << 4)
+#define RT5682_MIC2_OVTH_SFT 4
+#define RT5682_MIC2_OVTH_768UA (0x0 << 4)
+#define RT5682_MIC2_OVTH_960UA (0x1 << 4)
+#define RT5682_MIC2_OVTH_1152UA (0x2 << 4)
+#define RT5682_MIC2_OVTH_1960UA (0x3 << 4)
+#define RT5682_PWR_MB_MASK (0x1 << 3)
+#define RT5682_PWR_MB_SFT 3
+#define RT5682_PWR_MB_PD (0x0 << 3)
+#define RT5682_PWR_MB_PU (0x1 << 3)
+
+/* Micbias Control2 (0x0094) */
+#define RT5682_PWR_CLK25M_MASK (0x1 << 9)
+#define RT5682_PWR_CLK25M_SFT 9
+#define RT5682_PWR_CLK25M_PD (0x0 << 9)
+#define RT5682_PWR_CLK25M_PU (0x1 << 9)
+#define RT5682_PWR_CLK1M_MASK (0x1 << 8)
+#define RT5682_PWR_CLK1M_SFT 8
+#define RT5682_PWR_CLK1M_PD (0x0 << 8)
+#define RT5682_PWR_CLK1M_PU (0x1 << 8)
+
+/* RC Clock Control (0x009f) */
+#define RT5682_POW_IRQ (0x1 << 15)
+#define RT5682_POW_JDH (0x1 << 14)
+#define RT5682_POW_JDL (0x1 << 13)
+#define RT5682_POW_ANA (0x1 << 12)
+
+/* I2S Master Mode Clock Control 1 (0x00a0) */
+#define RT5682_CLK_SRC_MCLK (0x0)
+#define RT5682_CLK_SRC_PLL1 (0x1)
+#define RT5682_CLK_SRC_PLL2 (0x2)
+#define RT5682_CLK_SRC_SDW (0x3)
+#define RT5682_CLK_SRC_RCCLK (0x4)
+#define RT5682_I2S_PD_1 (0x0)
+#define RT5682_I2S_PD_2 (0x1)
+#define RT5682_I2S_PD_3 (0x2)
+#define RT5682_I2S_PD_4 (0x3)
+#define RT5682_I2S_PD_6 (0x4)
+#define RT5682_I2S_PD_8 (0x5)
+#define RT5682_I2S_PD_12 (0x6)
+#define RT5682_I2S_PD_16 (0x7)
+#define RT5682_I2S_PD_24 (0x8)
+#define RT5682_I2S_PD_32 (0x9)
+#define RT5682_I2S_PD_48 (0xa)
+#define RT5682_I2S2_SRC_MASK (0x3 << 4)
+#define RT5682_I2S2_SRC_SFT 4
+#define RT5682_I2S2_M_PD_MASK (0xf << 0)
+#define RT5682_I2S2_M_PD_SFT 0
+
+/* IRQ Control 1 (0x00b6) */
+#define RT5682_JD1_PULSE_EN_MASK (0x1 << 10)
+#define RT5682_JD1_PULSE_EN_SFT 10
+#define RT5682_JD1_PULSE_DIS (0x0 << 10)
+#define RT5682_JD1_PULSE_EN (0x1 << 10)
+
+/* IRQ Control 2 (0x00b7) */
+#define RT5682_JD1_EN_MASK (0x1 << 15)
+#define RT5682_JD1_EN_SFT 15
+#define RT5682_JD1_DIS (0x0 << 15)
+#define RT5682_JD1_EN (0x1 << 15)
+#define RT5682_JD1_POL_MASK (0x1 << 13)
+#define RT5682_JD1_POL_NOR (0x0 << 13)
+#define RT5682_JD1_POL_INV (0x1 << 13)
+
+/* IRQ Control 3 (0x00b8) */
+#define RT5682_IL_IRQ_MASK (0x1 << 7)
+#define RT5682_IL_IRQ_DIS (0x0 << 7)
+#define RT5682_IL_IRQ_EN (0x1 << 7)
+
+/* GPIO Control 1 (0x00c0) */
+#define RT5682_GP1_PIN_MASK (0x3 << 14)
+#define RT5682_GP1_PIN_SFT 14
+#define RT5682_GP1_PIN_GPIO1 (0x0 << 14)
+#define RT5682_GP1_PIN_IRQ (0x1 << 14)
+#define RT5682_GP1_PIN_DMIC_CLK (0x2 << 14)
+#define RT5682_GP2_PIN_MASK (0x3 << 12)
+#define RT5682_GP2_PIN_SFT 12
+#define RT5682_GP2_PIN_GPIO2 (0x0 << 12)
+#define RT5682_GP2_PIN_LRCK2 (0x1 << 12)
+#define RT5682_GP2_PIN_DMIC_SDA (0x2 << 12)
+#define RT5682_GP3_PIN_MASK (0x3 << 10)
+#define RT5682_GP3_PIN_SFT 10
+#define RT5682_GP3_PIN_GPIO3 (0x0 << 10)
+#define RT5682_GP3_PIN_BCLK2 (0x1 << 10)
+#define RT5682_GP3_PIN_DMIC_CLK (0x2 << 10)
+#define RT5682_GP4_PIN_MASK (0x3 << 8)
+#define RT5682_GP4_PIN_SFT 8
+#define RT5682_GP4_PIN_GPIO4 (0x0 << 8)
+#define RT5682_GP4_PIN_ADCDAT1 (0x1 << 8)
+#define RT5682_GP4_PIN_DMIC_CLK (0x2 << 8)
+#define RT5682_GP4_PIN_ADCDAT2 (0x3 << 8)
+#define RT5682_GP5_PIN_MASK (0x3 << 6)
+#define RT5682_GP5_PIN_SFT 6
+#define RT5682_GP5_PIN_GPIO5 (0x0 << 6)
+#define RT5682_GP5_PIN_DACDAT1 (0x1 << 6)
+#define RT5682_GP5_PIN_DMIC_SDA (0x2 << 6)
+#define RT5682_GP6_PIN_MASK (0x1 << 5)
+#define RT5682_GP6_PIN_SFT 5
+#define RT5682_GP6_PIN_GPIO6 (0x0 << 5)
+#define RT5682_GP6_PIN_LRCK1 (0x1 << 5)
+
+/* GPIO Control 2 (0x00c1)*/
+#define RT5682_GP1_PF_MASK (0x1 << 15)
+#define RT5682_GP1_PF_IN (0x0 << 15)
+#define RT5682_GP1_PF_OUT (0x1 << 15)
+#define RT5682_GP1_OUT_MASK (0x1 << 14)
+#define RT5682_GP1_OUT_L (0x0 << 14)
+#define RT5682_GP1_OUT_H (0x1 << 14)
+#define RT5682_GP2_PF_MASK (0x1 << 13)
+#define RT5682_GP2_PF_IN (0x0 << 13)
+#define RT5682_GP2_PF_OUT (0x1 << 13)
+#define RT5682_GP2_OUT_MASK (0x1 << 12)
+#define RT5682_GP2_OUT_L (0x0 << 12)
+#define RT5682_GP2_OUT_H (0x1 << 12)
+#define RT5682_GP3_PF_MASK (0x1 << 11)
+#define RT5682_GP3_PF_IN (0x0 << 11)
+#define RT5682_GP3_PF_OUT (0x1 << 11)
+#define RT5682_GP3_OUT_MASK (0x1 << 10)
+#define RT5682_GP3_OUT_L (0x0 << 10)
+#define RT5682_GP3_OUT_H (0x1 << 10)
+#define RT5682_GP4_PF_MASK (0x1 << 9)
+#define RT5682_GP4_PF_IN (0x0 << 9)
+#define RT5682_GP4_PF_OUT (0x1 << 9)
+#define RT5682_GP4_OUT_MASK (0x1 << 8)
+#define RT5682_GP4_OUT_L (0x0 << 8)
+#define RT5682_GP4_OUT_H (0x1 << 8)
+#define RT5682_GP5_PF_MASK (0x1 << 7)
+#define RT5682_GP5_PF_IN (0x0 << 7)
+#define RT5682_GP5_PF_OUT (0x1 << 7)
+#define RT5682_GP5_OUT_MASK (0x1 << 6)
+#define RT5682_GP5_OUT_L (0x0 << 6)
+#define RT5682_GP5_OUT_H (0x1 << 6)
+#define RT5682_GP6_PF_MASK (0x1 << 5)
+#define RT5682_GP6_PF_IN (0x0 << 5)
+#define RT5682_GP6_PF_OUT (0x1 << 5)
+#define RT5682_GP6_OUT_MASK (0x1 << 4)
+#define RT5682_GP6_OUT_L (0x0 << 4)
+#define RT5682_GP6_OUT_H (0x1 << 4)
+
+
+/* GPIO Status (0x00c2) */
+#define RT5682_GP6_STA (0x1 << 6)
+#define RT5682_GP5_STA (0x1 << 5)
+#define RT5682_GP4_STA (0x1 << 4)
+#define RT5682_GP3_STA (0x1 << 3)
+#define RT5682_GP2_STA (0x1 << 2)
+#define RT5682_GP1_STA (0x1 << 1)
+
+/* Soft volume and zero cross control 1 (0x00d9) */
+#define RT5682_SV_MASK (0x1 << 15)
+#define RT5682_SV_SFT 15
+#define RT5682_SV_DIS (0x0 << 15)
+#define RT5682_SV_EN (0x1 << 15)
+#define RT5682_ZCD_MASK (0x1 << 10)
+#define RT5682_ZCD_SFT 10
+#define RT5682_ZCD_PD (0x0 << 10)
+#define RT5682_ZCD_PU (0x1 << 10)
+#define RT5682_SV_DLY_MASK (0xf)
+#define RT5682_SV_DLY_SFT 0
+
+/* Soft volume and zero cross control 2 (0x00da) */
+#define RT5682_ZCD_BST1_CBJ_MASK (0x1 << 7)
+#define RT5682_ZCD_BST1_CBJ_SFT 7
+#define RT5682_ZCD_BST1_CBJ_DIS (0x0 << 7)
+#define RT5682_ZCD_BST1_CBJ_EN (0x1 << 7)
+#define RT5682_ZCD_RECMIX_MASK (0x1)
+#define RT5682_ZCD_RECMIX_SFT 0
+#define RT5682_ZCD_RECMIX_DIS (0x0)
+#define RT5682_ZCD_RECMIX_EN (0x1)
+
+/* 4 Button Inline Command Control 2 (0x00e3) */
+#define RT5682_4BTN_IL_MASK (0x1 << 15)
+#define RT5682_4BTN_IL_EN (0x1 << 15)
+#define RT5682_4BTN_IL_DIS (0x0 << 15)
+#define RT5682_4BTN_IL_RST_MASK (0x1 << 14)
+#define RT5682_4BTN_IL_NOR (0x1 << 14)
+#define RT5682_4BTN_IL_RST (0x0 << 14)
+
+/* Analog JD Control (0x00f0) */
+#define RT5682_JDH_RS_MASK (0x1 << 4)
+#define RT5682_JDH_NO_PLUG (0x1 << 4)
+#define RT5682_JDH_PLUG (0x0 << 4)
+
+/* Chopper and Clock control for DAC (0x013a)*/
+#define RT5682_CKXEN_DAC1_MASK (0x1 << 13)
+#define RT5682_CKXEN_DAC1_SFT 13
+#define RT5682_CKGEN_DAC1_MASK (0x1 << 12)
+#define RT5682_CKGEN_DAC1_SFT 12
+
+/* Chopper and Clock control for ADC (0x013b)*/
+#define RT5682_CKXEN_ADC1_MASK (0x1 << 13)
+#define RT5682_CKXEN_ADC1_SFT 13
+#define RT5682_CKGEN_ADC1_MASK (0x1 << 12)
+#define RT5682_CKGEN_ADC1_SFT 12
+
+/* Volume test (0x013f)*/
+#define RT5682_SEL_CLK_VOL_MASK (0x1 << 15)
+#define RT5682_SEL_CLK_VOL_EN (0x1 << 15)
+#define RT5682_SEL_CLK_VOL_DIS (0x0 << 15)
+
+/* Test Mode Control 1 (0x0145) */
+#define RT5682_AD2DA_LB_MASK (0x1 << 10)
+#define RT5682_AD2DA_LB_SFT 10
+
+/* Stereo Noise Gate Control 1 (0x0160) */
+#define RT5682_NG2_EN_MASK (0x1 << 15)
+#define RT5682_NG2_EN (0x1 << 15)
+#define RT5682_NG2_DIS (0x0 << 15)
+
+/* Stereo1 DAC Silence Detection Control (0x0190) */
+#define RT5682_DEB_STO_DAC_MASK (0x7 << 4)
+#define RT5682_DEB_80_MS (0x0 << 4)
+
+/* SAR ADC Inline Command Control 1 (0x0210) */
+#define RT5682_SAR_BUTT_DET_MASK (0x1 << 15)
+#define RT5682_SAR_BUTT_DET_EN (0x1 << 15)
+#define RT5682_SAR_BUTT_DET_DIS (0x0 << 15)
+#define RT5682_SAR_BUTDET_MODE_MASK (0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_SAV (0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_NORM (0x0 << 14)
+#define RT5682_SAR_BUTDET_RST_MASK (0x1 << 13)
+#define RT5682_SAR_BUTDET_RST_NORMAL (0x1 << 13)
+#define RT5682_SAR_BUTDET_RST (0x0 << 13)
+#define RT5682_SAR_POW_MASK (0x1 << 12)
+#define RT5682_SAR_POW_EN (0x1 << 12)
+#define RT5682_SAR_POW_DIS (0x0 << 12)
+#define RT5682_SAR_RST_MASK (0x1 << 11)
+#define RT5682_SAR_RST_NORMAL (0x1 << 11)
+#define RT5682_SAR_RST (0x0 << 11)
+#define RT5682_SAR_BYPASS_MASK (0x1 << 10)
+#define RT5682_SAR_BYPASS_EN (0x1 << 10)
+#define RT5682_SAR_BYPASS_DIS (0x0 << 10)
+#define RT5682_SAR_SEL_MB1_MASK (0x1 << 9)
+#define RT5682_SAR_SEL_MB1_SEL (0x1 << 9)
+#define RT5682_SAR_SEL_MB1_NOSEL (0x0 << 9)
+#define RT5682_SAR_SEL_MB2_MASK (0x1 << 8)
+#define RT5682_SAR_SEL_MB2_SEL (0x1 << 8)
+#define RT5682_SAR_SEL_MB2_NOSEL (0x0 << 8)
+#define RT5682_SAR_SEL_MODE_MASK (0x1 << 7)
+#define RT5682_SAR_SEL_MODE_CMP (0x1 << 7)
+#define RT5682_SAR_SEL_MODE_ADC (0x0 << 7)
+#define RT5682_SAR_SEL_MB1_MB2_MASK (0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_AUTO (0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_MANU (0x0 << 5)
+#define RT5682_SAR_SEL_SIGNAL_MASK (0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_AUTO (0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_MANU (0x0 << 4)
+
+/* SAR ADC Inline Command Control 13 (0x021c) */
+#define RT5682_SAR_SOUR_MASK (0x3f)
+#define RT5682_SAR_SOUR_BTN (0x3f)
+#define RT5682_SAR_SOUR_TYPE (0x0)
+
+
+/* System Clock Source */
+enum {
+ RT5682_SCLK_S_MCLK,
+ RT5682_SCLK_S_PLL1,
+ RT5682_SCLK_S_PLL2,
+ RT5682_SCLK_S_RCCLK,
+};
+
+/* PLL Source */
+enum {
+ RT5682_PLL1_S_MCLK,
+ RT5682_PLL1_S_BCLK1,
+ RT5682_PLL1_S_RCCLK,
+};
+
+enum {
+ RT5682_AIF1,
+ RT5682_AIF2,
+ RT5682_AIFS
+};
+
+/* filter mask */
+enum {
+ RT5682_DA_STEREO1_FILTER = 0x1,
+ RT5682_AD_STEREO1_FILTER = (0x1 << 1),
+};
+
+enum {
+ RT5682_CLK_SEL_SYS,
+ RT5682_CLK_SEL_I2S1_ASRC,
+ RT5682_CLK_SEL_I2S2_ASRC,
+};
+
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+ unsigned int filter_mask, unsigned int clk_src);
+
+#endif /* __RT5682_H__ */
diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c
index 09451cd44f9b..85524acf3e9c 100644
--- a/sound/soc/codecs/dio2125.c
+++ b/sound/soc/codecs/simple-amplifier.c
@@ -21,9 +21,9 @@
#include <linux/module.h>
#include <sound/soc.h>
-#define DRV_NAME "dio2125"
+#define DRV_NAME "simple-amplifier"
-struct dio2125 {
+struct simple_amp {
struct gpio_desc *gpiod_enable;
};
@@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *control, int event)
{
struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
- struct dio2125 *priv = snd_soc_component_get_drvdata(c);
+ struct simple_amp *priv = snd_soc_component_get_drvdata(c);
int val;
switch (event) {
@@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("INL"),
SND_SOC_DAPM_INPUT("INR"),
SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event,
@@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("OUTR"),
};
-static const struct snd_soc_dapm_route dio2125_dapm_routes[] = {
+static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = {
{ "DRV", NULL, "INL" },
{ "DRV", NULL, "INR" },
{ "OUTL", NULL, "DRV" },
{ "OUTR", NULL, "DRV" },
};
-static const struct snd_soc_component_driver dio2125_component_driver = {
- .dapm_widgets = dio2125_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets),
- .dapm_routes = dio2125_dapm_routes,
- .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes),
+static const struct snd_soc_component_driver simple_amp_component_driver = {
+ .dapm_widgets = simple_amp_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets),
+ .dapm_routes = simple_amp_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes),
};
-static int dio2125_probe(struct platform_device *pdev)
+static int simple_amp_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
- struct dio2125 *priv;
+ struct simple_amp *priv;
int err;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -93,28 +93,30 @@ static int dio2125_probe(struct platform_device *pdev)
return err;
}
- return devm_snd_soc_register_component(dev, &dio2125_component_driver,
+ return devm_snd_soc_register_component(dev,
+ &simple_amp_component_driver,
NULL, 0);
}
#ifdef CONFIG_OF
-static const struct of_device_id dio2125_ids[] = {
+static const struct of_device_id simple_amp_ids[] = {
{ .compatible = "dioo,dio2125", },
+ { .compatible = "simple-audio-amplifier", },
{ }
};
-MODULE_DEVICE_TABLE(of, dio2125_ids);
+MODULE_DEVICE_TABLE(of, simple_amp_ids);
#endif
-static struct platform_driver dio2125_driver = {
+static struct platform_driver simple_amp_driver = {
.driver = {
.name = DRV_NAME,
- .of_match_table = of_match_ptr(dio2125_ids),
+ .of_match_table = of_match_ptr(simple_amp_ids),
},
- .probe = dio2125_probe,
+ .probe = simple_amp_probe,
};
-module_platform_driver(dio2125_driver);
+module_platform_driver(simple_amp_driver);
-MODULE_DESCRIPTION("ASoC DIO2125 output driver");
+MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver");
MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 52f34c94ec25..ca2dfe12344e 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -7,6 +7,9 @@
* TAS5721 support:
* Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
*
+ * TAS5707 support:
+ * Copyright (C) 2018 Jerome Brunet, Baylibre SAS <jbrunet@baylibre.com>
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
@@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = {
.vol_reg_size = 1,
};
+static const struct regmap_range tas5707_volatile_regs_range[] = {
+ regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG),
+ regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG),
+ regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG),
+};
+
+static const struct regmap_access_table tas5707_volatile_regs = {
+ .yes_ranges = tas5707_volatile_regs_range,
+ .n_yes_ranges = ARRAY_SIZE(tas5707_volatile_regs_range),
+
+};
+
+static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1);
+
+static const char * const tas5707_volume_slew_step_txt[] = {
+ "256", "512", "1024", "2048",
+};
+
+static const unsigned int tas5707_volume_slew_step_values[] = {
+ 3, 0, 1, 2,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum,
+ TAS571X_VOL_CFG_REG, 0, 0x3,
+ tas5707_volume_slew_step_txt,
+ tas5707_volume_slew_step_values);
+
+static const struct snd_kcontrol_new tas5707_controls[] = {
+ SOC_SINGLE_TLV("Master Volume",
+ TAS571X_MVOL_REG,
+ 0, 0xff, 1, tas5707_volume_tlv),
+ SOC_DOUBLE_R_TLV("Speaker Volume",
+ TAS571X_CH1_VOL_REG,
+ TAS571X_CH2_VOL_REG,
+ 0, 0xff, 1, tas5707_volume_tlv),
+ SOC_DOUBLE("Speaker Switch",
+ TAS571X_SOFT_MUTE_REG,
+ TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+ 1, 1),
+
+ SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum),
+
+ BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG),
+ BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG),
+ BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG),
+ BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG),
+ BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG),
+ BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG),
+ BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG),
+
+ BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG),
+ BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG),
+ BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG),
+ BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG),
+ BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG),
+ BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG),
+ BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG),
+};
+
+static const struct reg_default tas5707_reg_defaults[] = {
+ {TAS571X_CLK_CTRL_REG, 0x6c},
+ {TAS571X_DEV_ID_REG, 0x70},
+ {TAS571X_ERR_STATUS_REG, 0x00},
+ {TAS571X_SYS_CTRL_1_REG, 0xa0},
+ {TAS571X_SDI_REG, 0x05},
+ {TAS571X_SYS_CTRL_2_REG, 0x40},
+ {TAS571X_SOFT_MUTE_REG, 0x00},
+ {TAS571X_MVOL_REG, 0xff},
+ {TAS571X_CH1_VOL_REG, 0x30},
+ {TAS571X_CH2_VOL_REG, 0x30},
+ {TAS571X_VOL_CFG_REG, 0x91},
+ {TAS571X_MODULATION_LIMIT_REG, 0x02},
+ {TAS571X_IC_DELAY_CH1_REG, 0xac},
+ {TAS571X_IC_DELAY_CH2_REG, 0x54},
+ {TAS571X_IC_DELAY_CH3_REG, 0xac},
+ {TAS571X_IC_DELAY_CH4_REG, 0x54},
+ {TAS571X_START_STOP_PERIOD_REG, 0x0f},
+ {TAS571X_OSC_TRIM_REG, 0x82},
+ {TAS571X_BKND_ERR_REG, 0x02},
+ {TAS571X_INPUT_MUX_REG, 0x17772},
+ {TAS571X_PWM_MUX_REG, 0x1021345},
+};
+
+static const struct regmap_config tas5707_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 32,
+ .max_register = 0xff,
+ .reg_read = tas571x_reg_read,
+ .reg_write = tas571x_reg_write,
+ .reg_defaults = tas5707_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(tas5707_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas5707_volatile_regs,
+};
+
+static const struct tas571x_chip tas5707_chip = {
+ .supply_names = tas5711_supply_names,
+ .num_supply_names = ARRAY_SIZE(tas5711_supply_names),
+ .controls = tas5707_controls,
+ .num_controls = ARRAY_SIZE(tas5707_controls),
+ .regmap_config = &tas5707_regmap_config,
+ .vol_reg_size = 1,
+};
+
static const char *const tas5717_supply_names[] = {
"AVDD",
"DVDD",
@@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client)
}
static const struct of_device_id tas571x_of_match[] = {
+ { .compatible = "ti,tas5707", .data = &tas5707_chip, },
{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
@@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = {
MODULE_DEVICE_TABLE(of, tas571x_of_match);
static const struct i2c_device_id tas571x_i2c_id[] = {
+ { "tas5707", (kernel_ulong_t) &tas5707_chip },
{ "tas5711", (kernel_ulong_t) &tas5711_chip },
{ "tas5717", (kernel_ulong_t) &tas5717_chip },
{ "tas5719", (kernel_ulong_t) &tas5717_chip },
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index c45677bc26ad..bd23e89cfe79 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -53,6 +53,22 @@
#define TAS571X_PWM_MUX_REG 0x25
/* 20-byte biquad registers */
+#define TAS5707_CH1_BQ0_REG 0x29
+#define TAS5707_CH1_BQ1_REG 0x2a
+#define TAS5707_CH1_BQ2_REG 0x2b
+#define TAS5707_CH1_BQ3_REG 0x2c
+#define TAS5707_CH1_BQ4_REG 0x2d
+#define TAS5707_CH1_BQ5_REG 0x2e
+#define TAS5707_CH1_BQ6_REG 0x2f
+
+#define TAS5707_CH2_BQ0_REG 0x30
+#define TAS5707_CH2_BQ1_REG 0x31
+#define TAS5707_CH2_BQ2_REG 0x32
+#define TAS5707_CH2_BQ3_REG 0x33
+#define TAS5707_CH2_BQ4_REG 0x34
+#define TAS5707_CH2_BQ5_REG 0x35
+#define TAS5707_CH2_BQ6_REG 0x36
+
#define TAS5717_CH1_BQ0_REG 0x26
#define TAS5717_CH1_BQ1_REG 0x27
#define TAS5717_CH1_BQ2_REG 0x28
diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c
index 225c210ac38f..7f3b79c5a563 100644
--- a/sound/soc/codecs/tda7419.c
+++ b/sound/soc/codecs/tda7419.c
@@ -142,9 +142,9 @@ struct tda7419_vol_control {
static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc)
{
if (tvc->reg == tvc->rreg)
- return 0;
+ return false;
- return 1;
+ return true;
}
static int tda7419_vol_info(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c
index d18ff17719cc..7396a6e5277e 100644
--- a/sound/soc/codecs/tscs42xx.c
+++ b/sound/soc/codecs/tscs42xx.c
@@ -625,25 +625,34 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol,
static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
/* Volumes */
- SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR,
+ SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR,
FB_HPVOLL, 0x7F, 0, hpvol_scale),
- SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR,
+ SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR,
FB_SPKVOLL, 0x7F, 0, spkvol_scale),
- SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR,
+ SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR,
FB_DACVOLL, 0xFF, 0, dacvol_scale),
- SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR,
+ SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR,
FB_ADCVOLL, 0xFF, 0, adcvol_scale),
- SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR,
+ SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR,
FB_INVOLL, 0x3F, 0, invol_scale),
/* INSEL */
- SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR,
+ SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR,
FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB,
0, mic_boost_scale),
/* Input Channel Map */
SOC_ENUM("Input Channel Map", ch_map_select_enum),
+ /* Mic Bias */
+ SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0),
+
+ /* Headphone Auto Switching */
+ SOC_SINGLE("Headphone Auto Switching Switch",
+ R_CTL, FB_CTL_HPSWEN, 1, 0),
+ SOC_SINGLE("Headphone Detect Polarity Toggle Switch",
+ R_CTL, FB_CTL_HPSWPOL, 1, 0),
+
/* Coefficient Ram */
COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00),
COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05),
@@ -733,9 +742,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0),
SOC_SINGLE("Comp Switch",
R_CLECTL, FB_CLECTL_COMP_EN, 1, 0),
- SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume",
+ SOC_SINGLE_TLV("CLE Make-Up Gain Volume",
R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale),
- SOC_SINGLE_TLV("Comp Thresh Playback Volume",
+ SOC_SINGLE_TLV("Comp Thresh Volume",
R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale),
SOC_ENUM("Comp Ratio", compressor_ratio_enum),
SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2),
@@ -766,9 +775,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
SOC_SINGLE("MBC1 Phase Invert Switch",
R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0),
- SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume",
R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale),
- SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume",
R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale),
SOC_ENUM("DAC MBC1 Comp Ratio",
dac_mbc1_compressor_ratio_enum),
@@ -778,9 +787,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
SOC_SINGLE("MBC2 Phase Invert Switch",
R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0),
- SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume",
R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale),
- SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume",
R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale),
SOC_ENUM("DAC MBC2 Comp Ratio",
dac_mbc2_compressor_ratio_enum),
@@ -790,9 +799,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
SOC_SINGLE("MBC3 Phase Invert Switch",
R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0),
- SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume",
R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale),
- SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume",
+ SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume",
R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale),
SOC_ENUM("DAC MBC3 Comp Ratio",
dac_mbc3_compressor_ratio_enum),
diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h
index 814c8f3c4a68..6b3a21081635 100644
--- a/sound/soc/codecs/tscs42xx.h
+++ b/sound/soc/codecs/tscs42xx.h
@@ -34,6 +34,7 @@ enum {
#define R_DACSR 0x19
#define R_PWRM1 0x1A
#define R_PWRM2 0x1B
+#define R_CTL 0x1C
#define R_CONFIG0 0x1F
#define R_CONFIG1 0x20
#define R_DMICCTL 0x24
@@ -1110,6 +1111,13 @@ enum {
#define RV_PWRM2_VREF_DISABLE \
RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF)
+/******************************
+ * R_CTL (0x1C) *
+ ******************************/
+
+/* Fiel Offsets */
+#define FB_CTL_HPSWEN 7
+#define FB_CTL_HPSWPOL 6
/******************************
* R_CONFIG0 (0x1F) *
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bfd1abd72253..94675da514c8 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component,
case TWL6040_REG_HFRCTL:
return priv->dl2_unmuted;
default:
- return 1;
+ return true;
}
}
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 3663b9fd4d65..deff65161504 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
WM2200_SPK1R_MUTE_SHIFT, 1, 1),
SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
};
WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = {
static int wm2200_probe(struct snd_soc_component *component)
{
struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component);
- int ret;
wm2200->component = component;
- ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2);
- if (ret != 0)
- return ret;
-
- return ret;
+ return 0;
}
static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e239f4bf2460..9e987cf07450 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
case WM5100_OUTPUT_STATUS_2:
case WM5100_INPUT_ENABLES_STATUS:
case WM5100_MIC_DETECT_3:
- return 1;
+ return true;
default:
if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
(reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
(reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
(reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
(reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
- return 1;
+ return true;
else
- return 0;
+ return false;
}
}
@@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
case WM5100_DSP3_CONTROL_28:
case WM5100_DSP3_CONTROL_29:
case WM5100_DSP3_CONTROL_30:
- return 1;
+ return true;
default:
if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
(reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
(reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
(reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
(reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
- return 1;
+ return true;
else
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1ac83388d1b8..7e817e1877c2 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
};
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2094,6 +2096,12 @@ static int wm5102_probe(struct platform_device *pdev)
return ret;
}
+ ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+ if (ret != 0)
+ dev_warn(&pdev->dev,
+ "Failed to set compressed IRQ as a wake source: %d\n",
+ ret);
+
arizona_init_common(arizona);
ret = arizona_init_vol_limit(arizona);
@@ -2117,6 +2125,7 @@ static int wm5102_probe(struct platform_device *pdev)
err_spk_irqs:
arizona_free_spk_irqs(arizona);
err_dsp_irq:
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
return ret;
@@ -2133,6 +2142,7 @@ static int wm5102_remove(struct platform_device *pdev)
arizona_free_spk_irqs(arizona);
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
return 0;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index fb9835dcd836..b0789a03d699 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
+WM_ADSP_FW_CONTROL("DSP4", 3),
};
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2455,6 +2460,12 @@ static int wm5110_probe(struct platform_device *pdev)
return ret;
}
+ ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+ if (ret != 0)
+ dev_warn(&pdev->dev,
+ "Failed to set compressed IRQ as a wake source: %d\n",
+ ret);
+
arizona_init_common(arizona);
ret = arizona_init_vol_limit(arizona);
@@ -2478,6 +2489,7 @@ static int wm5110_probe(struct platform_device *pdev)
err_spk_irqs:
arizona_free_spk_irqs(arizona);
err_dsp_irq:
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
return ret;
@@ -2496,6 +2508,7 @@ static int wm5110_remove(struct platform_device *pdev)
arizona_free_spk_irqs(arizona);
+ arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
return 0;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 7b8b6ef2f632..6cb3c153ba19 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg)
case WM8903_DC_SERVO_READBACK_2:
case WM8903_DC_SERVO_READBACK_3:
case WM8903_DC_SERVO_READBACK_4:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9037a35b931d..1965635ec07c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif1 |= 0x3;
break;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index ba44e3d6c1e0..cd204f79647d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif |= WM8955_LRP;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif |= 0x3;
break;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index c30f5aa392c6..8dc1f3d6a988 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
iface |= 0x000c;
break;
}
+ /* fall through */
default:
dev_err(component->dev, "unsupported width %d\n",
params_width(params));
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index f70f563d59f3..68b4cadc308f 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_B:
aif |= WM8961_LRP;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif |= 3;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index a11e9d6bf950..efd8910b1ff7 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif0 |= WM8962_LRCLK_INV | 3;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif0 |= 3;
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 62200117444b..6e52c6a8bab3 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate)
/* The set of rates we can generate from the above for each SYSCLK */
static const unsigned int rates_12288[] = {
- 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+ 8000, 12000, 16000, 24000, 32000, 48000, 96000,
};
static const struct snd_pcm_hw_constraint_list constraints_12288 = {
@@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = {
};
static const unsigned int rates_12[] = {
- 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000,
48000, 88235, 96000,
};
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 411b9eee88c2..457bc437ce54 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8990_RESET:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 7fdfdf3f6e67..14f1b0c0d286 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2,
WM8994_OPCLK_ENA, 0);
}
+ break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 60e227832331..68c99fe37097 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif |= WM8995_AIF1_LRCLK_INV;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif |= (0x3 << WM8995_AIF1_FMT_SHIFT);
break;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d9d206046f8c..91711f8958c5 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg)
case WM8996_RIGHT_PDM_SPEAKER:
case WM8996_PDM_SPEAKER_MUTE_SEQUENCE:
case WM8996_PDM_SPEAKER_VOLUME:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
@@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg)
case WM8996_MIC_DETECT_3:
case WM8996_HEADPHONE_DETECT_1:
case WM8996_HEADPHONE_DETECT_2:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
@@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
case 24576000:
ratediv = WM8996_SYSCLK_DIV;
wm8996->sysclk /= 2;
+ /* fall through */
case 11289600:
case 12288000:
snd_soc_component_update_bits(component, WM8996_AIF_RATE,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 5a0ea7b3c149..399255d1f78a 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif2 |= WM9081_AIF_LRCLK_INV;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif2 |= 0x3;
break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 2fcdd84021a5..f61656070225 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -10,6 +10,7 @@
* published by the Free Software Foundation.
*/
+#include <linux/ctype.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
@@ -35,15 +36,15 @@
#include "wm_adsp.h"
#define adsp_crit(_dsp, fmt, ...) \
- dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+ dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
#define adsp_err(_dsp, fmt, ...) \
- dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+ dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
#define adsp_warn(_dsp, fmt, ...) \
- dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+ dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
#define adsp_info(_dsp, fmt, ...) \
- dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+ dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
#define adsp_dbg(_dsp, fmt, ...) \
- dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+ dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
#define ADSP1_CONTROL_1 0x00
#define ADSP1_CONTROL_2 0x02
@@ -418,7 +419,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = {
{
.id = SND_AUDIOCODEC_BESPOKE,
.desc = {
- .max_ch = 1,
+ .max_ch = 8,
.sample_rates = { 16000 },
.num_sample_rates = 1,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
@@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
struct snd_soc_component *component)
{
struct dentry *root = NULL;
- char *root_name;
int i;
if (!component->debugfs_root) {
@@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
goto err;
}
- root_name = kmalloc(PAGE_SIZE, GFP_KERNEL);
- if (!root_name)
- goto err;
-
- snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num);
- root = debugfs_create_dir(root_name, component->debugfs_root);
- kfree(root_name);
+ root = debugfs_create_dir(dsp->name, component->debugfs_root);
if (!root)
goto err;
@@ -684,8 +678,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp)
}
#endif
-static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -695,9 +689,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
return 0;
}
+EXPORT_SYMBOL_GPL(wm_adsp_fw_get);
-static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -721,8 +716,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
return ret;
}
+EXPORT_SYMBOL_GPL(wm_adsp_fw_put);
-static const struct soc_enum wm_adsp_fw_enum[] = {
+const struct soc_enum wm_adsp_fw_enum[] = {
SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
@@ -731,24 +727,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = {
SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
};
-
-const struct snd_kcontrol_new wm_adsp_fw_controls[] = {
- SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5],
- wm_adsp_fw_get, wm_adsp_fw_put),
- SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6],
- wm_adsp_fw_get, wm_adsp_fw_put),
-};
-EXPORT_SYMBOL_GPL(wm_adsp_fw_controls);
+EXPORT_SYMBOL_GPL(wm_adsp_fw_enum);
static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp,
int type)
@@ -1330,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
switch (dsp->fw_ver) {
case 0:
case 1:
- snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x",
- dsp->num, region_name, alg_region->alg);
+ snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x",
+ dsp->name, region_name, alg_region->alg);
break;
default:
ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
- "DSP%d%c %.12s %x", dsp->num, *region_name,
+ "%s%c %.12s %x", dsp->name, *region_name,
wm_adsp_fw_text[dsp->fw], alg_region->alg);
/* Truncate the subname from the start if it is too long */
@@ -1343,6 +1322,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2;
int skip = 0;
+ if (dsp->component->name_prefix)
+ avail -= strlen(dsp->component->name_prefix) + 1;
+
if (subname_len > avail)
skip = subname_len - avail;
@@ -1604,6 +1586,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp,
if (ret)
return -EINVAL;
break;
+ case WMFW_CTL_TYPE_HOST_BUFFER:
+ ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk,
+ WMFW_CTL_FLAG_SYS |
+ WMFW_CTL_FLAG_VOLATILE |
+ WMFW_CTL_FLAG_READABLE,
+ 0);
+ if (ret)
+ return -EINVAL;
+ break;
default:
adsp_err(dsp, "Unknown control type: %d\n",
coeff_blk.ctl_type);
@@ -1651,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
if (file == NULL)
return -ENOMEM;
- snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num,
+ snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name,
wm_adsp_fw[dsp->fw].file);
file[PAGE_SIZE - 1] = '\0';
@@ -1871,9 +1862,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp,
}
static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
+ const struct wm_adsp_region *mem,
unsigned int pos, unsigned int len)
{
void *alg;
+ unsigned int reg;
int ret;
__be32 val;
@@ -1888,7 +1881,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
}
/* Read the terminator first to validate the length */
- ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val));
+ reg = wm_adsp_region_to_reg(mem, pos + len);
+
+ ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
if (ret != 0) {
adsp_err(dsp, "Failed to read algorithm list end: %d\n",
ret);
@@ -1897,13 +1892,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
if (be32_to_cpu(val) != 0xbedead)
adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n",
- pos + len, be32_to_cpu(val));
+ reg, be32_to_cpu(val));
+
+ /* Convert length from DSP words to bytes */
+ len *= sizeof(u32);
- alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA);
+ alg = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!alg)
return ERR_PTR(-ENOMEM);
- ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2);
+ reg = wm_adsp_region_to_reg(mem, pos);
+
+ ret = regmap_raw_read(dsp->regmap, reg, alg, len);
if (ret != 0) {
adsp_err(dsp, "Failed to read algorithm list: %d\n", ret);
kfree(alg);
@@ -2002,10 +2002,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
if (IS_ERR(alg_region))
return PTR_ERR(alg_region);
- pos = sizeof(adsp1_id) / 2;
- len = (sizeof(*adsp1_alg) * n_algs) / 2;
+ /* Calculate offset and length in DSP words */
+ pos = sizeof(adsp1_id) / sizeof(u32);
+ len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32);
- adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+ adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
if (IS_ERR(adsp1_alg))
return PTR_ERR(adsp1_alg);
@@ -2113,10 +2114,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp)
if (IS_ERR(alg_region))
return PTR_ERR(alg_region);
- pos = sizeof(adsp2_id) / 2;
- len = (sizeof(*adsp2_alg) * n_algs) / 2;
+ /* Calculate offset and length in DSP words */
+ pos = sizeof(adsp2_id) / sizeof(u32);
+ len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32);
- adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+ adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
if (IS_ERR(adsp2_alg))
return PTR_ERR(adsp2_alg);
@@ -2218,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
if (file == NULL)
return -ENOMEM;
- snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num,
+ snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name,
wm_adsp_fw[dsp->fw].file);
file[PAGE_SIZE - 1] = '\0';
@@ -2390,8 +2392,38 @@ out:
return ret;
}
+static int wm_adsp_create_name(struct wm_adsp *dsp)
+{
+ char *p;
+
+ if (!dsp->name) {
+ dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d",
+ dsp->num);
+ if (!dsp->name)
+ return -ENOMEM;
+ }
+
+ if (!dsp->fwf_name) {
+ p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ dsp->fwf_name = p;
+ for (; *p != 0; ++p)
+ *p = tolower(*p);
+ }
+
+ return 0;
+}
+
int wm_adsp1_init(struct wm_adsp *dsp)
{
+ int ret;
+
+ ret = wm_adsp_create_name(dsp);
+ if (ret)
+ return ret;
+
INIT_LIST_HEAD(&dsp->alg_regions);
mutex_init(&dsp->pwr_lock);
@@ -2642,7 +2674,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
- struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+ struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct wm_adsp *dsp = &dsps[mc->shift - 1];
ucontrol->value.integer.value[0] = dsp->preloaded;
@@ -2654,13 +2689,14 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
- struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+ struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
+ struct wm_adsp *dsp = &dsps[mc->shift - 1];
char preload[32];
- snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift);
+ snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
dsp->preloaded = ucontrol->value.integer.value[0];
@@ -2671,6 +2707,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
snd_soc_dapm_sync(dapm);
+ flush_work(&dsp->boot_work);
+
return 0;
}
EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put);
@@ -2853,17 +2891,14 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp
{
char preload[32];
- snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num);
-
+ snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
snd_soc_component_disable_pin(component, preload);
wm_adsp2_init_debugfs(dsp, component);
dsp->component = component;
- return snd_soc_add_component_controls(component,
- &wm_adsp_fw_controls[dsp->num - 1],
- 1);
+ return 0;
}
EXPORT_SYMBOL_GPL(wm_adsp2_component_probe);
@@ -2879,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp)
{
int ret;
+ ret = wm_adsp_create_name(dsp);
+ if (ret)
+ return ret;
+
switch (dsp->rev) {
case 0:
/*
@@ -3186,7 +3225,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf,
buf->host_buf_ptr + field_offset, data);
}
-static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf)
{
struct wm_adsp_alg_region *alg_region;
struct wm_adsp *dsp = buf->dsp;
@@ -3225,6 +3264,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
return 0;
}
+static struct wm_coeff_ctl *
+wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf)
+{
+ struct wm_adsp *dsp = buf->dsp;
+ struct wm_coeff_ctl *ctl;
+
+ list_for_each_entry(ctl, &dsp->ctl_list, list) {
+ if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER)
+ continue;
+
+ if (!ctl->enabled)
+ continue;
+
+ return ctl;
+ }
+
+ return NULL;
+}
+
+static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+{
+ struct wm_adsp *dsp = buf->dsp;
+ struct wm_coeff_ctl *ctl;
+ unsigned int reg;
+ u32 val;
+ int i, ret;
+
+ ctl = wm_adsp_find_host_buffer_ctrl(buf);
+ if (!ctl)
+ return wm_adsp_legacy_host_buf_addr(buf);
+
+ ret = wm_coeff_base_reg(ctl, &reg);
+ if (ret)
+ return ret;
+
+ for (i = 0; i < 5; ++i) {
+ ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
+ if (ret < 0)
+ return ret;
+
+ if (val)
+ break;
+
+ usleep_range(1000, 2000);
+ }
+
+ if (!val)
+ return -EIO;
+
+ buf->host_buf_ptr = be32_to_cpu(val);
+ adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr);
+
+ return 0;
+}
+
static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
{
const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps;
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index bc6d359f0533..4b8778b0b06c 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -57,6 +57,8 @@ struct wm_adsp_compr_buf;
struct wm_adsp {
const char *part;
+ const char *name;
+ const char *fwf_name;
int rev;
int num;
int type;
@@ -121,7 +123,11 @@ struct wm_adsp {
.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
-extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
+#define WM_ADSP_FW_CONTROL(dspname, num) \
+ SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \
+ wm_adsp_fw_get, wm_adsp_fw_put)
+
+extern const struct soc_enum wm_adsp_fw_enum[];
int wm_adsp1_init(struct wm_adsp *dsp);
int wm_adsp2_init(struct wm_adsp *dsp);
@@ -144,6 +150,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream);
int wm_adsp_compr_free(struct snd_compr_stream *stream);
diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h
index ec78b9da020f..0c3f50acb8b1 100644
--- a/sound/soc/codecs/wmfw.h
+++ b/sound/soc/codecs/wmfw.h
@@ -29,6 +29,7 @@
/* Non-ALSA coefficient types start at 0x1000 */
#define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */
#define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */
+#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */
struct wmfw_header {
char magic[4];
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 807040bb3921..a3206e65e5e5 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
* rate is lowered.
*/
inv_fs = true;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_A:
dev->mode = MOD_DSP_A;
break;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 47c0c821d325..f70db8412c7c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
handled_mask |= XUNDRN;
substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
- if (substream) {
- snd_pcm_stream_lock_irq(substream);
- if (snd_pcm_running(substream))
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- snd_pcm_stream_unlock_irq(substream);
- }
+ if (substream)
+ snd_pcm_stop_xrun(substream);
}
if (!handled_mask)
@@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
handled_mask |= ROVRN;
substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
- if (substream) {
- snd_pcm_stream_lock_irq(substream);
- if (snd_pcm_running(substream))
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- snd_pcm_stream_unlock_irq(substream);
- }
+ if (substream)
+ snd_pcm_stop_xrun(substream);
}
if (!handled_mask)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 4a6750aa3637..44433b20435c 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -1,14 +1,10 @@
-/*
- * Freescale Generic ASoC Sound Card driver with ASRC
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale Generic ASoC Sound Card driver with ASRC
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
#include <linux/clk.h>
#include <linux/i2c.h>
@@ -199,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_none(mask);
- snd_mask_set(mask, (__force int)priv->asrc_format);
+ snd_mask_set_format(mask, priv->asrc_format);
return 0;
}
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index adfb8135d739..528e8b108422 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
#include <linux/clk.h>
#include <linux/delay.h>
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index d558dd5499a5..c60075112570 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -1,13 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
/*
* fsl_asrc.h - Freescale ASRC ALSA SoC header file
*
* Copyright (C) 2014 Freescale Semiconductor, Inc.
*
* Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
*/
#ifndef _FSL_ASRC_H
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 565e16d8fe85..1033ac6631b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Platform (DMA) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Platform (DMA) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
#include <linux/dma-mapping.h>
#include <linux/module.h>
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 8f43110373b8..c1d1d06783e5 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
break;
case ESAI_HCKT_EXTAL:
ecr |= ESAI_ECR_ETI;
+ /* fall through */
case ESAI_HCKR_EXTAL:
ecr |= ESAI_ECR_ERI;
break;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 9b59d87b61bf..740b90df44bb 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
- rate_ideal = rate[index] * txclk_df * 64;
+ rate_ideal = rate[index] * txclk_df * 64ULL;
if (round)
rate_actual = clk_round_rate(clk, rate_ideal);
else
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7592b0406370..7f0fa4b52223 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -1,14 +1,10 @@
-/**
- * Freescale ALSA SoC Machine driver utility
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ALSA SoC Machine driver utility
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2010 Freescale Semiconductor, Inc.
#include <linux/module.h>
#include <linux/of_address.h>
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
index 1687b66ef18e..c5dc2a14b492 100644
--- a/sound/soc/fsl/fsl_utils.h
+++ b/sound/soc/fsl/fsl_utils.h
@@ -1,13 +1,10 @@
-/**
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
* Freescale ALSA SoC Machine driver utility
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
*/
#ifndef _FSL_UTILS_H
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..c29200cf755a 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -1,14 +1,7 @@
-/*
- * Copyright 2012 Freescale Semiconductor, Inc.
- * Copyright 2012 Linaro Ltd.
- *
- * The code contained herein is licensed under the GNU General Public
- * License. You may obtain a copy of the GNU General Public License
- * Version 2 or later at the following locations:
- *
- * http://www.opensource.org/licenses/gpl-license.html
- * http://www.gnu.org/copyleft/gpl.html
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+// Copyright 2012 Linaro Ltd.
#include <linux/module.h>
#include <linux/of.h>
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index d93bacacbd5b..2094d2c8919f 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC audio graph sound card support
- *
- * Copyright (C) 2016 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph sound card support
+//
+// Copyright (C) 2016 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
#include <linux/clk.h>
#include <linux/device.h>
#include <linux/gpio.h>
@@ -21,7 +18,6 @@
#include <linux/of_graph.h>
#include <linux/platform_device.h>
#include <linux/string.h>
-#include <sound/jack.h>
#include <sound/simple_card_utils.h>
struct graph_card_data {
@@ -32,6 +28,8 @@ struct graph_card_data {
unsigned int mclk_fs;
} *dai_props;
unsigned int mclk_fs;
+ struct asoc_simple_jack hp_jack;
+ struct asoc_simple_jack mic_jack;
struct snd_soc_dai_link *dai_link;
struct gpio_desc *pa_gpio;
};
@@ -278,6 +276,22 @@ static int asoc_graph_get_dais_count(struct device *dev)
return count;
}
+static int asoc_graph_soc_card_probe(struct snd_soc_card *card)
+{
+ struct graph_card_data *priv = snd_soc_card_get_drvdata(card);
+ int ret;
+
+ ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL);
+ if (ret < 0)
+ return ret;
+
+ ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
static int asoc_graph_card_probe(struct platform_device *pdev)
{
struct graph_card_data *priv;
@@ -319,6 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
card->num_links = num;
card->dapm_widgets = asoc_graph_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets);
+ card->probe = asoc_graph_soc_card_probe;
ret = asoc_graph_card_parse_of(priv);
if (ret < 0) {
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 095ef6426d42..92882e392d6c 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -1,17 +1,14 @@
-/*
- * ASoC audio graph SCU sound card support
- *
- * Copyright (C) 2017 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on
- * ${LINUX}/sound/soc/generic/simple-scu-card.c
- * ${LINUX}/sound/soc/generic/audio-graph-card.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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph SCU sound card support
+//
+// Copyright (C) 2017 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on
+// ${LINUX}/sound/soc/generic/simple-scu-card.c
+// ${LINUX}/sound/soc/generic/audio-graph-card.c
+
#include <linux/clk.h>
#include <linux/device.h>
#include <linux/gpio.h>
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 3751a07de6aa..d3f3f0fec74c 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -1,16 +1,17 @@
-/*
- * simple-card-utils.c
- *
- * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// simple-card-utils.c
+//
+// Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/of.h>
+#include <linux/of_gpio.h>
#include <linux/of_graph.h>
+#include <sound/jack.h>
#include <sound/simple_card_utils.h>
void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data,
@@ -419,6 +420,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets);
+int asoc_simple_card_init_jack(struct snd_soc_card *card,
+ struct asoc_simple_jack *sjack,
+ int is_hp, char *prefix)
+{
+ struct device *dev = card->dev;
+ enum of_gpio_flags flags;
+ char prop[128];
+ char *pin_name;
+ char *gpio_name;
+ int mask;
+ int det;
+
+ if (!prefix)
+ prefix = "";
+
+ sjack->gpio.gpio = -ENOENT;
+
+ if (is_hp) {
+ snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
+ pin_name = "Headphones";
+ gpio_name = "Headphone detection";
+ mask = SND_JACK_HEADPHONE;
+ } else {
+ snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
+ pin_name = "Mic Jack";
+ gpio_name = "Mic detection";
+ mask = SND_JACK_MICROPHONE;
+ }
+
+ det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
+ if (det == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+
+ if (gpio_is_valid(det)) {
+ sjack->pin.pin = pin_name;
+ sjack->pin.mask = mask;
+
+ sjack->gpio.name = gpio_name;
+ sjack->gpio.report = mask;
+ sjack->gpio.gpio = det;
+ sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW);
+ sjack->gpio.debounce_time = 150;
+
+ snd_soc_card_jack_new(card, pin_name, mask,
+ &sjack->jack,
+ &sjack->pin, 1);
+
+ snd_soc_jack_add_gpios(&sjack->jack, 1,
+ &sjack->gpio);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack);
+
/* Module information */
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 8b374af86a6e..64bf3560c1d1 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -1,32 +1,20 @@
-/*
- * ASoC simple sound card support
- *
- * Copyright (C) 2012 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple sound card support
+//
+// Copyright (C) 2012 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include <linux/clk.h>
#include <linux/device.h>
-#include <linux/gpio.h>
#include <linux/module.h>
#include <linux/of.h>
-#include <linux/of_gpio.h>
#include <linux/platform_device.h>
#include <linux/string.h>
-#include <sound/jack.h>
#include <sound/simple_card.h>
#include <sound/soc-dai.h>
#include <sound/soc.h>
-struct asoc_simple_jack {
- struct snd_soc_jack jack;
- struct snd_soc_jack_pin pin;
- struct snd_soc_jack_gpio gpio;
-};
-
struct simple_card_data {
struct snd_soc_card snd_card;
struct simple_dai_props {
@@ -49,61 +37,6 @@ struct simple_card_data {
#define CELL "#sound-dai-cells"
#define PREFIX "simple-audio-card,"
-#define asoc_simple_card_init_hp(card, sjack, prefix)\
- asoc_simple_card_init_jack(card, sjack, 1, prefix)
-#define asoc_simple_card_init_mic(card, sjack, prefix)\
- asoc_simple_card_init_jack(card, sjack, 0, prefix)
-static int asoc_simple_card_init_jack(struct snd_soc_card *card,
- struct asoc_simple_jack *sjack,
- int is_hp, char *prefix)
-{
- struct device *dev = card->dev;
- enum of_gpio_flags flags;
- char prop[128];
- char *pin_name;
- char *gpio_name;
- int mask;
- int det;
-
- sjack->gpio.gpio = -ENOENT;
-
- if (is_hp) {
- snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
- pin_name = "Headphones";
- gpio_name = "Headphone detection";
- mask = SND_JACK_HEADPHONE;
- } else {
- snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
- pin_name = "Mic Jack";
- gpio_name = "Mic detection";
- mask = SND_JACK_MICROPHONE;
- }
-
- det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
- if (det == -EPROBE_DEFER)
- return -EPROBE_DEFER;
-
- if (gpio_is_valid(det)) {
- sjack->pin.pin = pin_name;
- sjack->pin.mask = mask;
-
- sjack->gpio.name = gpio_name;
- sjack->gpio.report = mask;
- sjack->gpio.gpio = det;
- sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW);
- sjack->gpio.debounce_time = 150;
-
- snd_soc_card_jack_new(card, pin_name, mask,
- &sjack->jack,
- &sjack->pin, 1);
-
- snd_soc_jack_add_gpios(&sjack->jack, 1,
- &sjack->gpio);
- }
-
- return 0;
-}
-
static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -213,14 +146,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
if (ret < 0)
return ret;
- ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX);
- if (ret < 0)
- return ret;
-
- ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX);
- if (ret < 0)
- return ret;
-
return 0;
}
@@ -414,6 +339,22 @@ card_parse_end:
return ret;
}
+static int asoc_simple_soc_card_probe(struct snd_soc_card *card)
+{
+ struct simple_card_data *priv = snd_soc_card_get_drvdata(card);
+ int ret;
+
+ ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX);
+ if (ret < 0)
+ return ret;
+
+ ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
static int asoc_simple_card_probe(struct platform_device *pdev)
{
struct simple_card_data *priv;
@@ -449,6 +390,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
card->dev = dev;
card->dai_link = priv->dai_link;
card->num_links = num;
+ card->probe = asoc_simple_soc_card_probe;
if (np && of_device_is_available(np)) {
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 487716559deb..16a83bc51e0e 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC simple SCU sound card support
- *
- * Copyright (C) 2015 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple SCU sound card support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
#include <linux/clk.h>
#include <linux/device.h>
#include <linux/module.h>
diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 6a8b253c58d2..5455d6e0ab53 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id,
stream->cumm_bytes += bytes;
dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes);
- memcpy_fromio(&fw_tstamp,
- ((void *)(ctx->mailbox + ctx->tstamp)
- +(str_id * sizeof(fw_tstamp))),
- sizeof(fw_tstamp));
+ addr = ((void __iomem *)(ctx->mailbox + ctx->tstamp)) +
+ (str_id * sizeof(fw_tstamp));
+
+ memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
fw_tstamp.bytes_copied = stream->cumm_bytes;
dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n",
fw_tstamp.bytes_copied, bytes);
- addr = ((void *)(ctx->mailbox + ctx->tstamp)) +
- (str_id * sizeof(fw_tstamp));
offset = offsetof(struct snd_sst_tstamp, bytes_copied);
sst_shim_write(addr, offset, fw_tstamp.bytes_copied);
return 0;
@@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id,
struct snd_sst_tstamp fw_tstamp = {0,};
struct stream_info *stream;
struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+ void __iomem *addr;
+
+ addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+ (str_id * sizeof(fw_tstamp));
- memcpy_fromio(&fw_tstamp,
- ((void *)(ctx->mailbox + ctx->tstamp)
- +(str_id * sizeof(fw_tstamp))),
- sizeof(fw_tstamp));
+ memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
stream = get_stream_info(ctx, str_id);
if (!stream)
@@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
struct snd_sst_tstamp fw_tstamp;
unsigned int str_id;
struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+ void __iomem *addr;
str_id = info->str_id;
stream = get_stream_info(ctx, str_id);
@@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
return -EINVAL;
substream = stream->pcm_substream;
- memcpy_fromio(&fw_tstamp,
- ((void *)(ctx->mailbox + ctx->tstamp)
- + (str_id * sizeof(fw_tstamp))),
- sizeof(fw_tstamp));
+ addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+ (str_id * sizeof(fw_tstamp));
+
+ memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
+
return sst_calc_tstamp(ctx, info, substream, &fw_tstamp);
}
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index a686eef2cf7f..27413ebae956 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count)
/* __iowrite32_copy uses 32-bit count values so divide by 4 for
* right count in words
*/
- __iowrite32_copy(dst, src, count/4);
+ __iowrite32_copy(dst, src, count / 4);
}
void memcpy32_fromio(void *dst, const void __iomem *src, int count)
{
- /* __iowrite32_copy uses 32-bit count values so divide by 4 for
+ /* __ioread32_copy uses 32-bit count values so divide by 4 for
* right count in words
*/
- __iowrite32_copy(dst, src, count/4);
+ __ioread32_copy(dst, src, count / 4);
}
/**
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 24797482a3d2..cccda87f4b34 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
Say Y if you have such a device.
If unsure select "N".
+config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
+ tristate "GLK with RT5682 and MAX98357A in I2S Mode"
+ depends on MFD_INTEL_LPSS && I2C && ACPI
+ select SND_SOC_RT5682
+ select SND_SOC_MAX98357A
+ select SND_SOC_DMIC
+ select SND_SOC_HDAC_HDMI
+ select SND_HDA_DSP_LOADER
+ help
+ This adds support for ASoC machine driver for Geminilake platforms
+ with RT5682 + MAX98357A I2S audio codec.
+ Say Y if you have such a device.
+ If unsure select "N".
+
endif ## SND_SOC_INTEL_SKYLAKE
endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 92b5507291af..87ef8b4058e5 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
snd-soc-sst-broadwell-objs := broadwell.o
snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
+snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
+obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 6ea360f33575..efcfd906c856 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
channels->min = channels->max = 2;
/* set SSP0 to 16 bit */
- snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
- SNDRV_PCM_HW_PARAM_FIRST_MASK],
- SNDRV_PCM_FORMAT_S16_LE);
+ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 40eb979d5ac1..6f052fc8d1e2 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
@@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_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 broxton_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);
@@ -586,7 +600,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
static struct platform_driver broxton_audio = {
.probe = broxton_audio_probe,
.driver = {
- .name = "bxt_da7219_max98357a_i2s",
+ .name = "bxt_da7219_max98357a",
.pm = &snd_soc_pm_ops,
},
};
@@ -599,4 +613,4 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>");
MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s");
+MODULE_ALIAS("platform:bxt_da7219_max98357a");
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index b68c289558a8..27308337ab12 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP5 to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 33065ba294a9..d32844f94d74 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
},
.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
BYT_RT5640_JD_SRC_JD1_IN4P |
- BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_TH_1500UA |
BYT_RT5640_OVCD_SF_0P75 |
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
@@ -464,12 +464,38 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
BYT_RT5640_MCLK_EN),
},
{
+ /* Chuwi Vi10 (CWI505) */
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"),
+ DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"),
+ DMI_MATCH(DMI_SYS_VENDOR, "ilife"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "S165"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
+ {
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
},
.driver_data = (void *)(BYT_RT5640_DMIC1_MAP),
},
+ { /* Connect Tablet 9 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
@@ -536,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
+ { /* Lenovo Miix 2 8 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"),
+ DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"),
+ },
+ .driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_MCLK_EN),
+ },
{ /* MSI S100 tablet */
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."),
@@ -549,6 +588,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
BYT_RT5640_DIFF_MIC |
BYT_RT5640_MCLK_EN),
},
+ { /* Nuvison/TMax TM800W560 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_JD_NOT_INV |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{ /* Pipo W4 */
.matches = {
DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 987720e203f9..f8a68bdb3885 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -26,8 +26,12 @@
#include <linux/clk.h>
#include <linux/device.h>
#include <linux/dmi.h>
+#include <linux/input.h>
+#include <linux/gpio/consumer.h>
+#include <linux/gpio/machine.h>
#include <linux/slab.h>
#include <asm/cpu_device_id.h>
+#include <asm/intel-family.h>
#include <asm/platform_sst_audio.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -42,8 +46,6 @@ enum {
BYT_RT5651_IN1_MAP,
BYT_RT5651_IN2_MAP,
BYT_RT5651_IN1_IN2_MAP,
- BYT_RT5651_IN1_HS_IN3_MAP,
- BYT_RT5651_IN2_HS_IN3_MAP,
};
enum {
@@ -76,21 +78,26 @@ enum {
#define BYT_RT5651_SSP2_AIF2 BIT(19) /* default is using AIF1 */
#define BYT_RT5651_SSP0_AIF1 BIT(20)
#define BYT_RT5651_SSP0_AIF2 BIT(21)
+#define BYT_RT5651_HP_LR_SWAPPED BIT(22)
+#define BYT_RT5651_MONO_SPEAKER BIT(23)
+
+#define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \
+ BYT_RT5651_JD1_1 | \
+ BYT_RT5651_OVCD_TH_2000UA | \
+ BYT_RT5651_OVCD_SF_0P75)
/* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */
#define MAX_NO_PROPS 5
struct byt_rt5651_private {
struct clk *mclk;
+ struct gpio_desc *ext_amp_gpio;
struct snd_soc_jack jack;
};
/* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */
-static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN |
- BYT_RT5651_JD1_1 |
- BYT_RT5651_OVCD_TH_2000UA |
- BYT_RT5651_OVCD_SF_0P75 |
- BYT_RT5651_IN2_HS_IN3_MAP;
+static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP;
static void log_quirks(struct device *dev)
{
@@ -100,10 +107,8 @@ static void log_quirks(struct device *dev)
dev_info(dev, "quirk IN1_MAP enabled");
if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP)
dev_info(dev, "quirk IN2_MAP enabled");
- if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP)
- dev_info(dev, "quirk IN1_HS_IN3_MAP enabled");
- if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP)
- dev_info(dev, "quirk IN2_HS_IN3_MAP enabled");
+ if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP)
+ dev_info(dev, "quirk IN1_IN2_MAP enabled");
if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
dev_info(dev, "quirk realtek,jack-detect-source %ld\n",
BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -124,6 +129,8 @@ static void log_quirks(struct device *dev)
dev_info(dev, "quirk SSP0_AIF1 enabled\n");
if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)
dev_info(dev, "quirk SSP0_AIF2 enabled\n");
+ if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER)
+ dev_info(dev, "quirk MONO_SPEAKER enabled\n");
}
#define BYT_CODEC_DAI1 "rt5651-aif1"
@@ -211,6 +218,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_card *card = w->dapm->card;
+ struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpiod_set_value_cansleep(priv->ext_amp_gpio, 1);
+ else
+ gpiod_set_value_cansleep(priv->ext_amp_gpio, 0);
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
@@ -220,7 +241,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_PRE_PMU |
SND_SOC_DAPM_POST_PMD),
-
+ SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0,
+ rt5651_ext_amp_power_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
};
static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
@@ -228,6 +251,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
{"Headset Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "Platform Clock"},
{"Speaker", NULL, "Platform Clock"},
+ {"Speaker", NULL, "Ext Amp Power"},
{"Line In", NULL, "Platform Clock"},
{"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */
@@ -241,38 +265,26 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = {
- {"IN2P", NULL, "Headset Mic"},
{"DMIC L1", NULL, "Internal Mic"},
{"DMIC R1", NULL, "Internal Mic"},
+ {"IN3P", NULL, "Headset Mic"},
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = {
{"Internal Mic", NULL, "micbias1"},
{"IN1P", NULL, "Internal Mic"},
- {"IN2P", NULL, "Headset Mic"},
+ {"IN3P", NULL, "Headset Mic"},
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = {
{"Internal Mic", NULL, "micbias1"},
- {"IN1P", NULL, "Headset Mic"},
- {"IN2P", NULL, "Internal Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
- {"Internal Mic", NULL, "micbias1"},
- {"IN1P", NULL, "Internal Mic"},
{"IN2P", NULL, "Internal Mic"},
{"IN3P", NULL, "Headset Mic"},
};
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = {
+static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
{"Internal Mic", NULL, "micbias1"},
{"IN1P", NULL, "Internal Mic"},
- {"IN3P", NULL, "Headset Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = {
- {"Internal Mic", NULL, "micbias1"},
{"IN2P", NULL, "Internal Mic"},
{"IN3P", NULL, "Headset Mic"},
};
@@ -357,46 +369,72 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id)
static const struct dmi_system_id byt_rt5651_quirk_table[] = {
{
+ /* Chuwi Hi8 Pro (CWI513) */
.callback = byt_rt5651_quirk_cb,
.matches = {
- DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
- DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+ DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"),
},
- .driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP),
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP |
+ BYT_RT5651_HP_LR_SWAPPED |
+ BYT_RT5651_MONO_SPEAKER),
},
{
+ /* Chuwi Vi8 Plus (CWI519) */
.callback = byt_rt5651_quirk_cb,
.matches = {
- DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
- DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
+ DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
},
- .driver_data = (void *)(BYT_RT5651_MCLK_EN |
- BYT_RT5651_IN1_HS_IN3_MAP),
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP |
+ BYT_RT5651_HP_LR_SWAPPED |
+ BYT_RT5651_MONO_SPEAKER),
+ },
+ {
+ /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6
+ * (these all use the same mainboard) */
+ .callback = byt_rt5651_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+ /* Partial match for all of itWORKS.G.WI71C.JGBMRBA,
+ * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */
+ DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."),
+ },
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP |
+ BYT_RT5651_SSP0_AIF1 |
+ BYT_RT5651_MONO_SPEAKER),
},
{
+ /* KIANO SlimNote 14.2 */
.callback = byt_rt5651_quirk_cb,
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "KIANO"),
DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"),
},
- .driver_data = (void *)(BYT_RT5651_MCLK_EN |
- BYT_RT5651_JD1_1 |
- BYT_RT5651_OVCD_TH_2000UA |
- BYT_RT5651_OVCD_SF_0P75 |
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
BYT_RT5651_IN1_IN2_MAP),
},
{
- /* Chuwi Vi8 Plus (CWI519) */
+ /* Minnowboard Max B3 */
.callback = byt_rt5651_quirk_cb,
.matches = {
- DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
- DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
+ DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+ },
+ .driver_data = (void *)(BYT_RT5651_IN1_MAP),
+ },
+ {
+ /* Minnowboard Turbot */
+ .callback = byt_rt5651_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
},
.driver_data = (void *)(BYT_RT5651_MCLK_EN |
- BYT_RT5651_JD1_1 |
- BYT_RT5651_OVCD_TH_2000UA |
- BYT_RT5651_OVCD_SF_0P75 |
- BYT_RT5651_IN2_HS_IN3_MAP),
+ BYT_RT5651_IN1_MAP),
},
{
/* VIOS LTH17 */
@@ -405,11 +443,24 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "VIOS"),
DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"),
},
- .driver_data = (void *)(BYT_RT5651_MCLK_EN |
+ .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP |
BYT_RT5651_JD1_1 |
BYT_RT5651_OVCD_TH_2000UA |
BYT_RT5651_OVCD_SF_1P0 |
- BYT_RT5651_IN1_IN2_MAP),
+ BYT_RT5651_MCLK_EN),
+ },
+ {
+ /* Yours Y8W81 (and others using the same mainboard) */
+ .callback = byt_rt5651_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+ /* Partial match for all devs with a W86C mainboard */
+ DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."),
+ },
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP |
+ BYT_RT5651_SSP0_AIF1 |
+ BYT_RT5651_MONO_SPEAKER),
},
{}
};
@@ -418,15 +469,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
* Note this MUST be called before snd_soc_register_card(), so that the props
* are in place before the codec component driver's probe function parses them.
*/
-static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
+static int byt_rt5651_add_codec_device_props(struct device *i2c_dev)
{
struct property_entry props[MAX_NO_PROPS] = {};
- struct device *i2c_dev;
- int ret, cnt = 0;
-
- i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name);
- if (!i2c_dev)
- return -EPROBE_DEFER;
+ int cnt = 0;
props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source",
BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -440,10 +486,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN)
props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en");
- ret = device_add_properties(i2c_dev, props);
- put_device(i2c_dev);
-
- return ret;
+ return device_add_properties(i2c_dev, props);
}
static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
@@ -475,14 +518,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
custom_map = byt_rt5651_intmic_in1_in2_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map);
break;
- case BYT_RT5651_IN1_HS_IN3_MAP:
- custom_map = byt_rt5651_intmic_in1_hs_in3_map;
- num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map);
- break;
- case BYT_RT5651_IN2_HS_IN3_MAP:
- custom_map = byt_rt5651_intmic_in2_hs_in3_map;
- num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map);
- break;
default:
custom_map = byt_rt5651_intmic_dmic_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map);
@@ -546,13 +581,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
ret = snd_soc_card_jack_new(runtime->card, "Headset",
- SND_JACK_HEADSET, &priv->jack,
- bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins));
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &priv->jack, bytcr_jack_pins,
+ ARRAY_SIZE(bytcr_jack_pins));
if (ret) {
dev_err(runtime->dev, "jack creation failed %d\n", ret);
return ret;
}
+ snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0,
+ KEY_PLAYPAUSE);
+
ret = snd_soc_component_set_jack(codec, &priv->jack, NULL);
if (ret)
return ret;
@@ -691,6 +730,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
};
/* SoC card */
+static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
+static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */
+static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */
+static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */
+
+static int byt_rt5651_suspend(struct snd_soc_card *card)
+{
+ struct snd_soc_component *component;
+
+ if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+ return 0;
+
+ list_for_each_entry(component, &card->component_dev_list, card_list) {
+ 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);
+ break;
+ }
+ }
+
+ return 0;
+}
+
+static int byt_rt5651_resume(struct snd_soc_card *card)
+{
+ struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component;
+
+ if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+ return 0;
+
+ list_for_each_entry(component, &card->component_dev_list, card_list) {
+ 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);
+ break;
+ }
+ }
+
+ return 0;
+}
+
static struct snd_soc_card byt_rt5651_card = {
.name = "bytcr-rt5651",
.owner = THIS_MODULE,
@@ -701,23 +782,86 @@ static struct snd_soc_card byt_rt5651_card = {
.dapm_routes = byt_rt5651_audio_map,
.num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map),
.fully_routed = true,
+ .suspend_pre = byt_rt5651_suspend,
+ .resume_post = byt_rt5651_resume,
};
-static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
-static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */
-static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */
-static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */
+static const struct x86_cpu_id baytrail_cpu_ids[] = {
+ { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */
+ {}
+};
+
+static const struct x86_cpu_id cherrytrail_cpu_ids[] = {
+ { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */
+ {}
+};
+
+static const struct acpi_gpio_params first_gpio = { 0, 0, false };
+static const struct acpi_gpio_params second_gpio = { 1, 0, false };
+
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = {
+ { "ext-amp-enable-gpios", &first_gpio, 1 },
+ { },
+};
-static bool is_valleyview(void)
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = {
+ { "ext-amp-enable-gpios", &second_gpio, 1 },
+ { },
+};
+
+/*
+ * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
+ * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
+ * GpioIo one for the ext-amp-enable-gpio and both count for the index in
+ * acpi_gpio_params index. So we have 2 different mappings and the code
+ * below figures out which one to use.
+ */
+struct byt_rt5651_acpi_resource_data {
+ int gpio_count;
+ int gpio_int_idx;
+};
+
+static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg)
{
- static const struct x86_cpu_id cpu_ids[] = {
- { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */
- {}
- };
-
- if (!x86_match_cpu(cpu_ids))
- return false;
- return true;
+ struct byt_rt5651_acpi_resource_data *data = arg;
+
+ if (ares->type != ACPI_RESOURCE_TYPE_GPIO)
+ return 0;
+
+ if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT)
+ data->gpio_int_idx = data->gpio_count;
+
+ data->gpio_count++;
+ return 0;
+}
+
+static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec)
+{
+ struct byt_rt5651_acpi_resource_data data = { 0, -1 };
+ LIST_HEAD(resources);
+ int ret;
+
+ ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources,
+ snd_byt_rt5651_acpi_resource, &data);
+ if (ret < 0) {
+ dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n");
+ return;
+ }
+
+ /* All info we need is gathered during the walk */
+ acpi_dev_free_resource_list(&resources);
+
+ switch (data.gpio_int_idx) {
+ case 0:
+ devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second);
+ break;
+ case 1:
+ devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first);
+ break;
+ default:
+ dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n",
+ data.gpio_int_idx);
+ }
}
struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */
@@ -727,13 +871,12 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */
static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
{
- const char * const intmic_name[] =
- { "dmic", "in1", "in2", "in12", "in1", "in2" };
- const char * const hsmic_name[] =
- { "in2", "in2", "in1", "in3", "in3", "in3" };
+ const char * const mic_name[] = { "dmic", "in1", "in2", "in12" };
struct byt_rt5651_private *priv;
struct snd_soc_acpi_mach *mach;
+ struct device *codec_dev;
const char *i2c_name = NULL;
+ const char *hp_swapped;
bool is_bytcr = false;
int ret_val = 0;
int dai_index = 0;
@@ -767,11 +910,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
"%s%s", "i2c-", i2c_name);
byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name;
+ codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL,
+ byt_rt5651_codec_name);
+ if (!codec_dev)
+ return -EPROBE_DEFER;
+
/*
* swap SSP0 if bytcr is detected
* (will be overridden if DMI quirk is detected)
*/
- if (is_valleyview()) {
+ if (x86_match_cpu(baytrail_cpu_ids)) {
struct sst_platform_info *p_info = mach->pdata;
const struct sst_res_info *res_info = p_info->res_info;
@@ -830,9 +978,37 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
dmi_check_system(byt_rt5651_quirk_table);
/* Must be called before register_card, also see declaration comment. */
- ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name);
- if (ret_val)
+ ret_val = byt_rt5651_add_codec_device_props(codec_dev);
+ if (ret_val) {
+ put_device(codec_dev);
return ret_val;
+ }
+
+ /* Cherry Trail devices use an external amplifier enable gpio */
+ if (x86_match_cpu(cherrytrail_cpu_ids)) {
+ snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev);
+ priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child(
+ &pdev->dev, "ext-amp-enable", 0,
+ codec_dev->fwnode,
+ GPIOD_OUT_LOW, "speaker-amp");
+ if (IS_ERR(priv->ext_amp_gpio)) {
+ ret_val = PTR_ERR(priv->ext_amp_gpio);
+ switch (ret_val) {
+ case -ENOENT:
+ priv->ext_amp_gpio = NULL;
+ break;
+ default:
+ dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
+ ret_val);
+ /* fall through */
+ case -EPROBE_DEFER:
+ put_device(codec_dev);
+ return ret_val;
+ }
+ }
+ }
+
+ put_device(codec_dev);
log_quirks(&pdev->dev);
@@ -876,10 +1052,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
}
}
+ if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED)
+ hp_swapped = "-hp-swapped";
+ else
+ hp_swapped = "";
+
snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name),
- "bytcr-rt5651-%s-intmic-%s-hsmic",
- intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
- hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]);
+ "bytcr-rt5651-%s-spk-%s-mic%s",
+ (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ?
+ "mono" : "stereo",
+ mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped);
byt_rt5651_card.long_name = byt_rt5651_long_name;
ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card);
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
new file mode 100644
index 000000000000..c4b94e2617c5
--- /dev/null
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -0,0 +1,643 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs
+ *
+ * Modified from:
+ * Intel Apollolake I2S Machine driver
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../skylake/skl.h"
+#include "../../codecs/rt5682.h"
+#include "../../codecs/hdac_hdmi.h"
+
+/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */
+#define GLK_PLAT_CLK_FREQ 19200000
+#define RT5682_PLL_FREQ (48000 * 512)
+#define GLK_REALTEK_CODEC_DAI "rt5682-aif1"
+#define GLK_MAXIM_CODEC_DAI "HiFi"
+#define MAXIM_DEV0_NAME "MX98357A:00"
+#define DUAL_CHANNEL 2
+#define QUAD_CHANNEL 4
+#define NAME_SIZE 32
+
+static struct snd_soc_jack geminilake_hdmi[3];
+
+struct glk_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct glk_card_private {
+ struct snd_soc_jack geminilake_headset;
+ struct list_head hdmi_pcm_list;
+};
+
+enum {
+ GLK_DPCM_AUDIO_PB = 0,
+ GLK_DPCM_AUDIO_CP,
+ GLK_DPCM_AUDIO_HS_PB,
+ GLK_DPCM_AUDIO_ECHO_REF_CP,
+ GLK_DPCM_AUDIO_REF_CP,
+ GLK_DPCM_AUDIO_DMIC_CP,
+ GLK_DPCM_AUDIO_HDMI1_PB,
+ GLK_DPCM_AUDIO_HDMI2_PB,
+ GLK_DPCM_AUDIO_HDMI3_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, GLK_REALTEK_CODEC_DAI);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+ return -EIO;
+ }
+
+ if (SND_SOC_DAPM_EVENT_OFF(event)) {
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+ if (ret)
+ dev_err(card->dev, "failed to stop sysclk: %d\n", ret);
+ } else if (SND_SOC_DAPM_EVENT_ON(event)) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
+ GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
+ if (ret < 0) {
+ dev_err(card->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+ }
+
+ if (ret)
+ dev_err(card->dev, "failed to start internal clk: %d\n", ret);
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new geminilake_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+static const struct snd_soc_dapm_widget geminilake_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_SPK("Spk", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+ SND_SOC_DAPM_SPK("HDMI1", NULL),
+ SND_SOC_DAPM_SPK("HDMI2", NULL),
+ SND_SOC_DAPM_SPK("HDMI3", 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 geminilake_map[] = {
+ /* HP jack connectors - unknown if we have jack detection */
+ { "Headphone Jack", NULL, "Platform Clock" },
+ { "Headphone Jack", NULL, "HPOL" },
+ { "Headphone Jack", NULL, "HPOR" },
+
+ /* speaker */
+ { "Spk", NULL, "Speaker" },
+
+ /* other jacks */
+ { "Headset Mic", NULL, "Platform Clock" },
+ { "IN1P", NULL, "Headset Mic" },
+
+ /* digital mics */
+ { "DMic", NULL, "SoC DMIC" },
+
+ /* CODEC BE connections */
+ { "HiFi Playback", NULL, "ssp1 Tx" },
+ { "ssp1 Tx", NULL, "codec0_out" },
+
+ { "AIF1 Playback", NULL, "ssp2 Tx" },
+ { "ssp2 Tx", NULL, "codec1_out" },
+
+ { "codec0_in", NULL, "ssp2 Rx" },
+ { "ssp2 Rx", NULL, "AIF1 Capture" },
+
+ { "HDMI1", NULL, "hif5-0 Output" },
+ { "HDMI2", NULL, "hif6-0 Output" },
+ { "HDMI2", NULL, "hif7-0 Output" },
+
+ { "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" },
+
+ /* DMIC */
+ { "dmic01_hifi", NULL, "DMIC01 Rx" },
+ { "DMIC01 Rx", NULL, "DMIC AIF" },
+};
+
+static int geminilake_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);
+
+ /* The ADSP will convert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = DUAL_CHANNEL;
+
+ /* set SSP to 24 bit */
+ snd_mask_none(fmt);
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+ return 0;
+}
+
+static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_jack *jack;
+ int ret;
+
+ /* Configure sysclk for codec */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
+ RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
+
+ /*
+ * Headset buttons map to the google Reference headset.
+ * These can be configured by userspace.
+ */
+ ret = snd_soc_card_jack_new(rtd->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->geminilake_headset, NULL, 0);
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ jack = &ctx->geminilake_headset;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+ ret = snd_soc_component_set_jack(component, jack, NULL);
+
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+};
+
+static int geminilake_rt5682_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 = rtd->codec_dai;
+ int ret;
+
+ /* Set valid bitmask & configuration for I2S in 24 bit */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24);
+ if (ret < 0) {
+ dev_err(rtd->dev, "set TDM slot err:%d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops geminilake_rt5682_ops = {
+ .hw_params = geminilake_rt5682_hw_params,
+};
+
+static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *dai = rtd->codec_dai;
+ struct glk_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
+}
+
+static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_dapm_context *dapm;
+ int ret;
+
+ dapm = snd_soc_component_get_dapm(component);
+ ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+ if (ret) {
+ dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+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 geminilake_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
+ */
+ channels->min = channels->max = 4;
+
+ return 0;
+}
+
+static int geminilake_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 const struct snd_soc_ops geminilake_dmic_ops = {
+ .startup = geminilake_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 int geminilake_refcap_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_16000);
+};
+
+static const struct snd_soc_ops geminilake_refcap_ops = {
+ .startup = geminilake_refcap_startup,
+};
+
+/* geminilake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link geminilake_dais[] = {
+ /* Front End DAI links */
+ [GLK_DPCM_AUDIO_PB] = {
+ .name = "Glk Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .init = geminilake_rt5682_fe_init,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ [GLK_DPCM_AUDIO_CP] = {
+ .name = "Glk Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .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,
+ },
+ [GLK_DPCM_AUDIO_HS_PB] = {
+ .name = "Glk Audio Headset Playback",
+ .stream_name = "Headset Audio",
+ .cpu_dai_name = "System Pin2",
+ .platform_name = "0000:00:0e.0",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .dpcm_playback = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [GLK_DPCM_AUDIO_ECHO_REF_CP] = {
+ .name = "Glk Audio Echo Reference cap",
+ .stream_name = "Echoreference Capture",
+ .cpu_dai_name = "Echoref Pin",
+ .platform_name = "0000:00:0e.0",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = NULL,
+ .capture_only = 1,
+ .nonatomic = 1,
+ },
+ [GLK_DPCM_AUDIO_REF_CP] = {
+ .name = "Glk Audio Reference cap",
+ .stream_name = "Refcap",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &geminilake_refcap_ops,
+ },
+ [GLK_DPCM_AUDIO_DMIC_CP] = {
+ .name = "Glk 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 = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &geminilake_dmic_ops,
+ },
+ [GLK_DPCM_AUDIO_HDMI1_PB] = {
+ .name = "Glk HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [GLK_DPCM_AUDIO_HDMI2_PB] = {
+ .name = "Glk HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [GLK_DPCM_AUDIO_HDMI3_PB] = {
+ .name = "Glk HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ /* Back End DAI links */
+ {
+ /* SSP1 - Codec */
+ .name = "SSP1-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP1 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = MAXIM_DEV0_NAME,
+ .codec_dai_name = GLK_MAXIM_CODEC_DAI,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = geminilake_ssp_fixup,
+ .dpcm_playback = 1,
+ },
+ {
+ /* SSP2 - Codec */
+ .name = "SSP2-Codec",
+ .id = 1,
+ .cpu_dai_name = "SSP2 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = "i2c-10EC5682:00",
+ .codec_dai_name = GLK_REALTEK_CODEC_DAI,
+ .init = geminilake_rt5682_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 = geminilake_ssp_fixup,
+ .ops = &geminilake_rt5682_ops,
+ .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 = "0000:00:0e.0",
+ .ignore_suspend = 1,
+ .be_hw_params_fixup = geminilake_dmic_fixup,
+ .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 = "0000:00:0e.0",
+ .init = geminilake_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 4,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:0e.0",
+ .init = geminilake_hdmi_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 = "0000:00:0e.0",
+ .init = geminilake_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+static int glk_card_late_probe(struct snd_soc_card *card)
+{
+ struct glk_card_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component = NULL;
+ char jack_name[NAME_SIZE];
+ struct glk_hdmi_pcm *pcm;
+ int err = 0;
+ int i = 0;
+
+ 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, &geminilake_hdmi[i],
+ NULL, 0);
+
+ if (err)
+ return err;
+
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+ &geminilake_hdmi[i]);
+ if (err < 0)
+ return err;
+
+ i++;
+ }
+
+ if (!component)
+ return -EINVAL;
+
+ return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
+
+/* geminilake audio machine driver for SPT + RT5682 */
+static struct snd_soc_card glk_audio_card_rt5682_m98357a = {
+ .name = "glkrt5682max",
+ .owner = THIS_MODULE,
+ .dai_link = geminilake_dais,
+ .num_links = ARRAY_SIZE(geminilake_dais),
+ .controls = geminilake_controls,
+ .num_controls = ARRAY_SIZE(geminilake_controls),
+ .dapm_widgets = geminilake_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets),
+ .dapm_routes = geminilake_map,
+ .num_dapm_routes = ARRAY_SIZE(geminilake_map),
+ .fully_routed = true,
+ .late_probe = glk_card_late_probe,
+};
+
+static int geminilake_audio_probe(struct platform_device *pdev)
+{
+ struct glk_card_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+ glk_audio_card_rt5682_m98357a.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx);
+
+ return devm_snd_soc_register_card(&pdev->dev,
+ &glk_audio_card_rt5682_m98357a);
+}
+
+static const struct platform_device_id glk_board_ids[] = {
+ {
+ .name = "glk_rt5682_max98357a",
+ .driver_data =
+ (kernel_ulong_t)&glk_audio_card_rt5682_m98357a,
+ },
+ { }
+};
+
+static struct platform_driver geminilake_audio = {
+ .probe = geminilake_audio_probe,
+ .driver = {
+ .name = "glk_rt5682_max98357a",
+ .pm = &snd_soc_pm_ops,
+ },
+ .id_table = glk_board_ids,
+};
+module_platform_driver(geminilake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode");
+MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
+MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:glk_rt5682_max98357a");
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index 94294c27d1db..38f6ab74709d 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
@@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
.trigger = {
SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
+ .ops = &kabylake_da7219_fe_ops,
},
[KBL_DPCM_AUDIO_DMIC_CP] = {
.name = "Kbl Audio DMIC cap",
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3a61252fe450..21a6490746a6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(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);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 92f5fb2ae0a3..a892b37eab7c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
if (params_channels(params) == 2 ||
DMIC_CH(dmic_constraints) == 2)
@@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
* thus changing the mask here
*/
if (!strcmp(be_dai_link->name, "SSP0-Codec"))
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 3ff6646cfa21..d31482b8c9bb 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index b0610bba3cfa..e877bb60beb1 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 38a1495c29cf..0e1818dd4cc6 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
- snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 7379d8830c39..915a34cdc8ac 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o
snd-soc-sst-acpi-objs := sst-acpi.o
snd-soc-sst-ipc-objs := sst-ipc.o
snd-soc-sst-firmware-objs := sst-firmware.o
-snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o
+snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \
+ 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
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-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
new file mode 100644
index 000000000000..f39386e540d3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs bxt_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = {
+ {
+ .id = "INT343A",
+ .drv_name = "bxt_alc298s_i2s",
+ .fw_filename = "intel/dsp_fw_bxtn.bin",
+ },
+ {
+ .id = "DLGS7219",
+ .drv_name = "bxt_da7219_max98357a",
+ .fw_filename = "intel/dsp_fw_bxtn.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &bxt_codecs,
+ .sof_fw_filename = "intel/sof-apl.ri",
+ .sof_tplg_filename = "intel/sof-apl-da7219.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {
+ .id = "104C5122",
+ .drv_name = "bxt-pcm512x",
+ .sof_fw_filename = "intel/sof-apl.ri",
+ .sof_tplg_filename = "intel/sof-apl-pcm512x.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {
+ .id = "1AEC8804",
+ .drv_name = "bxt-wm8804",
+ .sof_fw_filename = "intel/sof-apl.ri",
+ .sof_tplg_filename = "intel/sof-apl-wm8804.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {
+ .id = "INT34C3",
+ .drv_name = "bxt_tdf8532",
+ .sof_fw_filename = "intel/sof-apl.ri",
+ .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
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 bfe1ca68a542..4daa8a4f0c0c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = {
.drv_name = "cht-bsw-rt5672",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5670.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5670.tplg",
.asoc_plat_name = "sst-mfld-platform",
};
@@ -98,8 +98,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcr_rt5640",
.machine_quirk = byt_quirk,
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -107,8 +107,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "bytcr_rt5640",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcr_rt5640",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -116,8 +116,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "bytcr_rt5640",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcr_rt5640",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -125,8 +125,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "bytcr_rt5651",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcr_rt5651",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5651.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5651.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -134,8 +134,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "bytcht_da7213",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcht_da7213",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-da7213.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -143,8 +143,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "bytcht_da7213",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "bytcht_da7213",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-da7213.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
/* some Baytrail platforms rely on RT5645, use CHT machine driver */
@@ -153,8 +153,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -162,8 +162,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
/* use CHT driver to Baytrail Chromebooks */
@@ -172,8 +172,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = {
.drv_name = "cht-bsw-max98090",
.fw_filename = "intel/fw_sst_0f28.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-byt.ri",
- .sof_tplg_filename = "intel/reef-byt-max98090.tplg",
+ .sof_fw_filename = "intel/sof-byt.ri",
+ .sof_tplg_filename = "intel/sof-byt-max98090.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
#if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index ad1eb2d644be..91bb99b69601 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
};
@@ -68,8 +68,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-rt5672",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -77,8 +77,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-rt5672",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -86,8 +86,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -95,8 +95,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -104,8 +104,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-rt5645",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -113,8 +113,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-max98090",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-max98090.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-max98090.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -122,8 +122,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "cht-bsw-nau8824",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "cht-bsw",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-nau8824.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-nau8824.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -131,8 +131,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "bytcht_da7213",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcht_da7213",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-da7213.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -140,8 +140,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "bytcht_da7213",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcht_da7213",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-da7213.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -149,8 +149,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "bytcht_es8316",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcht_es8316",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-es8316.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-es8316.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
/* some CHT-T platforms rely on RT5640, use Baytrail machine driver */
@@ -160,8 +160,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcr_rt5640",
.machine_quirk = cht_quirk,
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
{
@@ -169,8 +169,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "bytcr_rt5640",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcr_rt5640",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
/* some CHT-T platforms rely on RT5651, use Baytrail machine driver */
@@ -179,8 +179,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.drv_name = "bytcr_rt5651",
.fw_filename = "intel/fw_sst_22a8.bin",
.board = "bytcr_rt5651",
- .sof_fw_filename = "intel/reef-cht.ri",
- .sof_tplg_filename = "intel/reef-cht-rt5651.tplg",
+ .sof_fw_filename = "intel/sof-cht.ri",
+ .sof_tplg_filename = "intel/sof-cht-rt5651.tplg",
.asoc_plat_name = "sst-mfld-platform",
},
#if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
new file mode 100644
index 000000000000..ec8e28e7b937
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -0,0 +1,32 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata cnl_pdata = {
+ .use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = {
+ {
+ .id = "INT34C2",
+ .drv_name = "cnl_rt274",
+ .fw_filename = "intel/dsp_fw_cnl.bin",
+ .pdata = &cnl_pdata,
+ .sof_fw_filename = "intel/sof-cnl.ri",
+ .sof_tplg_filename = "intel/sof-cnl-rt274.tplg",
+ .asoc_plat_name = "0000:00:1f.3",
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
new file mode 100644
index 000000000000..305875af71ca
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -0,0 +1,41 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs glk_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = {
+ {
+ .id = "INT343A",
+ .drv_name = "glk_alc298s_i2s",
+ .fw_filename = "intel/dsp_fw_glk.bin",
+ .sof_fw_filename = "intel/sof-glk.ri",
+ .sof_tplg_filename = "intel/sof-glk-alc298.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {
+ .id = "DLGS7219",
+ .drv_name = "glk_da7219_max98357a",
+ .fw_filename = "intel/dsp_fw_glk.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &glk_codecs,
+ .sof_fw_filename = "intel/sof-glk.ri",
+ .sof_tplg_filename = "intel/sof-glk-da7219.tplg",
+ .asoc_plat_name = "0000:00:0e.0",
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index e0e8c8c27528..494a0ea9b029 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = {
.id = "INT33CA",
.drv_name = "haswell-audio",
.fw_filename = "intel/IntcSST1.bin",
- .sof_fw_filename = "intel/reef-hsw.ri",
- .sof_tplg_filename = "intel/reef-hsw.tplg",
+ .sof_fw_filename = "intel/sof-hsw.ri",
+ .sof_tplg_filename = "intel/sof-hsw.tplg",
.asoc_plat_name = "haswell-pcm-audio",
},
{}
@@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = {
.id = "INT343A",
.drv_name = "broadwell-audio",
.fw_filename = "intel/IntcSST2.bin",
- .sof_fw_filename = "intel/reef-bdw.ri",
- .sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+ .sof_fw_filename = "intel/sof-bdw.ri",
+ .sof_tplg_filename = "intel/sof-bdw-rt286.tplg",
.asoc_plat_name = "haswell-pcm-audio",
},
{
.id = "RT5677CE",
.drv_name = "bdw-rt5677",
.fw_filename = "intel/IntcSST2.bin",
- .sof_fw_filename = "intel/reef-bdw.ri",
- .sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+ .sof_fw_filename = "intel/sof-bdw.ri",
+ .sof_tplg_filename = "intel/sof-bdw-rt5677.tplg",
.asoc_plat_name = "haswell-pcm-audio",
},
{
.id = "INT33CA",
.drv_name = "haswell-audio",
.fw_filename = "intel/IntcSST2.bin",
- .sof_fw_filename = "intel/reef-bdw.ri",
- .sof_tplg_filename = "intel/reef-bdw-rt5640.tplg",
+ .sof_fw_filename = "intel/sof-bdw.ri",
+ .sof_tplg_filename = "intel/sof-bdw-rt5640.tplg",
.asoc_plat_name = "haswell-pcm-audio",
},
{}
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
new file mode 100644
index 000000000000..0ee173ca437d
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -0,0 +1,91 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs kbl_codecs = {
+ .num_codecs = 1,
+ .codecs = {"10508825"}
+};
+
+static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
+ .num_codecs = 1,
+ .codecs = {"10EC5663"}
+};
+
+static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
+ .num_codecs = 2,
+ .codecs = {"10EC5663", "10EC5514"}
+};
+
+static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
+ {
+ .id = "INT343A",
+ .drv_name = "kbl_alc286s_i2s",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ },
+ {
+ .id = "INT343B",
+ .drv_name = "kbl_n88l25_s4567",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {
+ .id = "MX98357A",
+ .drv_name = "kbl_n88l25_m98357a",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {
+ .id = "MX98927",
+ .drv_name = "kbl_r5514_5663_max",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_5663_5514_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {
+ .id = "MX98927",
+ .drv_name = "kbl_rt5663_m98927",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_poppy_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {
+ .id = "10EC5663",
+ .drv_name = "kbl_rt5663",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ },
+ {
+ .id = "DLGS7219",
+ .drv_name = "kbl_da7219_max98357a",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_7219_98357_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
new file mode 100644
index 000000000000..0c9c0edd35b3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -0,0 +1,47 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs skl_codecs = {
+ .num_codecs = 1,
+ .codecs = {"10508825"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = {
+ {
+ .id = "INT343A",
+ .drv_name = "skl_alc286s_i2s",
+ .fw_filename = "intel/dsp_fw_release.bin",
+ },
+ {
+ .id = "INT343B",
+ .drv_name = "skl_n88l25_s4567",
+ .fw_filename = "intel/dsp_fw_release.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &skl_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {
+ .id = "MX98357A",
+ .drv_name = "skl_n88l25_m98357a",
+ .fw_filename = "intel/dsp_fw_release.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &skl_codecs,
+ .pdata = &skl_dmic_data,
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 657afc02f1c4..11041aedea31 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp)
}
EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel);
-int sst_dma_new(struct sst_dsp *sst)
+static int sst_dma_new(struct sst_dsp *sst)
{
struct sst_pdata *sst_pdata = sst->pdata;
struct sst_dma *dma;
@@ -320,9 +320,8 @@ err_dma_dev:
devm_kfree(sst->dev, dma);
return ret;
}
-EXPORT_SYMBOL(sst_dma_new);
-void sst_dma_free(struct sst_dma *dma)
+static void sst_dma_free(struct sst_dma *dma)
{
if (dma == NULL)
@@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma)
dw_remove(dma->chip);
}
-EXPORT_SYMBOL(sst_dma_free);
/* create new generic firmware object */
struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c
index b2bec36d074c..a28220e67cdf 100644
--- a/sound/soc/intel/haswell/sst-haswell-dsp.c
+++ b/sound/soc/intel/haswell/sst-haswell-dsp.c
@@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
struct sst_module_template template;
int count, ret;
void __iomem *ram;
+ int type = le16_to_cpu(module->type);
+ int entry_point = le32_to_cpu(module->entry_point);
/* TODO: allowed module types need to be configurable */
- if (module->type != SST_HSW_MODULE_BASE_FW
- && module->type != SST_HSW_MODULE_PCM_SYSTEM
- && module->type != SST_HSW_MODULE_PCM
- && module->type != SST_HSW_MODULE_PCM_REFERENCE
- && module->type != SST_HSW_MODULE_PCM_CAPTURE
- && module->type != SST_HSW_MODULE_WAVES
- && module->type != SST_HSW_MODULE_LPAL)
+ if (type != SST_HSW_MODULE_BASE_FW &&
+ type != SST_HSW_MODULE_PCM_SYSTEM &&
+ type != SST_HSW_MODULE_PCM &&
+ type != SST_HSW_MODULE_PCM_REFERENCE &&
+ type != SST_HSW_MODULE_PCM_CAPTURE &&
+ type != SST_HSW_MODULE_WAVES &&
+ type != SST_HSW_MODULE_LPAL)
return 0;
dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n",
module->signature, module->mod_size,
- module->blocks, module->type);
- dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point);
+ module->blocks, type);
+ dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point);
dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n",
module->info.persistent_size, module->info.scratch_size);
memset(&template, 0, sizeof(template));
- template.id = module->type;
- template.entry = module->entry_point - 4;
- template.persistent_size = module->info.persistent_size;
- template.scratch_size = module->info.scratch_size;
+ template.id = type;
+ template.entry = entry_point - 4;
+ template.persistent_size = le32_to_cpu(module->info.persistent_size);
+ template.scratch_size = le32_to_cpu(module->info.scratch_size);
mod = sst_module_new(fw, &template, NULL);
if (mod == NULL)
@@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
block = (void *)module + sizeof(*module);
- for (count = 0; count < module->blocks; count++) {
+ for (count = 0; count < le32_to_cpu(module->blocks); count++) {
- if (block->size <= 0) {
+ if (le32_to_cpu(block->size) <= 0) {
dev_err(dsp->dev,
"error: block %d size invalid\n", count);
sst_module_free(mod);
return -EINVAL;
}
- switch (block->type) {
+ switch (le32_to_cpu(block->type)) {
case SST_HSW_IRAM:
ram = dsp->addr.lpe;
- mod->offset =
- block->ram_offset + dsp->addr.iram_offset;
+ mod->offset = le32_to_cpu(block->ram_offset) +
+ dsp->addr.iram_offset;
mod->type = SST_MEM_IRAM;
break;
case SST_HSW_DRAM:
case SST_HSW_REGS:
ram = dsp->addr.lpe;
- mod->offset = block->ram_offset;
+ mod->offset = le32_to_cpu(block->ram_offset);
mod->type = SST_MEM_DRAM;
break;
default:
@@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
return -EINVAL;
}
- mod->size = block->size;
+ mod->size = le32_to_cpu(block->size);
mod->data = (void *)block + sizeof(*block);
mod->data_offset = mod->data - fw->dma_buf;
@@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
return ret;
}
- block = (void *)block + sizeof(*block) + block->size;
+ block = (void *)block + sizeof(*block) +
+ le32_to_cpu(block->size);
}
mod->state = SST_MODULE_STATE_LOADED;
@@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
/* verify FW */
if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) ||
- (sst_fw->size != header->file_size + sizeof(*header))) {
+ (sst_fw->size !=
+ le32_to_cpu(header->file_size) + sizeof(*header))) {
dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n");
return -EINVAL;
}
@@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
/* parse each module */
module = (void *)sst_fw->dma_buf + sizeof(*header);
- for (count = 0; count < header->modules; count++) {
+ for (count = 0; count < le32_to_cpu(header->modules); count++) {
/* module */
ret = hsw_parse_module(dsp, sst_fw, module);
@@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
dev_err(dsp->dev, "error: invalid module %d\n", count);
return ret;
}
- module = (void *)module + sizeof(*module) + module->mod_size;
+ module = (void *)module + sizeof(*module) +
+ le32_to_cpu(module->mod_size);
}
return 0;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index d7fc3b2d3e68..823e39103edd 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1014,10 +1014,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
};
-int skl_dai_load(struct snd_soc_component *cmp,
- struct snd_soc_dai_driver *pcm_dai)
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+ struct snd_soc_dai_driver *dai_drv,
+ struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
{
- pcm_dai->ops = &skl_pcm_dai_ops;
+ dai_drv->ops = &skl_pcm_dai_ops;
return 0;
}
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c
index d2b1d60fec02..5bc0d38da7e3 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.c
+++ b/sound/soc/intel/skylake/skl-sst-cldma.c
@@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp *ctx)
/* Code loader helper APIs */
static void skl_cldma_setup_bdle(struct sst_dsp *ctx,
struct snd_dma_buffer *dmab_data,
- u32 **bdlp, int size, int with_ioc)
+ __le32 **bdlp, int size, int with_ioc)
{
- u32 *bdl = *bdlp;
+ __le32 *bdl = *bdlp;
ctx->cl_dev.frags = 0;
while (size > 0) {
@@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx)
int skl_cldma_prepare(struct sst_dsp *ctx)
{
int ret;
- u32 *bdl;
+ __le32 *bdl;
ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE;
@@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx)
ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data);
return ret;
}
- bdl = (u32 *)ctx->cl_dev.dmab_bdl.area;
+ bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area;
/* Allocate BDLs */
ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data,
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h
index 5b730a1a0ae4..ec736921a083 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.h
+++ b/sound/soc/intel/skylake/skl-sst-cldma.h
@@ -203,7 +203,7 @@ struct sst_dsp;
struct skl_cl_dev_ops {
void (*cl_setup_bdle)(struct sst_dsp *ctx,
struct snd_dma_buffer *dmab_data,
- u32 **bdlp, int size, int with_ioc);
+ __le32 **bdlp, int size, int with_ioc);
void (*cl_setup_controller)(struct sst_dsp *ctx,
struct snd_dma_buffer *dmab_bdl,
unsigned int max_size, u32 page_count);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index abfdb67c05cc..2620d77729c5 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w,
case snd_soc_dapm_aif_out:
case snd_soc_dapm_dai_out:
case snd_soc_dapm_switch:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_mux:
+
return false;
default:
return true;
@@ -3024,7 +3027,7 @@ void skl_cleanup_resources(struct skl *skl)
* information to the driver about module and pipeline parameters which DSP
* FW expects like ids, resource values, formats etc
*/
-static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
+static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index,
struct snd_soc_dapm_widget *w,
struct snd_soc_tplg_dapm_widget *tplg_w)
{
@@ -3130,6 +3133,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se,
}
static int skl_tplg_control_load(struct snd_soc_component *cmpnt,
+ int index,
struct snd_kcontrol_new *kctl,
struct snd_soc_tplg_ctl_hdr *hdr)
{
@@ -3617,7 +3621,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
return 0;
}
-static int skl_manifest_load(struct snd_soc_component *cmpnt,
+static int skl_manifest_load(struct snd_soc_component *cmpnt, int index,
struct snd_soc_tplg_manifest *manifest)
{
struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index daeb6d2bb7fc..82282cac9751 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev,
int skl_pcm_link_dma_prepare(struct device *dev,
struct skl_pipe_params *params);
-int skl_dai_load(struct snd_soc_component *cmp,
- struct snd_soc_dai_driver *pcm_dai);
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+ struct snd_soc_dai_driver *dai_drv,
+ struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
struct snd_soc_dapm_widget *w);
#endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 00e051467a40..dce649485649 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -29,6 +29,7 @@
#include <linux/delay.h>
#include <sound/pcm.h>
#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
#include <sound/hda_register.h>
#include <sound/hdaudio.h>
#include <sound/hda_i915.h>
@@ -36,8 +37,6 @@
#include "skl-sst-dsp.h"
#include "skl-sst-ipc.h"
-static struct skl_machine_pdata skl_dmic_data;
-
/*
* initialize the PCI registers
*/
@@ -487,10 +486,12 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
skl->mach = mach;
skl->fw_name = mach->fw_filename;
- pdata = skl->mach->pdata;
+ pdata = mach->pdata;
- if (mach->pdata)
+ if (pdata) {
skl->use_tplg_pcm = pdata->use_tplg_pcm;
+ pdata->dmic_num = skl_get_dmic_geo(skl);
+ }
return 0;
}
@@ -917,8 +918,6 @@ static int skl_probe(struct pci_dev *pci,
pci_set_drvdata(skl->pci, bus);
- skl_dmic_data.dmic_num = skl_get_dmic_geo(skl);
-
/* check if dsp is there */
if (bus->ppcap) {
/* create device for dsp clk */
@@ -1012,172 +1011,23 @@ static void skl_remove(struct pci_dev *pci)
dev_set_drvdata(&pci->dev, NULL);
}
-static struct snd_soc_acpi_codecs skl_codecs = {
- .num_codecs = 1,
- .codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs kbl_codecs = {
- .num_codecs = 1,
- .codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs bxt_codecs = {
- .num_codecs = 1,
- .codecs = {"MX98357A"}
-};
-
-static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
- .num_codecs = 1,
- .codecs = {"10EC5663"}
-};
-
-static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
- .num_codecs = 2,
- .codecs = {"10EC5663", "10EC5514"}
-};
-
-static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
- .num_codecs = 1,
- .codecs = {"MX98357A"}
-};
-
-static struct skl_machine_pdata cnl_pdata = {
- .use_tplg_pcm = true,
-};
-
-static struct snd_soc_acpi_mach sst_skl_devdata[] = {
- {
- .id = "INT343A",
- .drv_name = "skl_alc286s_i2s",
- .fw_filename = "intel/dsp_fw_release.bin",
- },
- {
- .id = "INT343B",
- .drv_name = "skl_n88l25_s4567",
- .fw_filename = "intel/dsp_fw_release.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &skl_codecs,
- .pdata = &skl_dmic_data
- },
- {
- .id = "MX98357A",
- .drv_name = "skl_n88l25_m98357a",
- .fw_filename = "intel/dsp_fw_release.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &skl_codecs,
- .pdata = &skl_dmic_data
- },
- {}
-};
-
-static struct snd_soc_acpi_mach sst_bxtp_devdata[] = {
- {
- .id = "INT343A",
- .drv_name = "bxt_alc298s_i2s",
- .fw_filename = "intel/dsp_fw_bxtn.bin",
- },
- {
- .id = "DLGS7219",
- .drv_name = "bxt_da7219_max98357a_i2s",
- .fw_filename = "intel/dsp_fw_bxtn.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &bxt_codecs,
- },
- {}
-};
-
-static struct snd_soc_acpi_mach sst_kbl_devdata[] = {
- {
- .id = "INT343A",
- .drv_name = "kbl_alc286s_i2s",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- },
- {
- .id = "INT343B",
- .drv_name = "kbl_n88l25_s4567",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &kbl_codecs,
- .pdata = &skl_dmic_data
- },
- {
- .id = "MX98357A",
- .drv_name = "kbl_n88l25_m98357a",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &kbl_codecs,
- .pdata = &skl_dmic_data
- },
- {
- .id = "MX98927",
- .drv_name = "kbl_r5514_5663_max",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &kbl_5663_5514_codecs,
- .pdata = &skl_dmic_data
- },
- {
- .id = "MX98927",
- .drv_name = "kbl_rt5663_m98927",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &kbl_poppy_codecs,
- .pdata = &skl_dmic_data
- },
- {
- .id = "10EC5663",
- .drv_name = "kbl_rt5663",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- },
- {
- .id = "DLGS7219",
- .drv_name = "kbl_da7219_max98357a",
- .fw_filename = "intel/dsp_fw_kbl.bin",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &kbl_7219_98357_codecs,
- .pdata = &skl_dmic_data
- },
-
- {}
-};
-
-static struct snd_soc_acpi_mach sst_glk_devdata[] = {
- {
- .id = "INT343A",
- .drv_name = "glk_alc298s_i2s",
- .fw_filename = "intel/dsp_fw_glk.bin",
- },
- {}
-};
-
-static const struct snd_soc_acpi_mach sst_cnl_devdata[] = {
- {
- .id = "INT34C2",
- .drv_name = "cnl_rt274",
- .fw_filename = "intel/dsp_fw_cnl.bin",
- .pdata = &cnl_pdata,
- },
- {}
-};
-
/* PCI IDs */
static const struct pci_device_id skl_ids[] = {
/* Sunrise Point-LP */
{ PCI_DEVICE(0x8086, 0x9d70),
- .driver_data = (unsigned long)&sst_skl_devdata},
+ .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines},
/* BXT-P */
{ PCI_DEVICE(0x8086, 0x5a98),
- .driver_data = (unsigned long)&sst_bxtp_devdata},
+ .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines},
/* KBL */
{ PCI_DEVICE(0x8086, 0x9D71),
- .driver_data = (unsigned long)&sst_kbl_devdata},
+ .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines},
/* GLK */
{ PCI_DEVICE(0x8086, 0x3198),
- .driver_data = (unsigned long)&sst_glk_devdata},
+ .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines},
/* CNL */
{ PCI_DEVICE(0x8086, 0x9dc8),
- .driver_data = (unsigned long)&sst_cnl_devdata},
+ .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines},
{ 0, }
};
MODULE_DEVICE_TABLE(pci, skl_ids);
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 51ec4ff6ed95..697aa50aff9a 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -15,20 +15,12 @@
int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
{
- struct snd_soc_dai_driver *sub_dai_drivers;
+ struct mtk_base_afe_dai *dai;
size_t num_dai_drivers = 0, dai_idx = 0;
- int i;
-
- if (!afe->sub_dais) {
- dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
- return -EINVAL;
- }
/* calcualte total dai driver size */
- for (i = 0; i < afe->num_sub_dais; i++) {
- if (afe->sub_dais[i].dai_drivers &&
- afe->sub_dais[i].num_dai_drivers != 0)
- num_dai_drivers += afe->sub_dais[i].num_dai_drivers;
+ list_for_each_entry(dai, &afe->sub_dais, list) {
+ num_dai_drivers += dai->num_dai_drivers;
}
dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers);
@@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
if (!afe->dai_drivers)
return -ENOMEM;
- for (i = 0; i < afe->num_sub_dais; i++) {
- if (afe->sub_dais[i].dai_drivers &&
- afe->sub_dais[i].num_dai_drivers != 0) {
- sub_dai_drivers = afe->sub_dais[i].dai_drivers;
- /* dai driver */
- memcpy(&afe->dai_drivers[dai_idx],
- sub_dai_drivers,
- afe->sub_dais[i].num_dai_drivers *
- sizeof(struct snd_soc_dai_driver));
- dai_idx += afe->sub_dais[i].num_dai_drivers;
- }
+ list_for_each_entry(dai, &afe->sub_dais, list) {
+ /* dai driver */
+ memcpy(&afe->dai_drivers[dai_idx],
+ dai->dai_drivers,
+ dai->num_dai_drivers *
+ sizeof(struct snd_soc_dai_driver));
+ dai_idx += dai->num_dai_drivers;
}
-
return 0;
}
EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
@@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
int mtk_afe_add_sub_dai_control(struct snd_soc_component *component)
{
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- int i;
+ struct mtk_base_afe_dai *dai;
- if (!afe->sub_dais) {
- dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
- return -EINVAL;
- }
-
- for (i = 0; i < afe->num_sub_dais; i++) {
- if (afe->sub_dais[i].controls)
+ list_for_each_entry(dai, &afe->sub_dais, list) {
+ if (dai->controls)
snd_soc_add_component_controls(component,
- afe->sub_dais[i].controls,
- afe->sub_dais[i].num_controls);
+ dai->controls,
+ dai->num_controls);
- if (afe->sub_dais[i].dapm_widgets)
+ if (dai->dapm_widgets)
snd_soc_dapm_new_controls(&component->dapm,
- afe->sub_dais[i].dapm_widgets,
- afe->sub_dais[i].num_dapm_widgets);
-
- if (afe->sub_dais[i].dapm_routes)
+ dai->dapm_widgets,
+ dai->num_dapm_widgets);
+ }
+ /* add routes after all widgets are added */
+ list_for_each_entry(dai, &afe->sub_dais, list) {
+ if (dai->dapm_routes)
snd_soc_dapm_add_routes(&component->dapm,
- afe->sub_dais[i].dapm_routes,
- afe->sub_dais[i].num_dapm_routes);
+ dai->dapm_routes,
+ dai->num_dapm_routes);
}
snd_soc_dapm_new_widgets(component->dapm.card);
diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h
index bcf562f029b6..bd8d5e0c6843 100644
--- a/sound/soc/mediatek/common/mtk-base-afe.h
+++ b/sound/soc/mediatek/common/mtk-base-afe.h
@@ -46,6 +46,7 @@ struct mtk_base_irq_data {
};
struct device;
+struct list_head;
struct mtk_base_afe_memif;
struct mtk_base_afe_irq;
struct mtk_base_afe_dai;
@@ -72,8 +73,7 @@ struct mtk_base_afe {
struct mtk_base_afe_irq *irqs;
int irqs_size;
- struct mtk_base_afe_dai *sub_dais;
- int num_sub_dais;
+ struct list_head sub_dais;
struct snd_soc_dai_driver *dai_drivers;
unsigned int num_dai_drivers;
@@ -110,6 +110,8 @@ struct mtk_base_afe_dai {
unsigned int num_dapm_widgets;
const struct snd_soc_dapm_route *dapm_routes;
unsigned int num_dapm_routes;
+
+ struct list_head list;
};
#endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
index 22eb7b455cf1..4eac9977b2b0 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
@@ -10,6 +10,7 @@
#define _MT_6797_AFE_COMMON_H_
#include <sound/soc.h>
+#include <linux/list.h>
#include <linux/regmap.h>
#include "../common/mtk-base-afe.h"
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index 6c5dd9fc9976..192f4d7b37b6 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = {
.probe = mt6797_afe_component_probe,
};
+static int mt6797_dai_memif_register(struct mtk_base_afe *afe)
+{
+ struct mtk_base_afe_dai *dai;
+
+ dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai)
+ return -ENOMEM;
+
+ list_add(&dai->list, &afe->sub_dais);
+
+ dai->dai_drivers = mt6797_memif_dai_driver;
+ dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver);
+
+ dai->dapm_widgets = mt6797_memif_widgets;
+ dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets);
+ dai->dapm_routes = mt6797_memif_routes;
+ dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes);
+ return 0;
+}
+
+typedef int (*dai_register_cb)(struct mtk_base_afe *);
+static const dai_register_cb dai_register_cbs[] = {
+ mt6797_dai_adda_register,
+ mt6797_dai_pcm_register,
+ mt6797_dai_hostless_register,
+ mt6797_dai_memif_register,
+};
+
static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
{
struct mtk_base_afe *afe;
@@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
}
/* init sub_dais */
- afe->num_sub_dais = MT6797_DAI_NUM;
- afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais,
- sizeof(*afe->sub_dais),
- GFP_KERNEL);
- if (!afe->sub_dais)
- return -ENOMEM;
-
- mt6797_dai_adda_register(afe);
- mt6797_dai_pcm_register(afe);
- mt6797_dai_hostless_register(afe);
-
- afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver;
- afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers =
- ARRAY_SIZE(mt6797_memif_dai_driver);
- afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets;
- afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets =
- ARRAY_SIZE(mt6797_memif_widgets);
- afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes;
- afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes =
- ARRAY_SIZE(mt6797_memif_routes);
+ INIT_LIST_HEAD(&afe->sub_dais);
+
+ for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) {
+ ret = dai_register_cbs[i](afe);
+ if (ret) {
+ dev_warn(afe->dev, "dai register i %d fail, ret %d\n",
+ i, ret);
+ return ret;
+ }
+ }
/* init dai_driver and component_driver */
- mtk_afe_combine_sub_dai(afe);
+ ret = mtk_afe_combine_sub_dai(afe);
+ if (ret) {
+ dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n",
+ ret);
+ return ret;
+ }
afe->mtk_afe_hardware = &mt6797_afe_hardware;
afe->memif_fs = mt6797_memif_fs;
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
index ad083265ce94..0ac6409c6d61 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
@@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = {
int mt6797_dai_adda_register(struct mtk_base_afe *afe)
{
- int id = MT6797_DAI_ADDA;
+ struct mtk_base_afe_dai *dai;
- afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver;
- afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+ dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai)
+ return -ENOMEM;
- afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets;
- afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
- afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes;
- afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
+ list_add(&dai->list, &afe->sub_dais);
+
+ dai->dai_drivers = mtk_dai_adda_driver;
+ dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+
+ dai->dapm_widgets = mtk_dai_adda_widgets;
+ dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
+ dai->dapm_routes = mtk_dai_adda_routes;
+ dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
return 0;
}
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
index 4cf985b15a11..ed23e6a53b08 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
@@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = {
int mt6797_dai_hostless_register(struct mtk_base_afe *afe)
{
- int id = MT6797_DAI_HOSTLESS_LPBK;
+ struct mtk_base_afe_dai *dai;
- afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver;
- afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+ dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai)
+ return -ENOMEM;
- afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes;
- afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
+ list_add(&dai->list, &afe->sub_dais);
+
+ dai->dai_drivers = mtk_dai_hostless_driver;
+ dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+
+ dai->dapm_routes = mtk_dai_hostless_routes;
+ dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
return 0;
}
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
index 16d5b5067204..3136f0bc7827 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
@@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = {
int mt6797_dai_pcm_register(struct mtk_base_afe *afe)
{
- int id = MT6797_DAI_PCM_1;
+ struct mtk_base_afe_dai *dai;
- afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver;
- afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+ dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai)
+ return -ENOMEM;
- afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets;
- afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
- afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes;
- afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
+ list_add(&dai->list, &afe->sub_dais);
+ dai->dai_drivers = mtk_dai_pcm_driver;
+ dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+
+ dai->dapm_widgets = mtk_dai_pcm_widgets;
+ dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
+ dai->dapm_routes = mtk_dai_pcm_routes;
+ dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
return 0;
}
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
new file mode 100644
index 000000000000..8af8bc358a90
--- /dev/null
+++ b/sound/soc/meson/Kconfig
@@ -0,0 +1,65 @@
+menu "ASoC support for Amlogic platforms"
+ depends on ARCH_MESON || COMPILE_TEST
+
+config SND_MESON_AXG_FIFO
+ tristate
+ select REGMAP_MMIO
+
+config SND_MESON_AXG_FRDDR
+ tristate "Amlogic AXG Playback FIFO support"
+ select SND_MESON_AXG_FIFO
+ help
+ Select Y or M to add support for the frontend playback interfaces
+ embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TODDR
+ tristate "Amlogic AXG Capture FIFO support"
+ select SND_MESON_AXG_FIFO
+ help
+ Select Y or M to add support for the frontend capture interfaces
+ embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDM_FORMATTER
+ tristate
+ select REGMAP_MMIO
+
+config SND_MESON_AXG_TDM_INTERFACE
+ tristate
+ select SND_MESON_AXG_TDM_FORMATTER
+
+config SND_MESON_AXG_TDMIN
+ tristate "Amlogic AXG TDM Input Support"
+ select SND_MESON_AXG_TDM_FORMATTER
+ select SND_MESON_AXG_TDM_INTERFACE
+ help
+ Select Y or M to add support for TDM input formatter embedded
+ in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDMOUT
+ tristate "Amlogic AXG TDM Output Support"
+ select SND_MESON_AXG_TDM_FORMATTER
+ select SND_MESON_AXG_TDM_INTERFACE
+ help
+ Select Y or M to add support for TDM output formatter embedded
+ in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_SOUND_CARD
+ tristate "Amlogic AXG Sound Card Support"
+ select SND_MESON_AXG_TDM_INTERFACE
+ imply SND_MESON_AXG_FRDDR
+ imply SND_MESON_AXG_TODDR
+ imply SND_MESON_AXG_TDMIN
+ imply SND_MESON_AXG_TDMOUT
+ imply SND_MESON_AXG_SPDIFOUT
+ help
+ Select Y or M to add support for the AXG SoC sound card
+
+config SND_MESON_AXG_SPDIFOUT
+ tristate "Amlogic AXG SPDIF Output Support"
+ select SND_PCM_IEC958
+ imply SND_SOC_SPDIF
+ help
+ Select Y or M to add support for SPDIF output serializer embedded
+ in the Amlogic AXG SoC family
+
+endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
new file mode 100644
index 000000000000..c5e003b093db
--- /dev/null
+++ b/sound/soc/meson/Makefile
@@ -0,0 +1,21 @@
+# SPDX-License-Identifier: (GPL-2.0 OR MIT)
+
+snd-soc-meson-axg-fifo-objs := axg-fifo.o
+snd-soc-meson-axg-frddr-objs := axg-frddr.o
+snd-soc-meson-axg-toddr-objs := axg-toddr.o
+snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o
+snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o
+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
+
+obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
+obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
+obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o
+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
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
new file mode 100644
index 000000000000..2914ba0d965b
--- /dev/null
+++ b/sound/soc/meson/axg-card.c
@@ -0,0 +1,671 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+struct axg_card {
+ struct snd_soc_card card;
+ void **link_data;
+};
+
+struct axg_dai_link_tdm_mask {
+ u32 tx;
+ u32 rx;
+};
+
+struct axg_dai_link_tdm_data {
+ unsigned int mclk_fs;
+ unsigned int slots;
+ unsigned int slot_width;
+ u32 *tx_mask;
+ u32 *rx_mask;
+ struct axg_dai_link_tdm_mask *codec_masks;
+};
+
+#define PREFIX "amlogic,"
+
+static int axg_card_reallocate_links(struct axg_card *priv,
+ unsigned int num_links)
+{
+ struct snd_soc_dai_link *links;
+ void **ldata;
+
+ links = krealloc(priv->card.dai_link,
+ num_links * sizeof(*priv->card.dai_link),
+ GFP_KERNEL | __GFP_ZERO);
+ ldata = krealloc(priv->link_data,
+ num_links * sizeof(*priv->link_data),
+ GFP_KERNEL | __GFP_ZERO);
+
+ if (!links || !ldata) {
+ dev_err(priv->card.dev, "failed to allocate links\n");
+ return -ENOMEM;
+ }
+
+ priv->card.dai_link = links;
+ priv->link_data = ldata;
+ priv->card.num_links = num_links;
+ return 0;
+}
+
+static int axg_card_parse_dai(struct snd_soc_card *card,
+ struct device_node *node,
+ struct device_node **dai_of_node,
+ const char **dai_name)
+{
+ struct of_phandle_args args;
+ int ret;
+
+ if (!dai_name || !dai_of_node || !node)
+ return -EINVAL;
+
+ ret = of_parse_phandle_with_args(node, "sound-dai",
+ "#sound-dai-cells", 0, &args);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(card->dev, "can't parse dai %d\n", ret);
+ return ret;
+ }
+ *dai_of_node = args.np;
+
+ return snd_soc_get_dai_name(&args, dai_name);
+}
+
+static int axg_card_set_link_name(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ const char *prefix)
+{
+ char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
+ prefix, link->cpu_of_node->full_name);
+ if (!name)
+ return -ENOMEM;
+
+ link->name = name;
+ link->stream_name = name;
+
+ return 0;
+}
+
+static void axg_card_clean_references(struct axg_card *priv)
+{
+ struct snd_soc_card *card = &priv->card;
+ struct snd_soc_dai_link *link;
+ 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);
+ }
+ }
+
+ if (card->aux_dev) {
+ for (i = 0; i < card->num_aux_devs; i++)
+ of_node_put(card->aux_dev[i].codec_of_node);
+ }
+
+ kfree(card->dai_link);
+ kfree(priv->link_data);
+}
+
+static int axg_card_add_aux_devices(struct snd_soc_card *card)
+{
+ struct device_node *node = card->dev->of_node;
+ struct snd_soc_aux_dev *aux;
+ int num, i;
+
+ num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
+ if (num == -ENOENT) {
+ /*
+ * It is ok to have no auxiliary devices but for this card it
+ * is a strange situtation. Let's warn the about it.
+ */
+ dev_warn(card->dev, "card has no auxiliary devices\n");
+ return 0;
+ } else if (num < 0) {
+ dev_err(card->dev, "error getting auxiliary devices: %d\n",
+ num);
+ return num;
+ }
+
+ aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
+ if (!aux)
+ return -ENOMEM;
+ card->aux_dev = aux;
+ card->num_aux_devs = num;
+
+ for (i = 0; i < card->num_aux_devs; i++, aux++) {
+ aux->codec_of_node =
+ of_parse_phandle(node, "audio-aux-devs", i);
+ if (!aux->codec_of_node)
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct axg_dai_link_tdm_data *be =
+ (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+ struct snd_soc_dai *codec_dai;
+ unsigned int mclk;
+ int ret, i;
+
+ 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];
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+ SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk,
+ SND_SOC_CLOCK_OUT);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_ops axg_card_tdm_be_ops = {
+ .hw_params = axg_card_tdm_be_hw_params,
+};
+
+static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct axg_dai_link_tdm_data *be =
+ (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+ struct snd_soc_dai *codec_dai;
+ int ret, i;
+
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ ret = snd_soc_dai_set_tdm_slot(codec_dai,
+ be->codec_masks[i].tx,
+ be->codec_masks[i].rx,
+ be->slots, be->slot_width);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(codec_dai->dev,
+ "setting tdm link slots failed\n");
+ return ret;
+ }
+ }
+
+ ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask,
+ be->slots, be->slot_width);
+ if (ret) {
+ dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct axg_dai_link_tdm_data *be =
+ (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+ int ret;
+
+ /* The loopback rx_mask is the pad tx_mask */
+ ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask,
+ be->slots, be->slot_width);
+ if (ret) {
+ dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
+ int *index)
+{
+ struct axg_card *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai_link *pad = &card->dai_link[*index];
+ struct snd_soc_dai_link *lb;
+ int ret;
+
+ /* extend links */
+ ret = axg_card_reallocate_links(priv, card->num_links + 1);
+ if (ret)
+ return ret;
+
+ lb = &card->dai_link[*index + 1];
+
+ lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+ if (!lb->name)
+ return -ENOMEM;
+
+ lb->stream_name = lb->name;
+ lb->cpu_of_node = pad->cpu_of_node;
+ lb->cpu_dai_name = "TDM Loopback";
+ lb->codec_name = "snd-soc-dummy";
+ lb->codec_dai_name = "snd-soc-dummy-dai";
+ lb->dpcm_capture = 1;
+ lb->no_pcm = 1;
+ lb->ops = &axg_card_tdm_be_ops;
+ lb->init = axg_card_tdm_dai_lb_init;
+
+ /* Provide the same link data to the loopback */
+ priv->link_data[*index + 1] = priv->link_data[*index];
+
+ /*
+ * axg_card_clean_references() will iterate over this link,
+ * make sure the node count is balanced
+ */
+ of_node_get(lb->cpu_of_node);
+
+ /* Let add_links continue where it should */
+ *index += 1;
+
+ return 0;
+}
+
+static unsigned int axg_card_parse_daifmt(struct device_node *node,
+ struct device_node *cpu_node)
+{
+ struct device_node *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
+ unsigned int daifmt;
+
+ daifmt = snd_soc_of_parse_daifmt(node, PREFIX,
+ &bitclkmaster, &framemaster);
+ daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+ /* If no master is provided, default to cpu master */
+ if (!bitclkmaster || bitclkmaster == cpu_node) {
+ daifmt |= (!framemaster || framemaster == cpu_node) ?
+ SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
+ } else {
+ daifmt |= (!framemaster || framemaster == cpu_node) ?
+ SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
+ }
+
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+
+ return daifmt;
+}
+
+static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node,
+ struct axg_dai_link_tdm_data *be)
+{
+ char propname[32];
+ u32 tx, rx;
+ int i;
+
+ be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+ sizeof(*be->tx_mask), GFP_KERNEL);
+ be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+ sizeof(*be->rx_mask), GFP_KERNEL);
+ if (!be->tx_mask || !be->rx_mask)
+ return -ENOMEM;
+
+ for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) {
+ snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i);
+ snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]);
+ tx = max(tx, be->tx_mask[i]);
+ }
+
+ /* Disable playback is the interface has no tx slots */
+ if (!tx)
+ link->dpcm_playback = 0;
+
+ for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) {
+ snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i);
+ snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]);
+ rx = max(rx, be->rx_mask[i]);
+ }
+
+ /* Disable capture is the interface has no rx slots */
+ if (!rx)
+ link->dpcm_capture = 0;
+
+ /* ... but the interface should at least have one of them */
+ if (!tx && !rx) {
+ dev_err(card->dev, "tdm link has no cpu slots\n");
+ return -EINVAL;
+ }
+
+ of_property_read_u32(node, "dai-tdm-slot-num", &be->slots);
+ if (!be->slots) {
+ /*
+ * If the slot number is not provided, set it such as it
+ * accommodates the largest mask
+ */
+ be->slots = fls(max(tx, rx));
+ } else if (be->slots < fls(max(tx, rx)) || be->slots > 32) {
+ /*
+ * Error if the slots can't accommodate the largest mask or
+ * if it is just too big
+ */
+ dev_err(card->dev, "bad slot number\n");
+ return -EINVAL;
+ }
+
+ of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width);
+
+ return 0;
+}
+
+static int axg_card_parse_codecs_masks(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node,
+ struct axg_dai_link_tdm_data *be)
+{
+ struct axg_dai_link_tdm_mask *codec_mask;
+ struct device_node *np;
+
+ codec_mask = devm_kcalloc(card->dev, link->num_codecs,
+ sizeof(*codec_mask), GFP_KERNEL);
+ if (!codec_mask)
+ return -ENOMEM;
+
+ be->codec_masks = codec_mask;
+
+ for_each_child_of_node(node, np) {
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask",
+ &codec_mask->rx);
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask",
+ &codec_mask->tx);
+
+ codec_mask++;
+ }
+
+ return 0;
+}
+
+static int axg_card_parse_tdm(struct snd_soc_card *card,
+ struct device_node *node,
+ int *index)
+{
+ struct axg_card *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai_link *link = &card->dai_link[*index];
+ struct axg_dai_link_tdm_data *be;
+ int ret;
+
+ /* Allocate tdm link parameters */
+ be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL);
+ if (!be)
+ return -ENOMEM;
+ priv->link_data[*index] = be;
+
+ /* Setup tdm link */
+ link->ops = &axg_card_tdm_be_ops;
+ link->init = axg_card_tdm_dai_init;
+ link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node);
+
+ of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
+
+ ret = axg_card_parse_cpu_tdm_slots(card, link, node, be);
+ if (ret) {
+ dev_err(card->dev, "error parsing tdm link slots\n");
+ return ret;
+ }
+
+ ret = axg_card_parse_codecs_masks(card, link, node, be);
+ if (ret)
+ return ret;
+
+ /* Add loopback if the pad dai has playback */
+ if (link->dpcm_playback) {
+ ret = axg_card_add_tdm_loopback(card, index);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_card_set_be_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node)
+{
+ struct snd_soc_dai_link_component *codec;
+ struct device_node *np;
+ int ret, num_codecs;
+
+ link->no_pcm = 1;
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+
+ num_codecs = of_get_child_count(node);
+ if (!num_codecs) {
+ dev_err(card->dev, "be link %s has no codec\n",
+ node->full_name);
+ return -EINVAL;
+ }
+
+ codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
+ if (!codec)
+ return -ENOMEM;
+
+ link->codecs = codec;
+ link->num_codecs = num_codecs;
+
+ for_each_child_of_node(node, np) {
+ ret = axg_card_parse_dai(card, np, &codec->of_node,
+ &codec->dai_name);
+ if (ret) {
+ of_node_put(np);
+ return ret;
+ }
+
+ codec++;
+ }
+
+ ret = axg_card_set_link_name(card, link, "be");
+ if (ret)
+ dev_err(card->dev, "error setting %s link name\n", np->name);
+
+ return ret;
+}
+
+static int axg_card_set_fe_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ bool is_playback)
+{
+ link->dynamic = 1;
+ link->dpcm_merged_format = 1;
+ link->dpcm_merged_chan = 1;
+ link->dpcm_merged_rate = 1;
+ link->codec_dai_name = "snd-soc-dummy-dai";
+ link->codec_name = "snd-soc-dummy";
+
+ if (is_playback)
+ link->dpcm_playback = 1;
+ else
+ link->dpcm_capture = 1;
+
+ return axg_card_set_link_name(card, link, "fe");
+}
+
+static int axg_card_cpu_is_capture_fe(struct device_node *np)
+{
+ return of_device_is_compatible(np, PREFIX "axg-toddr");
+}
+
+static int axg_card_cpu_is_playback_fe(struct device_node *np)
+{
+ return of_device_is_compatible(np, PREFIX "axg-frddr");
+}
+
+static int axg_card_cpu_is_tdm_iface(struct device_node *np)
+{
+ return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
+}
+
+static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
+ int *index)
+{
+ struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+ int ret;
+
+ ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node,
+ &dai_link->cpu_dai_name);
+ if (ret)
+ return ret;
+
+ if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node))
+ ret = axg_card_set_fe_link(card, dai_link, true);
+ else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node))
+ ret = axg_card_set_fe_link(card, dai_link, false);
+ else
+ ret = axg_card_set_be_link(card, dai_link, np);
+
+ if (ret)
+ return ret;
+
+ if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node))
+ ret = axg_card_parse_tdm(card, np, index);
+
+ return ret;
+}
+
+static int axg_card_add_links(struct snd_soc_card *card)
+{
+ struct axg_card *priv = snd_soc_card_get_drvdata(card);
+ struct device_node *node = card->dev->of_node;
+ struct device_node *np;
+ int num, i, ret;
+
+ num = of_get_child_count(node);
+ if (!num) {
+ dev_err(card->dev, "card has no links\n");
+ return -EINVAL;
+ }
+
+ ret = axg_card_reallocate_links(priv, num);
+ if (ret)
+ return ret;
+
+ i = 0;
+ for_each_child_of_node(node, np) {
+ ret = axg_card_add_link(card, np, &i);
+ if (ret) {
+ of_node_put(np);
+ return ret;
+ }
+
+ i++;
+ }
+
+ return 0;
+}
+
+static int axg_card_parse_of_optional(struct snd_soc_card *card,
+ const char *propname,
+ int (*func)(struct snd_soc_card *c,
+ const char *p))
+{
+ /* If property is not provided, don't fail ... */
+ if (!of_property_read_bool(card->dev->of_node, propname))
+ return 0;
+
+ /* ... but do fail if it is provided and the parsing fails */
+ return func(card, propname);
+}
+
+static const struct of_device_id axg_card_of_match[] = {
+ { .compatible = "amlogic,axg-sound-card", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, axg_card_of_match);
+
+static int axg_card_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct axg_card *priv;
+ int ret;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, priv);
+ snd_soc_card_set_drvdata(&priv->card, priv);
+
+ priv->card.owner = THIS_MODULE;
+ priv->card.dev = dev;
+
+ ret = snd_soc_of_parse_card_name(&priv->card, "model");
+ if (ret < 0)
+ return ret;
+
+ ret = axg_card_parse_of_optional(&priv->card, "audio-routing",
+ snd_soc_of_parse_audio_routing);
+ if (ret) {
+ dev_err(dev, "error while parsing routing\n");
+ return ret;
+ }
+
+ ret = axg_card_parse_of_optional(&priv->card, "audio-widgets",
+ snd_soc_of_parse_audio_simple_widgets);
+ if (ret) {
+ dev_err(dev, "error while parsing widgets\n");
+ return ret;
+ }
+
+ ret = axg_card_add_links(&priv->card);
+ if (ret)
+ goto out_err;
+
+ ret = axg_card_add_aux_devices(&priv->card);
+ if (ret)
+ goto out_err;
+
+ ret = devm_snd_soc_register_card(dev, &priv->card);
+ if (ret)
+ goto out_err;
+
+ return 0;
+
+out_err:
+ axg_card_clean_references(priv);
+ return ret;
+}
+
+static int axg_card_remove(struct platform_device *pdev)
+{
+ struct axg_card *priv = platform_get_drvdata(pdev);
+
+ axg_card_clean_references(priv);
+
+ return 0;
+}
+
+static struct platform_driver axg_card_pdrv = {
+ .probe = axg_card_probe,
+ .remove = axg_card_remove,
+ .driver = {
+ .name = "axg-sound-card",
+ .of_match_table = axg_card_of_match,
+ },
+};
+module_platform_driver(axg_card_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
new file mode 100644
index 000000000000..30262550e37b
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.c
@@ -0,0 +1,341 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+/*
+ * This file implements the platform operations common to the playback and
+ * capture frontend DAI. The logic behind this two types of fifo is very
+ * similar but some difference exist.
+ * These differences the respective DAI drivers
+ */
+
+static struct snd_pcm_hardware axg_fifo_hw = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE),
+
+ .formats = AXG_FIFO_FORMATS,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = AXG_FIFO_CH_MAX,
+ .period_bytes_min = AXG_FIFO_MIN_DEPTH,
+ .period_bytes_max = UINT_MAX,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+
+ /* No real justification for this */
+ .buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+ return rtd->cpu_dai;
+}
+
+static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+ return snd_soc_dai_get_drvdata(dai);
+}
+
+static struct device *axg_fifo_dev(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+ return dai->dev;
+}
+
+static void __dma_enable(struct axg_fifo *fifo, bool enable)
+{
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN,
+ enable ? CTRL0_DMA_EN : 0);
+}
+
+static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ __dma_enable(fifo, true);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_STOP:
+ __dma_enable(fifo, false);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss)
+{
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+ struct snd_pcm_runtime *runtime = ss->runtime;
+ unsigned int addr;
+
+ regmap_read(fifo->map, FIFO_STATUS2, &addr);
+
+ return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
+}
+
+static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = ss->runtime;
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+ dma_addr_t end_ptr;
+ unsigned int burst_num;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params));
+ if (ret < 0)
+ return ret;
+
+ /* Setup dma memory pointers */
+ end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST;
+ regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr);
+ regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr);
+
+ /* Setup interrupt periodicity */
+ burst_num = params_period_bytes(params) / AXG_FIFO_BURST;
+ regmap_write(fifo->map, FIFO_INT_ADDR, burst_num);
+
+ /* Enable block count irq */
+ regmap_update_bits(fifo->map, FIFO_CTRL0,
+ CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT),
+ CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT));
+
+ return 0;
+}
+
+static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss)
+{
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+
+ /* Disable the block count irq */
+ regmap_update_bits(fifo->map, FIFO_CTRL0,
+ CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0);
+
+ return snd_pcm_lib_free_pages(ss);
+}
+
+static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask)
+{
+ regmap_update_bits(fifo->map, FIFO_CTRL1,
+ CTRL1_INT_CLR(FIFO_INT_MASK),
+ CTRL1_INT_CLR(mask));
+
+ /* Clear must also be cleared */
+ regmap_update_bits(fifo->map, FIFO_CTRL1,
+ CTRL1_INT_CLR(FIFO_INT_MASK),
+ 0);
+}
+
+static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id)
+{
+ struct snd_pcm_substream *ss = dev_id;
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+ unsigned int status;
+
+ regmap_read(fifo->map, FIFO_STATUS1, &status);
+
+ status = STATUS1_INT_STS(status) & FIFO_INT_MASK;
+ if (status & FIFO_INT_COUNT_REPEAT)
+ snd_pcm_period_elapsed(ss);
+ else
+ dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n",
+ status);
+
+ /* Ack irqs */
+ axg_fifo_ack_irq(fifo, status);
+
+ return IRQ_RETVAL(status);
+}
+
+static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
+{
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+ struct device *dev = axg_fifo_dev(ss);
+ int ret;
+
+ snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw);
+
+ /*
+ * Make sure the buffer and period size are multiple of the FIFO
+ * minimum depth size
+ */
+ ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ AXG_FIFO_MIN_DEPTH);
+ if (ret)
+ return ret;
+
+ ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ AXG_FIFO_MIN_DEPTH);
+ if (ret)
+ return ret;
+
+ ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
+ dev_name(dev), ss);
+
+ /* Enable pclk to access registers and clock the fifo ip */
+ ret = clk_prepare_enable(fifo->pclk);
+ if (ret)
+ return ret;
+
+ /* Setup status2 so it reports the memory pointer */
+ regmap_update_bits(fifo->map, FIFO_CTRL1,
+ CTRL1_STATUS2_SEL_MASK,
+ CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ));
+
+ /* Make sure the dma is initially disabled */
+ __dma_enable(fifo, false);
+
+ /* Disable irqs until params are ready */
+ regmap_update_bits(fifo->map, FIFO_CTRL0,
+ CTRL0_INT_EN(FIFO_INT_MASK), 0);
+
+ /* Clear any pending interrupt */
+ axg_fifo_ack_irq(fifo, FIFO_INT_MASK);
+
+ /* Take memory arbitror out of reset */
+ ret = reset_control_deassert(fifo->arb);
+ if (ret)
+ clk_disable_unprepare(fifo->pclk);
+
+ return ret;
+}
+
+static int axg_fifo_pcm_close(struct snd_pcm_substream *ss)
+{
+ struct axg_fifo *fifo = axg_fifo_data(ss);
+ int ret;
+
+ /* Put the memory arbitror back in reset */
+ ret = reset_control_assert(fifo->arb);
+
+ /* Disable fifo ip and register access */
+ clk_disable_unprepare(fifo->pclk);
+
+ /* remove IRQ */
+ free_irq(fifo->irq, ss);
+
+ return ret;
+}
+
+const struct snd_pcm_ops axg_fifo_pcm_ops = {
+ .open = axg_fifo_pcm_open,
+ .close = axg_fifo_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = axg_fifo_pcm_hw_params,
+ .hw_free = axg_fifo_pcm_hw_free,
+ .pointer = axg_fifo_pcm_pointer,
+ .trigger = axg_fifo_pcm_trigger,
+};
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops);
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ size_t size = axg_fifo_hw.buffer_bytes_max;
+
+ return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream,
+ SNDRV_DMA_TYPE_DEV, card->dev,
+ size, size);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_new);
+
+static const struct regmap_config axg_fifo_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = FIFO_STATUS2,
+};
+
+int axg_fifo_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ const struct axg_fifo_match_data *data;
+ struct axg_fifo *fifo;
+ struct resource *res;
+ void __iomem *regs;
+
+ data = of_device_get_match_data(dev);
+ if (!data) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
+ fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL);
+ if (!fifo)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, fifo);
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg);
+ if (IS_ERR(fifo->map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(fifo->map));
+ return PTR_ERR(fifo->map);
+ }
+
+ fifo->pclk = devm_clk_get(dev, NULL);
+ if (IS_ERR(fifo->pclk)) {
+ if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER)
+ dev_err(dev, "failed to get pclk: %ld\n",
+ PTR_ERR(fifo->pclk));
+ return PTR_ERR(fifo->pclk);
+ }
+
+ fifo->arb = devm_reset_control_get_exclusive(dev, NULL);
+ if (IS_ERR(fifo->arb)) {
+ if (PTR_ERR(fifo->arb) != -EPROBE_DEFER)
+ dev_err(dev, "failed to get arb reset: %ld\n",
+ PTR_ERR(fifo->arb));
+ return PTR_ERR(fifo->arb);
+ }
+
+ fifo->irq = of_irq_get(dev->of_node, 0);
+ if (fifo->irq <= 0) {
+ dev_err(dev, "failed to get irq: %d\n", fifo->irq);
+ return fifo->irq;
+ }
+
+ return devm_snd_soc_register_component(dev, data->component_drv,
+ data->dai_drv, 1);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_probe);
+
+MODULE_DESCRIPTION("Amlogic AXG fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h
new file mode 100644
index 000000000000..cb6c4013ca33
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.h
@@ -0,0 +1,80 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2018 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_FIFO_H
+#define _MESON_AXG_FIFO_H
+
+struct clk;
+struct platform_device;
+struct regmap;
+struct reset_control;
+
+struct snd_soc_component_driver;
+struct snd_soc_dai;
+struct snd_soc_dai_driver;
+struct snd_pcm_ops;
+struct snd_soc_pcm_runtime;
+
+#define AXG_FIFO_CH_MAX 128
+#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \
+ SNDRV_PCM_RATE_8000_192000)
+#define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AXG_FIFO_BURST 8
+#define AXG_FIFO_MIN_CNT 64
+#define AXG_FIFO_MIN_DEPTH (AXG_FIFO_BURST * AXG_FIFO_MIN_CNT)
+
+#define FIFO_INT_ADDR_FINISH BIT(0)
+#define FIFO_INT_ADDR_INT BIT(1)
+#define FIFO_INT_COUNT_REPEAT BIT(2)
+#define FIFO_INT_COUNT_ONCE BIT(3)
+#define FIFO_INT_FIFO_ZERO BIT(4)
+#define FIFO_INT_FIFO_DEPTH BIT(5)
+#define FIFO_INT_MASK GENMASK(7, 0)
+
+#define FIFO_CTRL0 0x00
+#define CTRL0_DMA_EN BIT(31)
+#define CTRL0_INT_EN(x) ((x) << 16)
+#define CTRL0_SEL_MASK GENMASK(2, 0)
+#define CTRL0_SEL_SHIFT 0
+#define FIFO_CTRL1 0x04
+#define CTRL1_INT_CLR(x) ((x) << 0)
+#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8)
+#define CTRL1_STATUS2_SEL(x) ((x) << 8)
+#define STATUS2_SEL_DDR_READ 0
+#define CTRL1_THRESHOLD_MASK GENMASK(23, 16)
+#define CTRL1_THRESHOLD(x) ((x) << 16)
+#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24)
+#define CTRL1_FRDDR_DEPTH(x) ((x) << 24)
+#define FIFO_START_ADDR 0x08
+#define FIFO_FINISH_ADDR 0x0c
+#define FIFO_INT_ADDR 0x10
+#define FIFO_STATUS1 0x14
+#define STATUS1_INT_STS(x) ((x) << 0)
+#define FIFO_STATUS2 0x18
+
+struct axg_fifo {
+ struct regmap *map;
+ struct clk *pclk;
+ struct reset_control *arb;
+ int irq;
+};
+
+struct axg_fifo_match_data {
+ const struct snd_soc_component_driver *component_drv;
+ struct snd_soc_dai_driver *dai_drv;
+};
+
+extern const struct snd_pcm_ops axg_fifo_pcm_ops;
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type);
+int axg_fifo_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_FIFO_H */
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
new file mode 100644
index 000000000000..a6f6f6a2eca8
--- /dev/null
+++ b/sound/soc/meson/axg-frddr.c
@@ -0,0 +1,141 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend playback DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_FRDDR_PP_MODE BIT(30)
+
+static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+ unsigned int fifo_depth, fifo_threshold;
+ int ret;
+
+ /* Enable pclk to access registers and clock the fifo ip */
+ ret = clk_prepare_enable(fifo->pclk);
+ if (ret)
+ return ret;
+
+ /* Apply single buffer mode to the interface */
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0);
+
+ /*
+ * TODO: We could adapt the fifo depth and the fifo threshold
+ * depending on the expected memory throughput and lantencies
+ * For now, we'll just use the same values as the vendor kernel
+ * Depth and threshold are zero based.
+ */
+ fifo_depth = AXG_FIFO_MIN_CNT - 1;
+ fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1;
+ regmap_update_bits(fifo->map, FIFO_CTRL1,
+ CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK,
+ CTRL1_FRDDR_DEPTH(fifo_depth) |
+ CTRL1_THRESHOLD(fifo_threshold));
+
+ return 0;
+}
+
+static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable_unprepare(fifo->pclk);
+}
+
+static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *dai)
+{
+ return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static const struct snd_soc_dai_ops axg_frddr_ops = {
+ .startup = axg_frddr_dai_startup,
+ .shutdown = axg_frddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_frddr_dai_drv = {
+ .name = "FRDDR",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = AXG_FIFO_CH_MAX,
+ .rates = AXG_FIFO_RATES,
+ .formats = AXG_FIFO_FORMATS,
+ },
+ .ops = &axg_frddr_ops,
+ .pcm_new = axg_frddr_pcm_new,
+};
+
+static const char * const axg_frddr_sel_texts[] = {
+ "OUT 0", "OUT 1", "OUT 2", "OUT 3"
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
+ axg_frddr_sel_texts);
+
+static const struct snd_kcontrol_new axg_frddr_out_demux =
+ SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = {
+ SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0,
+ &axg_frddr_out_demux),
+ SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = {
+ { "SINK SEL", NULL, "Playback" },
+ { "OUT 0", "OUT 0", "SINK SEL" },
+ { "OUT 1", "OUT 1", "SINK SEL" },
+ { "OUT 2", "OUT 2", "SINK SEL" },
+ { "OUT 3", "OUT 3", "SINK SEL" },
+};
+
+static const struct snd_soc_component_driver axg_frddr_component_drv = {
+ .dapm_widgets = axg_frddr_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_frddr_dapm_widgets),
+ .dapm_routes = axg_frddr_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_frddr_dapm_routes),
+ .ops = &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_frddr_match_data = {
+ .component_drv = &axg_frddr_component_drv,
+ .dai_drv = &axg_frddr_dai_drv
+};
+
+static const struct of_device_id axg_frddr_of_match[] = {
+ {
+ .compatible = "amlogic,axg-frddr",
+ .data = &axg_frddr_match_data,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_frddr_of_match);
+
+static struct platform_driver axg_frddr_pdrv = {
+ .probe = axg_fifo_probe,
+ .driver = {
+ .name = "axg-frddr",
+ .of_match_table = axg_frddr_of_match,
+ },
+};
+module_platform_driver(axg_frddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG playback fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
new file mode 100644
index 000000000000..9dea528053ad
--- /dev/null
+++ b/sound/soc/meson/axg-spdifout.c
@@ -0,0 +1,456 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm_iec958.h>
+
+/*
+ * NOTE:
+ * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite
+ * of what the documentation says. Manual control on V, U and C bits is
+ * applied when the related sel bits are cleared
+ */
+
+#define SPDIFOUT_STAT 0x00
+#define SPDIFOUT_GAIN0 0x04
+#define SPDIFOUT_GAIN1 0x08
+#define SPDIFOUT_CTRL0 0x0c
+#define SPDIFOUT_CTRL0_EN BIT(31)
+#define SPDIFOUT_CTRL0_RST_OUT BIT(29)
+#define SPDIFOUT_CTRL0_RST_IN BIT(28)
+#define SPDIFOUT_CTRL0_USEL BIT(26)
+#define SPDIFOUT_CTRL0_USET BIT(25)
+#define SPDIFOUT_CTRL0_CHSTS_SEL BIT(24)
+#define SPDIFOUT_CTRL0_DATA_SEL BIT(20)
+#define SPDIFOUT_CTRL0_MSB_FIRST BIT(19)
+#define SPDIFOUT_CTRL0_VSEL BIT(18)
+#define SPDIFOUT_CTRL0_VSET BIT(17)
+#define SPDIFOUT_CTRL0_MASK_MASK GENMASK(11, 4)
+#define SPDIFOUT_CTRL0_MASK(x) ((x) << 4)
+#define SPDIFOUT_CTRL1 0x10
+#define SPDIFOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8)
+#define SPDIFOUT_CTRL1_MSB_POS(x) ((x) << 8)
+#define SPDIFOUT_CTRL1_TYPE_MASK GENMASK(6, 4)
+#define SPDIFOUT_CTRL1_TYPE(x) ((x) << 4)
+#define SPDIFOUT_PREAMB 0x14
+#define SPDIFOUT_SWAP 0x18
+#define SPDIFOUT_CHSTS0 0x1c
+#define SPDIFOUT_CHSTS1 0x20
+#define SPDIFOUT_CHSTS2 0x24
+#define SPDIFOUT_CHSTS3 0x28
+#define SPDIFOUT_CHSTS4 0x2c
+#define SPDIFOUT_CHSTS5 0x30
+#define SPDIFOUT_CHSTS6 0x34
+#define SPDIFOUT_CHSTS7 0x38
+#define SPDIFOUT_CHSTS8 0x3c
+#define SPDIFOUT_CHSTS9 0x40
+#define SPDIFOUT_CHSTSA 0x44
+#define SPDIFOUT_CHSTSB 0x48
+#define SPDIFOUT_MUTE_VAL 0x4c
+
+struct axg_spdifout {
+ struct regmap *map;
+ struct clk *mclk;
+ struct clk *pclk;
+};
+
+static void axg_spdifout_enable(struct regmap *map)
+{
+ /* Apply both reset */
+ regmap_update_bits(map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN,
+ 0);
+
+ /* Clear out reset before in reset */
+ regmap_update_bits(map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT);
+ regmap_update_bits(map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_RST_IN, SPDIFOUT_CTRL0_RST_IN);
+
+ /* Enable spdifout */
+ regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN,
+ SPDIFOUT_CTRL0_EN);
+}
+
+static void axg_spdifout_disable(struct regmap *map)
+{
+ regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0);
+}
+
+static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *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_spdifout_enable(priv->map);
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ axg_spdifout_disable(priv->map);
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+ /* Use spdif valid bit to perform digital mute */
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET,
+ mute ? SPDIFOUT_CTRL0_VSET : 0);
+
+ return 0;
+}
+
+static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned int val;
+
+ /* Set the samples spdifout will pull from the FIFO */
+ switch (params_channels(params)) {
+ case 1:
+ val = SPDIFOUT_CTRL0_MASK(0x1);
+ break;
+ case 2:
+ val = SPDIFOUT_CTRL0_MASK(0x3);
+ break;
+ default:
+ dev_err(dai->dev, "too many channels for spdif dai: %u\n",
+ params_channels(params));
+ return -EINVAL;
+ }
+
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_MASK_MASK, val);
+
+ /* FIFO data are arranged in chunks of 64bits */
+ switch (params_physical_width(params)) {
+ case 8:
+ /* 8 samples of 8 bits */
+ val = SPDIFOUT_CTRL1_TYPE(0);
+ break;
+ case 16:
+ /* 4 samples of 16 bits - right justified */
+ val = SPDIFOUT_CTRL1_TYPE(2);
+ break;
+ case 32:
+ /* 2 samples of 32 bits - right justified */
+ val = SPDIFOUT_CTRL1_TYPE(4);
+ break;
+ default:
+ dev_err(dai->dev, "Unsupported physical width: %u\n",
+ params_physical_width(params));
+ return -EINVAL;
+ }
+
+ /* Position of the MSB in FIFO samples */
+ val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1);
+
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL1,
+ SPDIFOUT_CTRL1_MSB_POS_MASK |
+ SPDIFOUT_CTRL1_TYPE_MASK, val);
+
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+ 0);
+
+ return 0;
+}
+
+static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned int offset;
+ int ret;
+ u8 cs[4];
+ u32 val;
+
+ ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4);
+ if (ret < 0) {
+ dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+ ret);
+ return ret;
+ }
+ val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24;
+
+ /* Setup channel status A bits [31 - 0]*/
+ regmap_write(priv->map, SPDIFOUT_CHSTS0, val);
+
+ /* Clear channel status A bits [191 - 32] */
+ for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5;
+ offset += regmap_get_reg_stride(priv->map))
+ regmap_write(priv->map, offset, 0);
+
+ /* Setup channel status B bits [31 - 0]*/
+ regmap_write(priv->map, SPDIFOUT_CHSTS6, val);
+
+ /* Clear channel status B bits [191 - 32] */
+ for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB;
+ offset += regmap_get_reg_stride(priv->map))
+ regmap_write(priv->map, offset, 0);
+
+ return 0;
+}
+
+static int axg_spdifout_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned int rate = params_rate(params);
+ int ret;
+
+ /* 2 * 32bits per subframe * 2 channels = 128 */
+ ret = clk_set_rate(priv->mclk, rate * 128);
+ if (ret) {
+ dev_err(dai->dev, "failed to set spdif clock\n");
+ return ret;
+ }
+
+ ret = axg_spdifout_sample_fmt(params, dai);
+ if (ret) {
+ dev_err(dai->dev, "failed to setup sample format\n");
+ return ret;
+ }
+
+ ret = axg_spdifout_set_chsts(params, dai);
+ if (ret) {
+ dev_err(dai->dev, "failed to setup channel status words\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_spdifout_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ /* Clock the spdif output block */
+ ret = clk_prepare_enable(priv->pclk);
+ if (ret) {
+ dev_err(dai->dev, "failed to enable pclk\n");
+ return ret;
+ }
+
+ /* Make sure the block is initially stopped */
+ axg_spdifout_disable(priv->map);
+
+ /* Insert data from bit 27 lsb first */
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+ 0);
+
+ /* Manual control of V, C and U, U = 0 */
+ regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+ SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL |
+ SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET,
+ 0);
+
+ /* Static SWAP configuration ATM */
+ regmap_write(priv->map, SPDIFOUT_SWAP, 0x10);
+
+ return 0;
+}
+
+static void axg_spdifout_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable_unprepare(priv->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_spdifout_ops = {
+ .trigger = axg_spdifout_trigger,
+ .digital_mute = axg_spdifout_digital_mute,
+ .hw_params = axg_spdifout_hw_params,
+ .startup = axg_spdifout_startup,
+ .shutdown = axg_spdifout_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = {
+ {
+ .name = "SPDIF Output",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000),
+ .formats = (SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &axg_spdifout_ops,
+ },
+};
+
+static const char * const spdifout_sel_texts[] = {
+ "IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24,
+ spdifout_sel_texts);
+
+static const struct snd_kcontrol_new axg_spdifout_in_mux =
+ SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux),
+};
+
+static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = {
+ { "SRC SEL", "IN 0", "IN 0" },
+ { "SRC SEL", "IN 1", "IN 1" },
+ { "SRC SEL", "IN 2", "IN 2" },
+ { "Playback", NULL, "SRC SEL" },
+};
+
+static const struct snd_kcontrol_new axg_spdifout_controls[] = {
+ SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0, 0, 8, 255, 0),
+ SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1),
+ SOC_SINGLE("Playback Gain Enable Switch",
+ SPDIFOUT_CTRL1, 26, 1, 0),
+ SOC_SINGLE("Playback Channels Mix Switch",
+ SPDIFOUT_CTRL0, 23, 1, 0),
+};
+
+static int axg_spdifout_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct axg_spdifout *priv = snd_soc_component_get_drvdata(component);
+ enum snd_soc_bias_level now =
+ snd_soc_component_get_bias_level(component);
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (now == SND_SOC_BIAS_STANDBY)
+ ret = clk_prepare_enable(priv->mclk);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (now == SND_SOC_BIAS_PREPARE)
+ clk_disable_unprepare(priv->mclk);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ case SND_SOC_BIAS_ON:
+ break;
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_component_driver axg_spdifout_component_drv = {
+ .controls = axg_spdifout_controls,
+ .num_controls = ARRAY_SIZE(axg_spdifout_controls),
+ .dapm_widgets = axg_spdifout_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_spdifout_dapm_widgets),
+ .dapm_routes = axg_spdifout_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes),
+ .set_bias_level = axg_spdifout_set_bias_level,
+};
+
+static const struct regmap_config axg_spdifout_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = SPDIFOUT_MUTE_VAL,
+};
+
+static const struct of_device_id axg_spdifout_of_match[] = {
+ { .compatible = "amlogic,axg-spdifout", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, axg_spdifout_of_match);
+
+static int axg_spdifout_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct axg_spdifout *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);
+
+ 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_spdifout_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->mclk = devm_clk_get(dev, "mclk");
+ if (IS_ERR(priv->mclk)) {
+ ret = PTR_ERR(priv->mclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get mclk: %d\n", ret);
+ return ret;
+ }
+
+ return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv,
+ axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv));
+}
+
+static struct platform_driver axg_spdifout_pdrv = {
+ .probe = axg_spdifout_probe,
+ .driver = {
+ .name = "axg-spdifout",
+ .of_match_table = axg_spdifout_of_match,
+ },
+};
+module_platform_driver(axg_spdifout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
new file mode 100644
index 000000000000..43e390f9358a
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -0,0 +1,381 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "axg-tdm-formatter.h"
+
+struct axg_tdm_formatter {
+ struct list_head list;
+ struct axg_tdm_stream *stream;
+ const struct axg_tdm_formatter_driver *drv;
+ struct clk *pclk;
+ struct clk *sclk;
+ struct clk *lrclk;
+ struct clk *sclk_sel;
+ struct clk *lrclk_sel;
+ bool enabled;
+ struct regmap *map;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+ struct axg_tdm_stream *ts,
+ unsigned int offset)
+{
+ unsigned int val, ch = ts->channels;
+ unsigned long mask;
+ int i, j;
+
+ /*
+ * Distribute the channels of the stream over the available slots
+ * of each TDM lane
+ */
+ for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
+ val = 0;
+ mask = ts->mask[i];
+
+ for (j = find_first_bit(&mask, 32);
+ (j < 32) && ch;
+ j = find_next_bit(&mask, 32, j + 1)) {
+ val |= 1 << j;
+ ch -= 1;
+ }
+
+ regmap_write(map, offset, val);
+ offset += regmap_get_reg_stride(map);
+ }
+
+ /*
+ * If we still have channel left at the end of the process, it means
+ * the stream has more channels than we can accommodate and we should
+ * have caught this earlier.
+ */
+ if (WARN_ON(ch != 0)) {
+ pr_err("channel mask error\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
+
+static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
+{
+ struct axg_tdm_stream *ts = formatter->stream;
+ bool invert = formatter->drv->invert_sclk;
+ int ret;
+
+ /* Do nothing if the formatter is already enabled */
+ if (formatter->enabled)
+ return 0;
+
+ /*
+ * If sclk is inverted, invert it back and provide the inversion
+ * required by the formatter
+ */
+ invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
+ ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+ if (ret)
+ return ret;
+
+ /* Setup the stream parameter in the formatter */
+ ret = formatter->drv->ops->prepare(formatter->map, formatter->stream);
+ if (ret)
+ return ret;
+
+ /* Enable the signal clocks feeding the formatter */
+ ret = clk_prepare_enable(formatter->sclk);
+ if (ret)
+ return ret;
+
+ ret = clk_prepare_enable(formatter->lrclk);
+ if (ret) {
+ clk_disable_unprepare(formatter->sclk);
+ return ret;
+ }
+
+ /* Finally, actually enable the formatter */
+ formatter->drv->ops->enable(formatter->map);
+ formatter->enabled = true;
+
+ return 0;
+}
+
+static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter)
+{
+ /* Do nothing if the formatter is already disabled */
+ if (!formatter->enabled)
+ return;
+
+ formatter->drv->ops->disable(formatter->map);
+ clk_disable_unprepare(formatter->lrclk);
+ clk_disable_unprepare(formatter->sclk);
+ formatter->enabled = false;
+}
+
+static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter)
+{
+ struct axg_tdm_stream *ts = formatter->stream;
+ int ret = 0;
+
+ mutex_lock(&ts->lock);
+
+ /* Catch up if the stream is already running when we attach */
+ if (ts->ready) {
+ ret = axg_tdm_formatter_enable(formatter);
+ if (ret) {
+ pr_err("failed to enable formatter\n");
+ goto out;
+ }
+ }
+
+ list_add_tail(&formatter->list, &ts->formatter_list);
+out:
+ mutex_unlock(&ts->lock);
+ return ret;
+}
+
+static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter)
+{
+ struct axg_tdm_stream *ts = formatter->stream;
+
+ mutex_lock(&ts->lock);
+ list_del(&formatter->list);
+ mutex_unlock(&ts->lock);
+
+ axg_tdm_formatter_disable(formatter);
+}
+
+static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter,
+ struct snd_soc_dapm_widget *w)
+{
+ struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w);
+ int ret;
+
+ /*
+ * If we don't get a stream at this stage, it would mean that the
+ * widget is powering up but is not attached to any backend DAI.
+ * It should not happen, ever !
+ */
+ if (WARN_ON(!ts))
+ return -ENODEV;
+
+ /* Clock our device */
+ ret = clk_prepare_enable(formatter->pclk);
+ if (ret)
+ return ret;
+
+ /* Reparent the bit clock to the TDM interface */
+ ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk);
+ if (ret)
+ goto disable_pclk;
+
+ /* Reparent the sample clock to the TDM interface */
+ ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk);
+ if (ret)
+ goto disable_pclk;
+
+ formatter->stream = ts;
+ ret = axg_tdm_formatter_attach(formatter);
+ if (ret)
+ goto disable_pclk;
+
+ return 0;
+
+disable_pclk:
+ clk_disable_unprepare(formatter->pclk);
+ return ret;
+}
+
+static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter)
+{
+ axg_tdm_formatter_dettach(formatter);
+ clk_disable_unprepare(formatter->pclk);
+ formatter->stream = NULL;
+}
+
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *control,
+ int event)
+{
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c);
+ int ret = 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ ret = axg_tdm_formatter_power_up(formatter, w);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ axg_tdm_formatter_power_down(formatter);
+ break;
+
+ default:
+ dev_err(c->dev, "Unexpected event %d\n", event);
+ return -EINVAL;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_event);
+
+int axg_tdm_formatter_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ const struct axg_tdm_formatter_driver *drv;
+ struct axg_tdm_formatter *formatter;
+ struct resource *res;
+ void __iomem *regs;
+ int ret;
+
+ drv = of_device_get_match_data(dev);
+ if (!drv) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
+ formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL);
+ if (!formatter)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, formatter);
+ formatter->drv = drv;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg);
+ if (IS_ERR(formatter->map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(formatter->map));
+ return PTR_ERR(formatter->map);
+ }
+
+ /* Peripharal clock */
+ formatter->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(formatter->pclk)) {
+ ret = PTR_ERR(formatter->pclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get pclk: %d\n", ret);
+ return ret;
+ }
+
+ /* Formatter bit clock */
+ formatter->sclk = devm_clk_get(dev, "sclk");
+ if (IS_ERR(formatter->sclk)) {
+ ret = PTR_ERR(formatter->sclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get sclk: %d\n", ret);
+ return ret;
+ }
+
+ /* Formatter sample clock */
+ formatter->lrclk = devm_clk_get(dev, "lrclk");
+ if (IS_ERR(formatter->lrclk)) {
+ ret = PTR_ERR(formatter->lrclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get lrclk: %d\n", ret);
+ return ret;
+ }
+
+ /* Formatter bit clock input multiplexer */
+ formatter->sclk_sel = devm_clk_get(dev, "sclk_sel");
+ if (IS_ERR(formatter->sclk_sel)) {
+ ret = PTR_ERR(formatter->sclk_sel);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get sclk_sel: %d\n", ret);
+ return ret;
+ }
+
+ /* Formatter sample clock input multiplexer */
+ formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel");
+ if (IS_ERR(formatter->lrclk_sel)) {
+ ret = PTR_ERR(formatter->lrclk_sel);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get lrclk_sel: %d\n", ret);
+ return ret;
+ }
+
+ return devm_snd_soc_register_component(dev, drv->component_drv,
+ NULL, 0);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe);
+
+int axg_tdm_stream_start(struct axg_tdm_stream *ts)
+{
+ struct axg_tdm_formatter *formatter;
+ int ret = 0;
+
+ mutex_lock(&ts->lock);
+ ts->ready = true;
+
+ /* Start all the formatters attached to the stream */
+ list_for_each_entry(formatter, &ts->formatter_list, list) {
+ ret = axg_tdm_formatter_enable(formatter);
+ if (ret) {
+ pr_err("failed to start tdm stream\n");
+ goto out;
+ }
+ }
+
+out:
+ mutex_unlock(&ts->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_start);
+
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts)
+{
+ struct axg_tdm_formatter *formatter;
+
+ mutex_lock(&ts->lock);
+ ts->ready = false;
+
+ /* Stop all the formatters attached to the stream */
+ list_for_each_entry(formatter, &ts->formatter_list, list) {
+ axg_tdm_formatter_disable(formatter);
+ }
+
+ mutex_unlock(&ts->lock);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_stop);
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface)
+{
+ struct axg_tdm_stream *ts;
+
+ ts = kzalloc(sizeof(*ts), GFP_KERNEL);
+ if (ts) {
+ INIT_LIST_HEAD(&ts->formatter_list);
+ mutex_init(&ts->lock);
+ ts->iface = iface;
+ }
+
+ return ts;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc);
+
+void axg_tdm_stream_free(struct axg_tdm_stream *ts)
+{
+ /*
+ * If the list is not empty, it would mean that one of the formatter
+ * widget is still powered and attached to the interface while we
+ * we are removing the TDM DAI. It should not be possible
+ */
+ WARN_ON(!list_empty(&ts->formatter_list));
+ mutex_destroy(&ts->lock);
+ kfree(ts);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_free);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
new file mode 100644
index 000000000000..cf947caf3cb1
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -0,0 +1,39 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_FORMATTER_H
+#define _MESON_AXG_TDM_FORMATTER_H
+
+#include "axg-tdm.h"
+
+struct platform_device;
+struct regmap;
+struct snd_soc_dapm_widget;
+struct snd_kcontrol;
+
+struct axg_tdm_formatter_ops {
+ struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w);
+ void (*enable)(struct regmap *map);
+ void (*disable)(struct regmap *map);
+ int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts);
+};
+
+struct axg_tdm_formatter_driver {
+ const struct snd_soc_component_driver *component_drv;
+ const struct regmap_config *regmap_cfg;
+ const struct axg_tdm_formatter_ops *ops;
+ bool invert_sclk;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+ struct axg_tdm_stream *ts,
+ unsigned int offset);
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *control,
+ int event);
+int axg_tdm_formatter_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_TDM_FORMATTER_H */
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
new file mode 100644
index 000000000000..7b8baf46d968
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -0,0 +1,542 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+enum {
+ TDM_IFACE_PAD,
+ TDM_IFACE_LOOPBACK,
+};
+
+static unsigned int axg_tdm_slots_total(u32 *mask)
+{
+ unsigned int slots = 0;
+ int i;
+
+ if (!mask)
+ return 0;
+
+ /* Count the total number of slots provided by all 4 lanes */
+ for (i = 0; i < AXG_TDM_NUM_LANES; i++)
+ slots += hweight32(mask[i]);
+
+ return slots;
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+ u32 *rx_mask, unsigned int slots,
+ unsigned int slot_width)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ struct axg_tdm_stream *tx = (struct axg_tdm_stream *)
+ dai->playback_dma_data;
+ struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
+ dai->capture_dma_data;
+ unsigned int tx_slots, rx_slots;
+
+ tx_slots = axg_tdm_slots_total(tx_mask);
+ rx_slots = axg_tdm_slots_total(rx_mask);
+
+ /* We should at least have a slot for a valid interface */
+ if (!tx_slots && !rx_slots) {
+ dev_err(dai->dev, "interface has no slot\n");
+ 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:
+ case 32:
+ iface->slot_width = slot_width;
+ break;
+ default:
+ dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
+
+static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ int ret = -ENOTSUPP;
+
+ if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) {
+ if (!iface->mclk) {
+ dev_warn(dai->dev, "master clock not provided\n");
+ } else {
+ ret = clk_set_rate(iface->mclk, freq);
+ if (!ret)
+ iface->mclk_rate = freq;
+ }
+ }
+
+ return ret;
+}
+
+static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+ /* These modes are not supported */
+ if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+ dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+ return -EINVAL;
+ }
+
+ /* If the TDM interface is the clock master, it requires mclk */
+ if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
+ dev_err(dai->dev, "cpu clock master: mclk missing\n");
+ return -ENODEV;
+ }
+
+ iface->fmt = fmt;
+ return 0;
+}
+
+static int axg_tdm_iface_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ struct axg_tdm_stream *ts =
+ snd_soc_dai_get_dma_data(dai, substream);
+ int ret;
+
+ if (!axg_tdm_slots_total(ts->mask)) {
+ dev_err(dai->dev, "interface has not slots\n");
+ return -EINVAL;
+ }
+
+ /* Apply component wide rate symmetry */
+ if (dai->component->active) {
+ ret = snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ iface->rate);
+ if (ret < 0) {
+ dev_err(dai->dev,
+ "can't set iface rate constraint\n");
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+ unsigned int channels = params_channels(params);
+ unsigned int width = params_width(params);
+
+ /* Save rate and sample_bits for component symmetry */
+ iface->rate = params_rate(params);
+
+ /* Make sure this interface can cope with the stream */
+ if (axg_tdm_slots_total(ts->mask) < channels) {
+ dev_err(dai->dev, "not enough slots for channels\n");
+ return -EINVAL;
+ }
+
+ if (iface->slot_width < width) {
+ dev_err(dai->dev, "incompatible slots width for stream\n");
+ return -EINVAL;
+ }
+
+ /* Save the parameter for tdmout/tdmin widgets */
+ ts->physical_width = params_physical_width(params);
+ ts->width = params_width(params);
+ ts->channels = params_channels(params);
+
+ return 0;
+}
+
+static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai,
+ struct snd_pcm_hw_params *params)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ unsigned int ratio_num;
+ int ret;
+
+ ret = clk_set_rate(iface->lrclk, params_rate(params));
+ if (ret) {
+ dev_err(dai->dev, "setting sample clock failed: %d\n", ret);
+ return ret;
+ }
+
+ switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ /* 50% duty cycle ratio */
+ ratio_num = 1;
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /*
+ * A zero duty cycle ratio will result in setting the mininum
+ * ratio possible which, for this clock, is 1 cycle of the
+ * parent bclk clock high and the rest low, This is exactly
+ * what we want here.
+ */
+ ratio_num = 0;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2);
+ if (ret) {
+ dev_err(dai->dev,
+ "setting sample clock duty cycle failed: %d\n", ret);
+ return ret;
+ }
+
+ /* Set sample clock inversion */
+ ret = clk_set_phase(iface->lrclk,
+ axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0);
+ if (ret) {
+ dev_err(dai->dev,
+ "setting sample clock phase failed: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai,
+ struct snd_pcm_hw_params *params)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ unsigned long srate;
+ int ret;
+
+ srate = iface->slots * iface->slot_width * params_rate(params);
+
+ if (!iface->mclk_rate) {
+ /* If no specific mclk is requested, default to bit clock * 4 */
+ clk_set_rate(iface->mclk, 4 * srate);
+ } else {
+ /* Check if we can actually get the bit clock from mclk */
+ if (iface->mclk_rate % srate) {
+ dev_err(dai->dev,
+ "can't derive sclk %lu from mclk %lu\n",
+ srate, iface->mclk_rate);
+ return -EINVAL;
+ }
+ }
+
+ ret = clk_set_rate(iface->sclk, srate);
+ if (ret) {
+ dev_err(dai->dev, "setting bit clock failed: %d\n", ret);
+ return ret;
+ }
+
+ /* Set the bit clock inversion */
+ ret = clk_set_phase(iface->sclk,
+ axg_tdm_sclk_invert(iface->fmt) ? 0 : 180);
+ if (ret) {
+ dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ if (iface->slots > 2) {
+ dev_err(dai->dev, "bad slot number for format: %d\n",
+ iface->slots);
+ return -EINVAL;
+ }
+ break;
+
+ case SND_SOC_DAI_FORMAT_DSP_A:
+ case SND_SOC_DAI_FORMAT_DSP_B:
+ break;
+
+ default:
+ dev_err(dai->dev, "unsupported dai format\n");
+ return -EINVAL;
+ }
+
+ ret = axg_tdm_iface_set_stream(substream, params, dai);
+ if (ret)
+ return ret;
+
+ if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+ ret = axg_tdm_iface_set_sclk(dai, params);
+ if (ret)
+ return ret;
+
+ ret = axg_tdm_iface_set_lrclk(dai, params);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+ /* Stop all attached formatters */
+ axg_tdm_stream_stop(ts);
+
+ return 0;
+}
+
+static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+ /* Force all attached formatters to update */
+ return axg_tdm_stream_reset(ts);
+}
+
+static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai)
+{
+ if (dai->capture_dma_data)
+ axg_tdm_stream_free(dai->capture_dma_data);
+
+ if (dai->playback_dma_data)
+ axg_tdm_stream_free(dai->playback_dma_data);
+
+ return 0;
+}
+
+static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai)
+{
+ struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+ if (dai->capture_widget) {
+ dai->capture_dma_data = axg_tdm_stream_alloc(iface);
+ if (!dai->capture_dma_data)
+ return -ENOMEM;
+ }
+
+ if (dai->playback_widget) {
+ dai->playback_dma_data = axg_tdm_stream_alloc(iface);
+ if (!dai->playback_dma_data) {
+ axg_tdm_iface_remove_dai(dai);
+ return -ENOMEM;
+ }
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops axg_tdm_iface_ops = {
+ .set_sysclk = axg_tdm_iface_set_sysclk,
+ .set_fmt = axg_tdm_iface_set_fmt,
+ .startup = axg_tdm_iface_startup,
+ .hw_params = axg_tdm_iface_hw_params,
+ .prepare = axg_tdm_iface_prepare,
+ .hw_free = axg_tdm_iface_hw_free,
+};
+
+/* TDM Backend DAIs */
+static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = {
+ [TDM_IFACE_PAD] = {
+ .name = "TDM Pad",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = AXG_TDM_CHANNEL_MAX,
+ .rates = AXG_TDM_RATES,
+ .formats = AXG_TDM_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = AXG_TDM_CHANNEL_MAX,
+ .rates = AXG_TDM_RATES,
+ .formats = AXG_TDM_FORMATS,
+ },
+ .id = TDM_IFACE_PAD,
+ .ops = &axg_tdm_iface_ops,
+ .probe = axg_tdm_iface_probe_dai,
+ .remove = axg_tdm_iface_remove_dai,
+ },
+ [TDM_IFACE_LOOPBACK] = {
+ .name = "TDM Loopback",
+ .capture = {
+ .stream_name = "Loopback",
+ .channels_min = 1,
+ .channels_max = AXG_TDM_CHANNEL_MAX,
+ .rates = AXG_TDM_RATES,
+ .formats = AXG_TDM_FORMATS,
+ },
+ .id = TDM_IFACE_LOOPBACK,
+ .ops = &axg_tdm_iface_ops,
+ .probe = axg_tdm_iface_probe_dai,
+ .remove = axg_tdm_iface_remove_dai,
+ },
+};
+
+static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component);
+ enum snd_soc_bias_level now =
+ snd_soc_component_get_bias_level(component);
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (now == SND_SOC_BIAS_STANDBY)
+ ret = clk_prepare_enable(iface->mclk);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (now == SND_SOC_BIAS_PREPARE)
+ clk_disable_unprepare(iface->mclk);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ case SND_SOC_BIAS_ON:
+ break;
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
+ .set_bias_level = axg_tdm_iface_set_bias_level,
+};
+
+static const struct of_device_id axg_tdm_iface_of_match[] = {
+ { .compatible = "amlogic,axg-tdm-iface", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match);
+
+static int axg_tdm_iface_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct snd_soc_dai_driver *dai_drv;
+ struct axg_tdm_iface *iface;
+ int ret, i;
+
+ iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL);
+ if (!iface)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, iface);
+
+ /*
+ * Duplicate dai driver: depending on the slot masks configuration
+ * We'll change the number of channel provided by DAI stream, so dpcm
+ * channel merge can be done properly
+ */
+ dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv),
+ sizeof(*dai_drv), GFP_KERNEL);
+ if (!dai_drv)
+ return -ENOMEM;
+
+ for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++)
+ memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i],
+ sizeof(*dai_drv));
+
+ /* Bit clock provided on the pad */
+ iface->sclk = devm_clk_get(dev, "sclk");
+ if (IS_ERR(iface->sclk)) {
+ ret = PTR_ERR(iface->sclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get sclk: %d\n", ret);
+ return ret;
+ }
+
+ /* Sample clock provided on the pad */
+ iface->lrclk = devm_clk_get(dev, "lrclk");
+ if (IS_ERR(iface->lrclk)) {
+ ret = PTR_ERR(iface->lrclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get lrclk: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * mclk maybe be missing when the cpu dai is in slave mode and
+ * the codec does not require it to provide a master clock.
+ * At this point, ignore the error if mclk is missing. We'll
+ * throw an error if the cpu dai is master and mclk is missing
+ */
+ iface->mclk = devm_clk_get(dev, "mclk");
+ if (IS_ERR(iface->mclk)) {
+ ret = PTR_ERR(iface->mclk);
+ if (ret == -ENOENT) {
+ iface->mclk = NULL;
+ } else {
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get mclk: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return devm_snd_soc_register_component(dev,
+ &axg_tdm_iface_component_drv, dai_drv,
+ ARRAY_SIZE(axg_tdm_iface_dai_drv));
+}
+
+static struct platform_driver axg_tdm_iface_pdrv = {
+ .probe = axg_tdm_iface_probe,
+ .driver = {
+ .name = "axg-tdm-iface",
+ .of_match_table = axg_tdm_iface_of_match,
+ },
+};
+module_platform_driver(axg_tdm_iface_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM interface driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
new file mode 100644
index 000000000000..e578b6f40a07
--- /dev/null
+++ b/sound/soc/meson/axg-tdm.h
@@ -0,0 +1,78 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_H
+#define _MESON_AXG_TDM_H
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#define AXG_TDM_NUM_LANES 4
+#define AXG_TDM_CHANNEL_MAX 128
+#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \
+ SNDRV_PCM_RATE_8000_192000)
+#define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+struct axg_tdm_iface {
+ struct clk *sclk;
+ struct clk *lrclk;
+ struct clk *mclk;
+ unsigned long mclk_rate;
+
+ /* format is common to all the DAIs of the iface */
+ unsigned int fmt;
+ unsigned int slots;
+ unsigned int slot_width;
+
+ /* For component wide symmetry */
+ int rate;
+};
+
+static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
+{
+ return (fmt & SND_SOC_DAIFMT_I2S) ^
+ !!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
+}
+
+static inline bool axg_tdm_sclk_invert(unsigned int fmt)
+{
+ return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF);
+}
+
+struct axg_tdm_stream {
+ struct axg_tdm_iface *iface;
+ struct list_head formatter_list;
+ struct mutex lock;
+ unsigned int channels;
+ unsigned int width;
+ unsigned int physical_width;
+ u32 *mask;
+ bool ready;
+};
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface);
+void axg_tdm_stream_free(struct axg_tdm_stream *ts);
+int axg_tdm_stream_start(struct axg_tdm_stream *ts);
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts);
+
+static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts)
+{
+ axg_tdm_stream_stop(ts);
+ return axg_tdm_stream_start(ts);
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+ u32 *rx_mask, unsigned int slots,
+ unsigned int slot_width);
+
+#endif /* _MESON_AXG_TDM_H */
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
new file mode 100644
index 000000000000..bbac44c81688
--- /dev/null
+++ b/sound/soc/meson/axg-tdmin.c
@@ -0,0 +1,229 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMIN_CTRL 0x00
+#define TDMIN_CTRL_ENABLE BIT(31)
+#define TDMIN_CTRL_I2S_MODE BIT(30)
+#define TDMIN_CTRL_RST_OUT BIT(29)
+#define TDMIN_CTRL_RST_IN BIT(28)
+#define TDMIN_CTRL_WS_INV BIT(25)
+#define TDMIN_CTRL_SEL_SHIFT 20
+#define TDMIN_CTRL_IN_BIT_SKEW_MASK GENMASK(18, 16)
+#define TDMIN_CTRL_IN_BIT_SKEW(x) ((x) << 16)
+#define TDMIN_CTRL_LSB_FIRST BIT(5)
+#define TDMIN_CTRL_BITNUM_MASK GENMASK(4, 0)
+#define TDMIN_CTRL_BITNUM(x) ((x) << 0)
+#define TDMIN_SWAP 0x04
+#define TDMIN_MASK0 0x08
+#define TDMIN_MASK1 0x0c
+#define TDMIN_MASK2 0x10
+#define TDMIN_MASK3 0x14
+#define TDMIN_STAT 0x18
+#define TDMIN_MUTE_VAL 0x1c
+#define TDMIN_MUTE0 0x20
+#define TDMIN_MUTE1 0x24
+#define TDMIN_MUTE2 0x28
+#define TDMIN_MUTE3 0x2c
+
+static const struct regmap_config axg_tdmin_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = TDMIN_MUTE3,
+};
+
+static const char * const axg_tdmin_sel_texts[] = {
+ "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5",
+};
+
+/* Change to special mux control to reset dapm */
+static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL,
+ TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmin_in_mux =
+ SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmin_get_be(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *p = NULL;
+ struct snd_soc_dai *be;
+
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ if (p->source->id == snd_soc_dapm_dai_out)
+ return (struct snd_soc_dai *)p->source->priv;
+
+ be = axg_tdmin_get_be(p->source);
+ if (be)
+ return be;
+ }
+
+ return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dai *be = axg_tdmin_get_be(w);
+
+ if (!be)
+ return NULL;
+
+ return be->capture_dma_data;
+}
+
+static void axg_tdmin_enable(struct regmap *map)
+{
+ /* Apply both reset */
+ regmap_update_bits(map, TDMIN_CTRL,
+ TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0);
+
+ /* Clear out reset before in reset */
+ regmap_update_bits(map, TDMIN_CTRL,
+ TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT);
+ regmap_update_bits(map, TDMIN_CTRL,
+ TDMIN_CTRL_RST_IN, TDMIN_CTRL_RST_IN);
+
+ /* Actually enable tdmin */
+ regmap_update_bits(map, TDMIN_CTRL,
+ TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE);
+}
+
+static void axg_tdmin_disable(struct regmap *map)
+{
+ regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0);
+}
+
+static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+ unsigned int val = 0;
+
+ /* Set stream skew */
+ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_DSP_A:
+ val |= TDMIN_CTRL_IN_BIT_SKEW(3);
+ break;
+
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_DSP_B:
+ val = TDMIN_CTRL_IN_BIT_SKEW(2);
+ break;
+
+ default:
+ pr_err("Unsupported format: %u\n",
+ ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ return -EINVAL;
+ }
+
+ /* Set stream format mode */
+ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val |= TDMIN_CTRL_I2S_MODE;
+ break;
+ }
+
+ /* If the sample clock is inverted, invert it back for the formatter */
+ if (axg_tdm_lrclk_invert(ts->iface->fmt))
+ val |= TDMIN_CTRL_WS_INV;
+
+ /* Set the slot width */
+ val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1);
+
+ /*
+ * The following also reset LSB_FIRST which result in the formatter
+ * placing the first bit received at bit 31
+ */
+ regmap_update_bits(map, TDMIN_CTRL,
+ (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV |
+ TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST |
+ TDMIN_CTRL_BITNUM_MASK), val);
+
+ /* Set static swap mask configuration */
+ regmap_write(map, TDMIN_SWAP, 0x76543210);
+
+ return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux),
+ SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0,
+ axg_tdm_formatter_event,
+ (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+ SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = {
+ { "SRC SEL", "IN 0", "IN 0" },
+ { "SRC SEL", "IN 1", "IN 1" },
+ { "SRC SEL", "IN 2", "IN 2" },
+ { "SRC SEL", "IN 3", "IN 3" },
+ { "SRC SEL", "IN 4", "IN 4" },
+ { "SRC SEL", "IN 5", "IN 5" },
+ { "DEC", NULL, "SRC SEL" },
+ { "OUT", NULL, "DEC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmin_component_drv = {
+ .dapm_widgets = axg_tdmin_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_tdmin_dapm_widgets),
+ .dapm_routes = axg_tdmin_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_tdmin_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmin_ops = {
+ .get_stream = axg_tdmin_get_tdm_stream,
+ .prepare = axg_tdmin_prepare,
+ .enable = axg_tdmin_enable,
+ .disable = axg_tdmin_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
+ .component_drv = &axg_tdmin_component_drv,
+ .regmap_cfg = &axg_tdmin_regmap_cfg,
+ .ops = &axg_tdmin_ops,
+ .invert_sclk = false,
+};
+
+static const struct of_device_id axg_tdmin_of_match[] = {
+ {
+ .compatible = "amlogic,axg-tdmin",
+ .data = &axg_tdmin_drv,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
+
+static struct platform_driver axg_tdmin_pdrv = {
+ .probe = axg_tdm_formatter_probe,
+ .driver = {
+ .name = "axg-tdmin",
+ .of_match_table = axg_tdmin_of_match,
+ },
+};
+module_platform_driver(axg_tdmin_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
new file mode 100644
index 000000000000..f73368ee1088
--- /dev/null
+++ b/sound/soc/meson/axg-tdmout.c
@@ -0,0 +1,259 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMOUT_CTRL0 0x00
+#define TDMOUT_CTRL0_BITNUM_MASK GENMASK(4, 0)
+#define TDMOUT_CTRL0_BITNUM(x) ((x) << 0)
+#define TDMOUT_CTRL0_SLOTNUM_MASK GENMASK(9, 5)
+#define TDMOUT_CTRL0_SLOTNUM(x) ((x) << 5)
+#define TDMOUT_CTRL0_INIT_BITNUM_MASK GENMASK(19, 15)
+#define TDMOUT_CTRL0_INIT_BITNUM(x) ((x) << 15)
+#define TDMOUT_CTRL0_ENABLE BIT(31)
+#define TDMOUT_CTRL0_RST_OUT BIT(29)
+#define TDMOUT_CTRL0_RST_IN BIT(28)
+#define TDMOUT_CTRL1 0x04
+#define TDMOUT_CTRL1_TYPE_MASK GENMASK(6, 4)
+#define TDMOUT_CTRL1_TYPE(x) ((x) << 4)
+#define TDMOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8)
+#define TDMOUT_CTRL1_MSB_POS(x) ((x) << 8)
+#define TDMOUT_CTRL1_SEL_SHIFT 24
+#define TDMOUT_CTRL1_GAIN_EN 26
+#define TDMOUT_CTRL1_WS_INV BIT(28)
+#define TDMOUT_SWAP 0x08
+#define TDMOUT_MASK0 0x0c
+#define TDMOUT_MASK1 0x10
+#define TDMOUT_MASK2 0x14
+#define TDMOUT_MASK3 0x18
+#define TDMOUT_STAT 0x1c
+#define TDMOUT_GAIN0 0x20
+#define TDMOUT_GAIN1 0x24
+#define TDMOUT_MUTE_VAL 0x28
+#define TDMOUT_MUTE0 0x2c
+#define TDMOUT_MUTE1 0x30
+#define TDMOUT_MUTE2 0x34
+#define TDMOUT_MUTE3 0x38
+#define TDMOUT_MASK_VAL 0x3c
+
+static const struct regmap_config axg_tdmout_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = TDMOUT_MASK_VAL,
+};
+
+static const struct snd_kcontrol_new axg_tdmout_controls[] = {
+ SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0, 0, 8, 255, 0),
+ SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
+ SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1, 0, 8, 255, 0),
+ SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
+ SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
+ TDMOUT_CTRL1_GAIN_EN, 1, 0),
+};
+
+static const char * const tdmout_sel_texts[] = {
+ "IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1,
+ TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmout_in_mux =
+ SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmout_get_be(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *p = NULL;
+ struct snd_soc_dai *be;
+
+ snd_soc_dapm_widget_for_each_sink_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ if (p->sink->id == snd_soc_dapm_dai_in)
+ return (struct snd_soc_dai *)p->sink->priv;
+
+ be = axg_tdmout_get_be(p->sink);
+ if (be)
+ return be;
+ }
+
+ return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dai *be = axg_tdmout_get_be(w);
+
+ if (!be)
+ return NULL;
+
+ return be->playback_dma_data;
+}
+
+static void axg_tdmout_enable(struct regmap *map)
+{
+ /* Apply both reset */
+ regmap_update_bits(map, TDMOUT_CTRL0,
+ TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0);
+
+ /* Clear out reset before in reset */
+ regmap_update_bits(map, TDMOUT_CTRL0,
+ TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT);
+ regmap_update_bits(map, TDMOUT_CTRL0,
+ TDMOUT_CTRL0_RST_IN, TDMOUT_CTRL0_RST_IN);
+
+ /* Actually enable tdmout */
+ regmap_update_bits(map, TDMOUT_CTRL0,
+ TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE);
+}
+
+static void axg_tdmout_disable(struct regmap *map)
+{
+ regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0);
+}
+
+static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+ unsigned int val = 0;
+
+ /* Set the stream skew */
+ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_DSP_A:
+ val |= TDMOUT_CTRL0_INIT_BITNUM(1);
+ break;
+
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_DSP_B:
+ val |= TDMOUT_CTRL0_INIT_BITNUM(2);
+ break;
+
+ default:
+ pr_err("Unsupported format: %u\n",
+ ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ return -EINVAL;
+ }
+
+ /* Set the slot width */
+ val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1);
+
+ /* Set the slot number */
+ val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1);
+
+ regmap_update_bits(map, TDMOUT_CTRL0,
+ TDMOUT_CTRL0_INIT_BITNUM_MASK |
+ TDMOUT_CTRL0_BITNUM_MASK |
+ TDMOUT_CTRL0_SLOTNUM_MASK, val);
+
+ /* Set the sample width */
+ val = TDMOUT_CTRL1_MSB_POS(ts->width - 1);
+
+ /* FIFO data are arranged in chunks of 64bits */
+ switch (ts->physical_width) {
+ case 8:
+ /* 8 samples of 8 bits */
+ val |= TDMOUT_CTRL1_TYPE(0);
+ break;
+ case 16:
+ /* 4 samples of 16 bits - right justified */
+ val |= TDMOUT_CTRL1_TYPE(2);
+ break;
+ case 32:
+ /* 2 samples of 32 bits - right justified */
+ val |= TDMOUT_CTRL1_TYPE(4);
+ break;
+ default:
+ pr_err("Unsupported physical width: %u\n",
+ ts->physical_width);
+ return -EINVAL;
+ }
+
+ /* If the sample clock is inverted, invert it back for the formatter */
+ if (axg_tdm_lrclk_invert(ts->iface->fmt))
+ val |= TDMOUT_CTRL1_WS_INV;
+
+ regmap_update_bits(map, TDMOUT_CTRL1,
+ (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK |
+ TDMOUT_CTRL1_WS_INV), val);
+
+ /* Set static swap mask configuration */
+ regmap_write(map, TDMOUT_SWAP, 0x76543210);
+
+ return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux),
+ SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0,
+ axg_tdm_formatter_event,
+ (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+ SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = {
+ { "SRC SEL", "IN 0", "IN 0" },
+ { "SRC SEL", "IN 1", "IN 1" },
+ { "SRC SEL", "IN 2", "IN 2" },
+ { "ENC", NULL, "SRC SEL" },
+ { "OUT", NULL, "ENC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmout_component_drv = {
+ .controls = axg_tdmout_controls,
+ .num_controls = ARRAY_SIZE(axg_tdmout_controls),
+ .dapm_widgets = axg_tdmout_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_tdmout_dapm_widgets),
+ .dapm_routes = axg_tdmout_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_tdmout_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmout_ops = {
+ .get_stream = axg_tdmout_get_tdm_stream,
+ .prepare = axg_tdmout_prepare,
+ .enable = axg_tdmout_enable,
+ .disable = axg_tdmout_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
+ .component_drv = &axg_tdmout_component_drv,
+ .regmap_cfg = &axg_tdmout_regmap_cfg,
+ .ops = &axg_tdmout_ops,
+ .invert_sclk = true,
+};
+
+static const struct of_device_id axg_tdmout_of_match[] = {
+ {
+ .compatible = "amlogic,axg-tdmout",
+ .data = &axg_tdmout_drv,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmout_of_match);
+
+static struct platform_driver axg_tdmout_pdrv = {
+ .probe = axg_tdm_formatter_probe,
+ .driver = {
+ .name = "axg-tdmout",
+ .of_match_table = axg_tdmout_of_match,
+ },
+};
+module_platform_driver(axg_tdmout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
new file mode 100644
index 000000000000..c2c9bb312586
--- /dev/null
+++ b/sound/soc/meson/axg-toddr.c
@@ -0,0 +1,199 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend capture DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_TODDR_SEL_RESAMPLE BIT(30)
+#define CTRL0_TODDR_EXT_SIGNED BIT(29)
+#define CTRL0_TODDR_PP_MODE BIT(28)
+#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13)
+#define CTRL0_TODDR_TYPE(x) ((x) << 13)
+#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8)
+#define CTRL0_TODDR_MSB_POS(x) ((x) << 8)
+#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3)
+#define CTRL0_TODDR_LSB_POS(x) ((x) << 3)
+
+static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *dai)
+{
+ return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+ unsigned int type, width, msb = 31;
+
+ /*
+ * NOTE:
+ * Almost all backend will place the MSB at bit 31, except SPDIF Input
+ * which will put it at index 28. When adding support for the SPDIF
+ * Input, we'll need to find which type of backend we are connected to.
+ */
+
+ switch (params_physical_width(params)) {
+ case 8:
+ type = 0; /* 8 samples of 8 bits */
+ break;
+ case 16:
+ type = 2; /* 4 samples of 16 bits - right justified */
+ break;
+ case 32:
+ type = 4; /* 2 samples of 32 bits - right justified */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ width = params_width(params);
+
+ regmap_update_bits(fifo->map, FIFO_CTRL0,
+ CTRL0_TODDR_TYPE_MASK |
+ CTRL0_TODDR_MSB_POS_MASK |
+ CTRL0_TODDR_LSB_POS_MASK,
+ CTRL0_TODDR_TYPE(type) |
+ CTRL0_TODDR_MSB_POS(msb) |
+ CTRL0_TODDR_LSB_POS(msb - (width - 1)));
+
+ return 0;
+}
+
+static int axg_toddr_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+ unsigned int fifo_threshold;
+ int ret;
+
+ /* Enable pclk to access registers and clock the fifo ip */
+ ret = clk_prepare_enable(fifo->pclk);
+ if (ret)
+ return ret;
+
+ /* Select orginal data - resampling not supported ATM */
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0);
+
+ /* Only signed format are supported ATM */
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED,
+ CTRL0_TODDR_EXT_SIGNED);
+
+ /* Apply single buffer mode to the interface */
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0);
+
+ /* TODDR does not have a configurable fifo depth */
+ fifo_threshold = AXG_FIFO_MIN_CNT - 1;
+ regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK,
+ CTRL1_THRESHOLD(fifo_threshold));
+
+ return 0;
+}
+
+static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable_unprepare(fifo->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_toddr_ops = {
+ .hw_params = axg_toddr_dai_hw_params,
+ .startup = axg_toddr_dai_startup,
+ .shutdown = axg_toddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_toddr_dai_drv = {
+ .name = "TODDR",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = AXG_FIFO_CH_MAX,
+ .rates = AXG_FIFO_RATES,
+ .formats = AXG_FIFO_FORMATS,
+ },
+ .ops = &axg_toddr_ops,
+ .pcm_new = axg_toddr_pcm_new,
+};
+
+static const char * const axg_toddr_sel_texts[] = {
+ "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6"
+};
+
+static const unsigned int axg_toddr_sel_values[] = {
+ 0, 1, 2, 3, 4, 6
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0,
+ CTRL0_SEL_SHIFT, CTRL0_SEL_MASK,
+ axg_toddr_sel_texts, axg_toddr_sel_values);
+
+static const struct snd_kcontrol_new axg_toddr_in_mux =
+ SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux),
+ SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = {
+ { "Capture", NULL, "SRC SEL" },
+ { "SRC SEL", "IN 0", "IN 0" },
+ { "SRC SEL", "IN 1", "IN 1" },
+ { "SRC SEL", "IN 2", "IN 2" },
+ { "SRC SEL", "IN 3", "IN 3" },
+ { "SRC SEL", "IN 4", "IN 4" },
+ { "SRC SEL", "IN 6", "IN 6" },
+};
+
+static const struct snd_soc_component_driver axg_toddr_component_drv = {
+ .dapm_widgets = axg_toddr_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets),
+ .dapm_routes = axg_toddr_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes),
+ .ops = &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_toddr_match_data = {
+ .component_drv = &axg_toddr_component_drv,
+ .dai_drv = &axg_toddr_dai_drv
+};
+
+static const struct of_device_id axg_toddr_of_match[] = {
+ {
+ .compatible = "amlogic,axg-toddr",
+ .data = &axg_toddr_match_data,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_toddr_of_match);
+
+static struct platform_driver axg_toddr_pdrv = {
+ .probe = axg_fifo_probe,
+ .driver = {
+ .name = "axg-toddr",
+ .of_match_table = axg_toddr_of_match,
+ },
+};
+module_platform_driver(axg_toddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG capture fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 15ccbf479c96..d5ae9eb8c756 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -40,7 +40,7 @@ struct abe_twl6040 {
int mclk_freq; /* MCLK frequency speed for twl6040 */
};
-struct platform_device *dmic_codec_dev;
+static struct platform_device *dmic_codec_dev;
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 51dd7c65096b..fe966272bd0c 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
switch (channels) {
case 6:
dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
+ /* fall through */
case 4:
dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
+ /* fall through */
case 2:
dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
break;
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 0e97360f9890..4c1be36c2207 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 4;
+ /* fall through */
case 4:
if (stream == SNDRV_PCM_STREAM_CAPTURE)
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 3;
+ /* fall through */
case 3:
link_mask |= 1 << 2;
+ /* fall through */
case 2:
link_mask |= 1 << 1;
+ /* fall through */
case 1:
link_mask |= 1 << 0;
break;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 960744e46edc..776e148b0aa2 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -24,15 +24,19 @@ config SND_PXA2XX_AC97
config SND_PXA2XX_SOC_AC97
tristate
select AC97_BUS
+ select SND_PXA2XX_LIB
select SND_PXA2XX_LIB_AC97
select SND_SOC_AC97_BUS
config SND_PXA2XX_SOC_I2S
+ select SND_PXA2XX_LIB
tristate
config SND_PXA_SOC_SSP
- tristate
+ tristate "Soc Audio via PXA2xx/PXA3xx SSP ports"
+ depends on PLAT_PXA
select PXA_SSP
+ select SND_PXA2XX_LIB
config SND_MMP_SOC_SSPA
tristate
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 2fc012b06c43..935a248e5bf6 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- unsigned int acps, acds, width;
- unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+ unsigned int width;
int ret = 0;
- width = snd_pcm_format_physical_width(params_format(params));
-
- /*
- * rate = SSPSCLK / (2 * width(16 or 32))
- * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
- */
- switch (params_rate(params)) {
- case 8000:
- /* off by a factor of 2: bug in the PXA27x audio clock? */
- acps = 32842000;
- switch (width) {
- case 16:
- /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_16;
- break;
- default: /* 32 */
- /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_8;
- }
- break;
- case 11025:
- acps = 5622000;
- switch (width) {
- case 16:
- /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_4;
- break;
- default: /* 32 */
- /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_2;
- }
- break;
- case 22050:
- acps = 5622000;
- switch (width) {
- case 16:
- /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_2;
- break;
- default: /* 32 */
- /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_1;
- }
- break;
- case 44100:
- acps = 5622000;
- switch (width) {
- case 16:
- /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_2;
- break;
- default: /* 32 */
- /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_1;
- }
- break;
- case 48000:
- acps = 12235000;
- switch (width) {
- case 16:
- /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_2;
- break;
- default: /* 32 */
- /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_1;
- }
- break;
- case 96000:
- default:
- acps = 12235000;
- switch (width) {
- case 16:
- /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_1;
- break;
- default: /* 32 */
- /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
- acds = PXA_SSP_CLK_AUDIO_DIV_2;
- div4 = PXA_SSP_CLK_SCDB_1;
- break;
- }
- break;
- }
-
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
@@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
+ width = snd_pcm_format_physical_width(params_format(params));
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
if (ret < 0)
return ret;
@@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- /* set the SSP audio system clock ACDS divider */
- ret = snd_soc_dai_set_clkdiv(cpu_dai,
- PXA_SSP_AUDIO_DIV_ACDS, acds);
- if (ret < 0)
- return ret;
-
- /* set the SSP audio system clock SCDB divider4 */
- ret = snd_soc_dai_set_clkdiv(cpu_dai,
- PXA_SSP_AUDIO_DIV_SCDB, div4);
- if (ret < 0)
- return ret;
-
- /* set SSP audio pll clock */
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
- if (ret < 0)
- return ret;
-
return 0;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6fc986080130..69033e1a84e6 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -34,7 +34,6 @@
#include <sound/pxa2xx-lib.h>
#include <sound/dmaengine_pcm.h>
-#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
/*
@@ -42,6 +41,8 @@
*/
struct ssp_priv {
struct ssp_device *ssp;
+ struct clk *extclk;
+ unsigned long ssp_clk;
unsigned int sysclk;
unsigned int dai_fmt;
unsigned int configured_dai_fmt;
@@ -105,9 +106,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
if (!dma)
return -ENOMEM;
-
- dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- &ssp->drcmr_tx : &ssp->drcmr_rx;
+ dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "tx" : "rx";
snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
@@ -194,21 +194,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
pxa_ssp_write_reg(ssp, SSCR0, sscr0);
}
-/**
- * pxa_ssp_get_clkdiv - get SSP clock divider
- */
-static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
-{
- u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
- u32 div;
-
- if (ssp->type == PXA25x_SSP)
- div = ((sscr0 >> 8) & 0xff) * 2 + 2;
- else
- div = ((sscr0 >> 8) & 0xfff) + 1;
- return div;
-}
-
/*
* Set the SSP ports SYSCLK.
*/
@@ -221,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+ if (priv->extclk) {
+ int ret;
+
+ /*
+ * For DT based boards, if an extclk is given, use it
+ * here and configure PXA_SSP_CLK_EXT.
+ */
+
+ ret = clk_set_rate(priv->extclk, freq);
+ if (ret < 0)
+ return ret;
+
+ clk_id = PXA_SSP_CLK_EXT;
+ }
+
dev_dbg(&ssp->pdev->dev,
"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
cpu_dai->id, clk_id, freq);
@@ -265,66 +265,17 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
}
/*
- * Set the SSP clock dividers.
- */
-static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
- int div_id, int div)
-{
- struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
- struct ssp_device *ssp = priv->ssp;
- int val;
-
- switch (div_id) {
- case PXA_SSP_AUDIO_DIV_ACDS:
- val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
- pxa_ssp_write_reg(ssp, SSACD, val);
- break;
- case PXA_SSP_AUDIO_DIV_SCDB:
- val = pxa_ssp_read_reg(ssp, SSACD);
- val &= ~SSACD_SCDB;
- if (ssp->type == PXA3xx_SSP)
- val &= ~SSACD_SCDX8;
- switch (div) {
- case PXA_SSP_CLK_SCDB_1:
- val |= SSACD_SCDB;
- break;
- case PXA_SSP_CLK_SCDB_4:
- break;
- case PXA_SSP_CLK_SCDB_8:
- if (ssp->type == PXA3xx_SSP)
- val |= SSACD_SCDX8;
- else
- return -EINVAL;
- break;
- default:
- return -EINVAL;
- }
- pxa_ssp_write_reg(ssp, SSACD, val);
- break;
- case PXA_SSP_DIV_SCR:
- pxa_ssp_set_scr(ssp, div);
- break;
- default:
- return -ENODEV;
- }
-
- return 0;
-}
-
-/*
* Configure the PLL frequency pxa27x and (afaik - pxa320 only)
*/
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
- int source, unsigned int freq_in, unsigned int freq_out)
+static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq)
{
- struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
struct ssp_device *ssp = priv->ssp;
u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
if (ssp->type == PXA3xx_SSP)
pxa_ssp_write_reg(ssp, SSACDD, 0);
- switch (freq_out) {
+ switch (freq) {
case 5622000:
break;
case 11345000:
@@ -355,7 +306,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
u64 tmp = 19968;
tmp *= 1000000;
- do_div(tmp, freq_out);
+ do_div(tmp, freq);
val = tmp;
val = (val << 16) | 64;
@@ -365,7 +316,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
dev_dbg(&ssp->pdev->dev,
"Using SSACDD %x to supply %uHz\n",
- val, freq_out);
+ val, freq);
break;
}
@@ -535,6 +486,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
case SND_SOC_DAIFMT_DSP_A:
sspsp |= SSPSP_FSRT;
+ /* fall through */
case SND_SOC_DAIFMT_DSP_B:
sscr0 |= SSCR0_MOD | SSCR0_PSP;
sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
@@ -570,6 +522,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
return 0;
}
+struct pxa_ssp_clock_mode {
+ int rate;
+ int pll;
+ u8 acds;
+ u8 scdb;
+};
+
+static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = {
+ { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X },
+ { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X },
+ { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X },
+ { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X },
+ { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X },
+ { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X },
+ { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X },
+ {}
+};
+
/*
* Set the SSP audio DMA parameters and sample size.
* Can be called multiple times by oss emulation.
@@ -581,11 +551,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
struct ssp_device *ssp = priv->ssp;
int chn = params_channels(params);
- u32 sscr0;
- u32 sspsp;
+ u32 sscr0, sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
struct snd_dmaengine_dai_dma_data *dma_data;
+ int rate = params_rate(params);
+ int bclk = rate * chn * (width / 8);
int ret;
dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
@@ -625,11 +596,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
}
pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+ if (sscr0 & SSCR0_ACS) {
+ ret = pxa_ssp_set_pll(priv, bclk);
+
+ /*
+ * If we were able to generate the bclk directly,
+ * all is fine. Otherwise, look up the closest rate
+ * from the table and also set the dividers.
+ */
+
+ if (ret < 0) {
+ const struct pxa_ssp_clock_mode *m;
+ int ssacd, acds;
+
+ for (m = pxa_ssp_clock_modes; m->rate; m++) {
+ if (m->rate == rate)
+ break;
+ }
+
+ if (!m->rate)
+ return -EINVAL;
+
+ acds = m->acds;
+
+ /* The values in the table are for 16 bits */
+ if (width == 32)
+ acds--;
+
+ ret = pxa_ssp_set_pll(priv, bclk);
+ if (ret < 0)
+ return ret;
+
+ ssacd = pxa_ssp_read_reg(ssp, SSACD);
+ ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X);
+ ssacd |= SSACD_ACDS(m->acds);
+ ssacd |= m->scdb;
+ pxa_ssp_write_reg(ssp, SSACD, ssacd);
+ }
+ } else if (sscr0 & SSCR0_ECS) {
+ /*
+ * For setups with external clocking, the PLL and its diviers
+ * are not active. Instead, the SCR bits in SSCR0 can be used
+ * to divide the clock.
+ */
+ pxa_ssp_set_scr(ssp, bclk / rate);
+ }
+
switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
sspsp = pxa_ssp_read_reg(ssp, SSPSP);
- if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
+ if (((priv->sysclk / bclk) == 64) && (width == 16)) {
/* This is a special case where the bitclk is 64fs
* and we're not dealing with 2*32 bits of audio
* samples.
@@ -773,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
ret = -ENODEV;
goto err_priv;
}
+
+ priv->extclk = devm_clk_get(dev, "extclk");
+ if (IS_ERR(priv->extclk)) {
+ ret = PTR_ERR(priv->extclk);
+ if (ret == -EPROBE_DEFER)
+ return ret;
+
+ priv->extclk = NULL;
+ }
} else {
priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
if (priv->ssp == NULL) {
@@ -814,8 +840,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
.trigger = pxa_ssp_trigger,
.hw_params = pxa_ssp_hw_params,
.set_sysclk = pxa_ssp_set_dai_sysclk,
- .set_clkdiv = pxa_ssp_set_dai_clkdiv,
- .set_pll = pxa_ssp_set_dai_pll,
.set_fmt = pxa_ssp_set_dai_fmt,
.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
.set_tristate = pxa_ssp_set_dai_tristate,
@@ -843,6 +867,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = {
static const struct snd_soc_component_driver pxa_ssp_component = {
.name = "pxa-ssp",
+ .ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_soc_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
};
#ifdef CONFIG_OF
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 803818aabee9..9f779657bc86 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
.reset = pxa2xx_ac97_cold_reset,
};
-static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = {
- .prio = PXAD_PRIO_LOWEST,
- .drcmr = 11,
-};
-
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .chan_name = "pcm_pcm_stereo_in",
.maxburst = 32,
- .filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
-};
-
-static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = {
- .prio = PXAD_PRIO_LOWEST,
- .drcmr = 12,
};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .chan_name = "pcm_pcm_stereo_out",
.maxburst = 32,
- .filter_data = &pxa2xx_ac97_pcm_stereo_out_req,
};
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = {
- .prio = PXAD_PRIO_LOWEST,
- .drcmr = 10,
-};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
.addr = __PREG(MODR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .chan_name = "pcm_aux_mono_out",
.maxburst = 16,
- .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req,
};
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = {
- .prio = PXAD_PRIO_LOWEST,
- .drcmr = 9,
-};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
.addr = __PREG(MODR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .chan_name = "pcm_aux_mono_in",
.maxburst = 16,
- .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req,
};
-static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = {
- .prio = PXAD_PRIO_LOWEST,
- .drcmr = 8,
-};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
.addr = __PREG(MCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .chan_name = "pcm_aux_mic_mono",
.maxburst = 16,
- .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req,
};
static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream,
@@ -236,7 +214,21 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
static const struct snd_soc_component_driver pxa_ac97_component = {
.name = "pxa-ac97",
+ .ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_soc_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa2xx_ac97_dt_ids[] = {
+ { .compatible = "marvell,pxa250-ac97", },
+ { .compatible = "marvell,pxa270-ac97", },
+ { .compatible = "marvell,pxa300-ac97", },
+ { }
};
+MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
+
+#endif
static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
@@ -296,6 +288,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
#ifdef CONFIG_PM_SLEEP
.pm = &pxa2xx_ac97_pm_ops,
#endif
+ .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids),
},
};
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 3fb60baf6eab..42820121e5b9 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s;
static struct clk *clk_i2s;
static int clk_ena = 0;
-static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
.addr = __PREG(SADR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .chan_name = "tx",
.maxburst = 32,
- .filter_data = &pxa2xx_i2s_pcm_stereo_out_req,
};
-static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
.addr = __PREG(SADR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .chan_name = "rx",
.maxburst = 32,
- .filter_data = &pxa2xx_i2s_pcm_stereo_in_req,
};
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -366,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = {
static const struct snd_soc_component_driver pxa_i2s_component = {
.name = "pxa-i2s",
+ .ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_soc_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
};
static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 8b6a70e94c01..72eaaef1b426 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -20,70 +20,6 @@
#include <sound/pxa2xx-lib.h>
#include <sound/dmaengine_pcm.h>
-#include "../../arm/pxa2xx-pcm.h"
-
-static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_dmaengine_dai_dma_data *dma;
-
- dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- /* return if this is a bufferless transfer e.g.
- * codec <--> BT codec or GSM modem -- lg FIXME */
- if (!dma)
- return 0;
-
- return __pxa2xx_pcm_hw_params(substream, params);
-}
-
-static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- __pxa2xx_pcm_hw_free(substream);
-
- return 0;
-}
-
-static const struct snd_pcm_ops pxa2xx_pcm_ops = {
- .open = __pxa2xx_pcm_open,
- .close = __pxa2xx_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = pxa2xx_pcm_hw_params,
- .hw_free = pxa2xx_pcm_hw_free,
- .prepare = __pxa2xx_pcm_prepare,
- .trigger = pxa2xx_pcm_trigger,
- .pointer = pxa2xx_pcm_pointer,
- .mmap = pxa2xx_pcm_mmap,
-};
-
-static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- struct snd_pcm *pcm = rtd->pcm;
- int ret;
-
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
- if (ret)
- return ret;
-
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
- out:
- return ret;
-}
-
static const struct snd_soc_component_driver pxa2xx_soc_platform = {
.ops = &pxa2xx_pcm_ops,
.pcm_new = pxa2xx_soc_pcm_new,
@@ -96,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
NULL, 0);
}
-#ifdef CONFIG_OF
-static const struct of_device_id snd_soc_pxa_audio_match[] = {
- { .compatible = "mrvl,pxa-pcm-audio" },
- { }
-};
-MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
-#endif
-
static struct platform_driver pxa_pcm_driver = {
.driver = {
.name = "pxa-pcm-audio",
- .of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
},
.probe = pxa2xx_soc_platform_probe,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ba468e560dd2..230eee450f45 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- unsigned int pll_out = 0;
unsigned int wm9713_div = 0;
int ret = 0;
int rate = params_rate(params);
- int width = snd_pcm_format_physical_width(params_format(params));
/* Only support ratios that we can generate neatly from the AC97
* based master clock - in particular, this excludes 44.1kHz.
@@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- /* Add 1 to the width for the leading clock cycle */
- pll_out = rate * (width + 1) * 8;
-
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
- if (ret < 0)
- return ret;
-
if (clk_pout)
ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
WM9713_PCMDIV(wm9713_div));
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 87838fa27997..2a4c912d1e48 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC
APQ8016 SOC-based systems.
Say Y if you want to use audio devices on MI2S.
+config SND_SOC_QCOM_COMMON
+ tristate
+
config SND_SOC_QDSP6_COMMON
tristate
@@ -86,7 +89,18 @@ config SND_SOC_MSM8996
tristate "SoC Machine driver for MSM8996 and APQ8096 boards"
depends on QCOM_APR
select SND_SOC_QDSP6
+ select SND_SOC_QCOM_COMMON
help
Support for Qualcomm Technologies LPASS audio block in
APQ8096 SoC-based systems.
Say Y if you want to use audio device on this SoCs
+
+config SND_SOC_SDM845
+ tristate "SoC Machine driver for SDM845 boards"
+ depends on QCOM_APR
+ select SND_SOC_QDSP6
+ select SND_SOC_QCOM_COMMON
+ help
+ To add support for audio on Qualcomm Technologies Inc.
+ SDM845 SoC-based systems.
+ Say Y if you want to use audio device on this SoCs.
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 206945bb9ba1..41b2c7a23a4d 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -14,10 +14,14 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
snd-soc-storm-objs := storm.o
snd-soc-apq8016-sbc-objs := apq8016_sbc.o
snd-soc-apq8096-objs := apq8096.o
+snd-soc-sdm845-objs := sdm845.o
+snd-soc-qcom-common-objs := common.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
+obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
+obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o
#DSP lib
obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 561cd429e6f2..1543e85629f8 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -1,14 +1,13 @@
// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2018, Linaro Limited
-#include <linux/soc/qcom/apr.h>
#include <linux/module.h>
-#include <linux/component.h>
#include <linux/platform_device.h>
#include <linux/of_device.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
+#include "common.h"
static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
@@ -24,211 +23,57 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
-static int apq8096_sbc_parse_of(struct snd_soc_card *card)
+static void apq8096_add_be_ops(struct snd_soc_card *card)
{
- struct device_node *np;
- struct device_node *codec = NULL;
- struct device_node *platform = NULL;
- struct device_node *cpu = NULL;
- struct device *dev = card->dev;
- struct snd_soc_dai_link *link;
- int ret, num_links;
-
- ret = snd_soc_of_parse_card_name(card, "qcom,model");
- if (ret) {
- dev_err(dev, "Error parsing card name: %d\n", ret);
- return ret;
- }
-
- /* DAPM routes */
- if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) {
- ret = snd_soc_of_parse_audio_routing(card,
- "qcom,audio-routing");
- if (ret)
- return ret;
- }
-
- /* Populate links */
- num_links = of_get_child_count(dev->of_node);
+ struct snd_soc_dai_link *link = card->dai_link;
+ int i, num_links = card->num_links;
- /* Allocate the DAI link array */
- card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
- if (!card->dai_link)
- return -ENOMEM;
-
- card->num_links = num_links;
- link = card->dai_link;
-
- for_each_child_of_node(dev->of_node, np) {
- cpu = of_get_child_by_name(np, "cpu");
- if (!cpu) {
- dev_err(dev, "Can't find cpu DT node\n");
- ret = -EINVAL;
- goto err;
- }
-
- link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
- if (!link->cpu_of_node) {
- dev_err(card->dev, "error getting cpu phandle\n");
- ret = -EINVAL;
- goto err;
- }
-
- ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
- if (ret) {
- dev_err(card->dev, "error getting cpu dai name\n");
- goto err;
- }
-
- platform = of_get_child_by_name(np, "platform");
- codec = of_get_child_by_name(np, "codec");
- if (codec && platform) {
- link->platform_of_node = of_parse_phandle(platform,
- "sound-dai",
- 0);
- if (!link->platform_of_node) {
- dev_err(card->dev, "platform dai not found\n");
- ret = -EINVAL;
- goto err;
- }
-
- ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
- if (ret < 0) {
- dev_err(card->dev, "codec dai not found\n");
- goto err;
- }
- link->no_pcm = 1;
- link->ignore_pmdown_time = 1;
+ for (i = 0; i < num_links; i++) {
+ if (link->no_pcm == 1)
link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
- } else {
- link->platform_of_node = link->cpu_of_node;
- link->codec_dai_name = "snd-soc-dummy-dai";
- link->codec_name = "snd-soc-dummy";
- link->dynamic = 1;
- }
-
- link->ignore_suspend = 1;
- ret = of_property_read_string(np, "link-name", &link->name);
- if (ret) {
- dev_err(card->dev, "error getting codec dai_link name\n");
- goto err;
- }
-
- link->dpcm_playback = 1;
- link->dpcm_capture = 1;
- link->stream_name = link->name;
link++;
}
-
- return 0;
-err:
- of_node_put(cpu);
- of_node_put(codec);
- of_node_put(platform);
- kfree(card->dai_link);
- return ret;
}
-static int apq8096_bind(struct device *dev)
+static int apq8096_platform_probe(struct platform_device *pdev)
{
struct snd_soc_card *card;
+ struct device *dev = &pdev->dev;
int ret;
card = kzalloc(sizeof(*card), GFP_KERNEL);
if (!card)
return -ENOMEM;
- component_bind_all(dev, card);
card->dev = dev;
- ret = apq8096_sbc_parse_of(card);
+ dev_set_drvdata(dev, card);
+ ret = qcom_snd_parse_of(card);
if (ret) {
dev_err(dev, "Error parsing OF data\n");
goto err;
}
+ apq8096_add_be_ops(card);
ret = snd_soc_register_card(card);
if (ret)
- goto err;
+ goto err_card_register;
return 0;
+err_card_register:
+ kfree(card->dai_link);
err:
- component_unbind_all(dev, card);
kfree(card);
return ret;
}
-static void apq8096_unbind(struct device *dev)
+static int apq8096_platform_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_card(card);
- component_unbind_all(dev, card);
kfree(card->dai_link);
kfree(card);
-}
-
-static const struct component_master_ops apq8096_ops = {
- .bind = apq8096_bind,
- .unbind = apq8096_unbind,
-};
-
-static int apq8016_compare_of(struct device *dev, void *data)
-{
- return dev->of_node == data;
-}
-
-static void apq8016_release_of(struct device *dev, void *data)
-{
- of_node_put(data);
-}
-
-static int add_audio_components(struct device *dev,
- struct component_match **matchptr)
-{
- struct device_node *np, *platform, *cpu, *node, *dai_node;
-
- node = dev->of_node;
-
- for_each_child_of_node(node, np) {
- cpu = of_get_child_by_name(np, "cpu");
- if (cpu) {
- dai_node = of_parse_phandle(cpu, "sound-dai", 0);
- of_node_get(dai_node);
- component_match_add_release(dev, matchptr,
- apq8016_release_of,
- apq8016_compare_of,
- dai_node);
- }
-
- platform = of_get_child_by_name(np, "platform");
- if (platform) {
- dai_node = of_parse_phandle(platform, "sound-dai", 0);
- component_match_add_release(dev, matchptr,
- apq8016_release_of,
- apq8016_compare_of,
- dai_node);
- }
- }
-
- return 0;
-}
-
-static int apq8096_platform_probe(struct platform_device *pdev)
-{
- struct component_match *match = NULL;
- int ret;
-
- ret = add_audio_components(&pdev->dev, &match);
- if (ret)
- return ret;
-
- return component_master_add_with_match(&pdev->dev, &apq8096_ops, match);
-}
-
-static int apq8096_platform_remove(struct platform_device *pdev)
-{
- component_master_del(&pdev->dev, &apq8096_ops);
return 0;
}
@@ -245,7 +90,6 @@ static struct platform_driver msm_snd_apq8096_driver = {
.remove = apq8096_platform_remove,
.driver = {
.name = "msm-snd-apq8096",
- .owner = THIS_MODULE,
.of_match_table = msm_snd_apq8096_dt_match,
},
};
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
new file mode 100644
index 000000000000..eb1b9da05dd4
--- /dev/null
+++ b/sound/soc/qcom/common.c
@@ -0,0 +1,112 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited.
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#include <linux/module.h>
+#include "common.h"
+
+int qcom_snd_parse_of(struct snd_soc_card *card)
+{
+ struct device_node *np;
+ struct device_node *codec = NULL;
+ struct device_node *platform = NULL;
+ struct device_node *cpu = NULL;
+ struct device *dev = card->dev;
+ struct snd_soc_dai_link *link;
+ int ret, num_links;
+
+ ret = snd_soc_of_parse_card_name(card, "model");
+ if (ret) {
+ dev_err(dev, "Error parsing card name: %d\n", ret);
+ return ret;
+ }
+
+ /* DAPM routes */
+ if (of_property_read_bool(dev->of_node, "audio-routing")) {
+ ret = snd_soc_of_parse_audio_routing(card,
+ "audio-routing");
+ if (ret)
+ return ret;
+ }
+
+ /* Populate links */
+ num_links = of_get_child_count(dev->of_node);
+
+ /* Allocate the DAI link array */
+ card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
+ if (!card->dai_link)
+ return -ENOMEM;
+
+ card->num_links = num_links;
+ link = card->dai_link;
+ for_each_child_of_node(dev->of_node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ if (!cpu) {
+ dev_err(dev, "Can't find cpu DT node\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
+ if (!link->cpu_of_node) {
+ dev_err(card->dev, "error getting cpu phandle\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+ if (ret) {
+ dev_err(card->dev, "error getting cpu dai name\n");
+ goto err;
+ }
+
+ platform = of_get_child_by_name(np, "platform");
+ codec = of_get_child_by_name(np, "codec");
+ if (codec && platform) {
+ link->platform_of_node = of_parse_phandle(platform,
+ "sound-dai",
+ 0);
+ if (!link->platform_of_node) {
+ dev_err(card->dev, "platform dai not found\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+ if (ret < 0) {
+ dev_err(card->dev, "codec dai not found\n");
+ goto err;
+ }
+ link->no_pcm = 1;
+ link->ignore_pmdown_time = 1;
+ } else {
+ link->platform_of_node = link->cpu_of_node;
+ link->codec_dai_name = "snd-soc-dummy-dai";
+ link->codec_name = "snd-soc-dummy";
+ link->dynamic = 1;
+ }
+
+ link->ignore_suspend = 1;
+ ret = of_property_read_string(np, "link-name", &link->name);
+ if (ret) {
+ dev_err(card->dev, "error getting codec dai_link name\n");
+ goto err;
+ }
+
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+ link->stream_name = link->name;
+ link++;
+ }
+
+ return 0;
+err:
+ of_node_put(cpu);
+ of_node_put(codec);
+ of_node_put(platform);
+ kfree(card->dai_link);
+ return ret;
+}
+EXPORT_SYMBOL(qcom_snd_parse_of);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h
new file mode 100644
index 000000000000..f05c05b12bd7
--- /dev/null
+++ b/sound/soc/qcom/common.h
@@ -0,0 +1,11 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#ifndef __QCOM_SND_COMMON_H__
+#define __QCOM_SND_COMMON_H__
+
+#include <sound/soc.h>
+
+int qcom_snd_parse_of(struct snd_soc_card *card);
+
+#endif
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 31fe78aa207f..d07271ea4c45 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler(
return IRQ_NONE;
}
dev_warn(soc_runtime->dev, "xrun warning\n");
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stop_xrun(substream);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 9983c665a941..932c3ebfd252 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -64,7 +64,6 @@ struct q6adm {
struct aprv2_ibasic_rsp_result_t result;
struct mutex lock;
wait_queue_head_t matrix_map_wait;
- struct platform_device *pdev_routing;
};
struct q6adm_cmd_device_open_v5 {
@@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close);
static int q6adm_probe(struct apr_device *adev)
{
struct device *dev = &adev->dev;
- struct device_node *dais_np;
struct q6adm *adm;
adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
@@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev)
INIT_LIST_HEAD(&adm->copps_list);
spin_lock_init(&adm->copps_list_lock);
- dais_np = of_get_child_by_name(dev->of_node, "routing");
- if (dais_np) {
- adm->pdev_routing = of_platform_device_create(dais_np,
- "q6routing", dev);
- of_node_put(dais_np);
- }
-
- return 0;
+ return of_platform_populate(dev->of_node, NULL, NULL, dev);
}
static int q6adm_remove(struct apr_device *adev)
{
- struct q6adm *adm = dev_get_drvdata(&adev->dev);
-
- if (adm->pdev_routing)
- of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+ of_platform_depopulate(&adev->dev);
return 0;
}
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 5002dd05bf27..60ff4a2d3577 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -4,7 +4,6 @@
#include <linux/err.h>
#include <linux/init.h>
-#include <linux/component.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/platform_device.h>
@@ -81,7 +80,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream,
struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
- slim->num_channels = params_channels(params);
slim->sample_rate = params_rate(params);
switch (params_format(params)) {
@@ -315,6 +313,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
int rc;
+ if (!dai_data->is_port_started[dai->id])
+ return;
+
rc = q6afe_port_stop(dai_data->port[dai->id]);
if (rc < 0)
dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
@@ -382,23 +383,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai,
struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
int i;
- if (!rx_slot) {
- pr_err("%s: rx slot not found\n", __func__);
- return -EINVAL;
- }
+ if (dai->id & 0x1) {
+ /* TX */
+ if (!tx_slot) {
+ pr_err("%s: tx slot not found\n", __func__);
+ return -EINVAL;
+ }
- for (i = 0; i < rx_num; i++) {
- pcfg->slim.ch_mapping[i] = rx_slot[i];
- pr_debug("%s: find number of channels[%d] ch[%d]\n",
- __func__, i, rx_slot[i]);
- }
+ for (i = 0; i < tx_num; i++)
+ pcfg->slim.ch_mapping[i] = tx_slot[i];
+
+ pcfg->slim.num_channels = tx_num;
+
+
+ } else {
+ if (!rx_slot) {
+ pr_err("%s: rx slot not found\n", __func__);
+ return -EINVAL;
+ }
- pcfg->slim.num_channels = rx_num;
+ for (i = 0; i < rx_num; i++)
+ pcfg->slim.ch_mapping[i] = rx_slot[i];
- pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
- (dai->id - SLIMBUS_0_RX) / 2, rx_num,
- pcfg->slim.ch_mapping[0],
- pcfg->slim.ch_mapping[1]);
+ pcfg->slim.num_channels = rx_num;
+
+ }
return 0;
}
@@ -443,6 +452,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
{"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
{"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
+ {"SLIMBUS_0_TX", NULL, "Slimbus Capture"},
+ {"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"},
+ {"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"},
+ {"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"},
+ {"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"},
+ {"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"},
+ {"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"},
+
{"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
{"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
@@ -637,6 +654,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.rate_max = 192000,
},
}, {
+ .name = "SLIMBUS_0_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_0_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ }, {
.playback = {
.stream_name = "Slimbus1 Playback",
.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
@@ -655,6 +690,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
}, {
+ .name = "SLIMBUS_1_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_1_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus1 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ }, {
.playback = {
.stream_name = "Slimbus2 Playback",
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
@@ -672,6 +725,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.id = SLIMBUS_2_RX,
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
+
+ }, {
+ .name = "SLIMBUS_2_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_2_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus2 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
}, {
.playback = {
.stream_name = "Slimbus3 Playback",
@@ -690,6 +762,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.id = SLIMBUS_3_RX,
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
+
+ }, {
+ .name = "SLIMBUS_3_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_3_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus3 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
}, {
.playback = {
.stream_name = "Slimbus4 Playback",
@@ -708,6 +799,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.id = SLIMBUS_4_RX,
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
+
+ }, {
+ .name = "SLIMBUS_4_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_4_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus4 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
}, {
.playback = {
.stream_name = "Slimbus5 Playback",
@@ -726,6 +836,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.id = SLIMBUS_5_RX,
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
+
+ }, {
+ .name = "SLIMBUS_5_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_5_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus5 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
}, {
.playback = {
.stream_name = "Slimbus6 Playback",
@@ -744,6 +873,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
.id = SLIMBUS_6_RX,
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
+
+ }, {
+ .name = "SLIMBUS_6_TX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_6_TX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .capture = {
+ .stream_name = "Slimbus6 Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
}, {
.playback = {
.stream_name = "Primary MI2S Playback",
@@ -972,6 +1120,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
@@ -1180,7 +1335,7 @@ static void of_q6afe_parse_dai_data(struct device *dev,
int id, i, num_lines;
ret = of_property_read_u32(node, "reg", &id);
- if (ret || id > AFE_PORT_MAX) {
+ if (ret || id < 0 || id >= AFE_PORT_MAX) {
dev_err(dev, "valid dai id not found:%d\n", ret);
continue;
}
@@ -1249,11 +1404,12 @@ static void of_q6afe_parse_dai_data(struct device *dev,
}
}
-static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6afe_dai_dev_probe(struct platform_device *pdev)
{
struct q6afe_dai_data *dai_data;
+ struct device *dev = &pdev->dev;
- dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL);
+ dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL);
if (!dai_data)
return -ENOMEM;
@@ -1261,41 +1417,22 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
of_q6afe_parse_dai_data(dev, dai_data);
- return snd_soc_register_component(dev, &q6afe_dai_component,
+ return devm_snd_soc_register_component(dev, &q6afe_dai_component,
q6afe_dais, ARRAY_SIZE(q6afe_dais));
}
-static void q6afe_dai_unbind(struct device *dev, struct device *master,
- void *data)
-{
- struct q6afe_dai_data *dai_data = dev_get_drvdata(dev);
-
- snd_soc_unregister_component(dev);
- kfree(dai_data);
-}
-
-static const struct component_ops q6afe_dai_comp_ops = {
- .bind = q6afe_dai_bind,
- .unbind = q6afe_dai_unbind,
+static const struct of_device_id q6afe_dai_device_id[] = {
+ { .compatible = "qcom,q6afe-dais" },
+ {},
};
-
-static int q6afe_dai_dev_probe(struct platform_device *pdev)
-{
- return component_add(&pdev->dev, &q6afe_dai_comp_ops);
-}
-
-static int q6afe_dai_dev_remove(struct platform_device *pdev)
-{
- component_del(&pdev->dev, &q6afe_dai_comp_ops);
- return 0;
-}
+MODULE_DEVICE_TABLE(of, q6afe_dai_device_id);
static struct platform_driver q6afe_dai_platform_driver = {
.driver = {
.name = "q6afe-dai",
+ .of_match_table = of_match_ptr(q6afe_dai_device_id),
},
.probe = q6afe_dai_dev_probe,
- .remove = q6afe_dai_dev_remove,
};
module_platform_driver(q6afe_dai_platform_driver);
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 01f43218984b..000775b4bba8 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -316,7 +316,6 @@ struct q6afe {
struct mutex lock;
struct list_head port_list;
spinlock_t port_list_lock;
- struct platform_device *pdev_dais;
};
struct afe_port_cmd_device_start {
@@ -515,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
SLIMBUS_5_RX, 1, 1},
[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
SLIMBUS_6_RX, 1, 1},
+ [SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX,
+ SLIMBUS_0_TX, 0, 1},
+ [SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX,
+ SLIMBUS_1_TX, 0, 1},
+ [SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX,
+ SLIMBUS_2_TX, 0, 1},
+ [SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX,
+ SLIMBUS_3_TX, 0, 1},
+ [SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX,
+ SLIMBUS_4_TX, 0, 1},
+ [SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX,
+ SLIMBUS_5_TX, 0, 1},
+ [SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX,
+ SLIMBUS_6_TX, 0, 1},
[PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
PRIMARY_MI2S_RX, 1, 1},
[PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
@@ -777,7 +790,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data)
*/
int q6afe_get_port_id(int index)
{
- if (index < 0 || index > AFE_PORT_MAX)
+ if (index < 0 || index >= AFE_PORT_MAX)
return -EINVAL;
return port_maps[index].port_id;
@@ -1014,7 +1027,7 @@ int q6afe_port_stop(struct q6afe_port *port)
port_id = port->id;
index = port->token;
- if (index < 0 || index > AFE_PORT_MAX) {
+ if (index < 0 || index >= AFE_PORT_MAX) {
dev_err(afe->dev, "AFE port index[%d] invalid!\n", index);
return -EINVAL;
}
@@ -1355,7 +1368,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
unsigned long flags;
int cfg_type;
- if (id < 0 || id > AFE_PORT_MAX) {
+ if (id < 0 || id >= AFE_PORT_MAX) {
dev_err(dev, "AFE port token[%d] invalid!\n", id);
return ERR_PTR(-EINVAL);
}
@@ -1373,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
case AFE_PORT_ID_MULTICHAN_HDMI_RX:
cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
break;
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX:
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
@@ -1438,7 +1458,6 @@ static int q6afe_probe(struct apr_device *adev)
{
struct q6afe *afe;
struct device *dev = &adev->dev;
- struct device_node *dais_np;
afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
if (!afe)
@@ -1453,22 +1472,12 @@ static int q6afe_probe(struct apr_device *adev)
dev_set_drvdata(dev, afe);
- dais_np = of_get_child_by_name(dev->of_node, "dais");
- if (dais_np) {
- afe->pdev_dais = of_platform_device_create(dais_np,
- "q6afe-dai", dev);
- of_node_put(dais_np);
- }
-
- return 0;
+ return of_platform_populate(dev->of_node, NULL, NULL, dev);
}
static int q6afe_remove(struct apr_device *adev)
{
- struct q6afe *afe = dev_get_drvdata(&adev->dev);
-
- if (afe->pdev_dais)
- of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+ of_platform_depopulate(&adev->dev);
return 0;
}
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 349c6a883c63..9db9a2944ef2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -7,7 +7,6 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
-#include <linux/component.h>
#include <sound/soc.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
@@ -390,7 +389,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->audio_client) {
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ if (prtd->state)
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6asm_audio_client_free(prtd->audio_client);
@@ -561,14 +562,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
Q6ASM_FEDAI_DRIVER(8),
};
-static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6asm_dai_probe(struct platform_device *pdev)
{
+ struct device *dev = &pdev->dev;
struct device_node *node = dev->of_node;
struct of_phandle_args args;
struct q6asm_dai_data *pdata;
int rc;
- pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
+ pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
if (!pdata)
return -ENOMEM;
@@ -580,43 +582,23 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
dev_set_drvdata(dev, pdata);
- return snd_soc_register_component(dev, &q6asm_fe_dai_component,
+ return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
q6asm_fe_dais,
ARRAY_SIZE(q6asm_fe_dais));
}
-static void q6asm_dai_unbind(struct device *dev, struct device *master,
- void *data)
-{
- struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
-
- snd_soc_unregister_component(dev);
-
- kfree(pdata);
-
-}
-static const struct component_ops q6asm_dai_comp_ops = {
- .bind = q6asm_dai_bind,
- .unbind = q6asm_dai_unbind,
+static const struct of_device_id q6asm_dai_device_id[] = {
+ { .compatible = "qcom,q6asm-dais" },
+ {},
};
-
-static int q6asm_dai_probe(struct platform_device *pdev)
-{
- return component_add(&pdev->dev, &q6asm_dai_comp_ops);
-}
-
-static int q6asm_dai_dev_remove(struct platform_device *pdev)
-{
- component_del(&pdev->dev, &q6asm_dai_comp_ops);
- return 0;
-}
+MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
static struct platform_driver q6asm_dai_platform_driver = {
.driver = {
.name = "q6asm-dai",
+ .of_match_table = of_match_ptr(q6asm_dai_device_id),
},
.probe = q6asm_dai_probe,
- .remove = q6asm_dai_dev_remove,
};
module_platform_driver(q6asm_dai_platform_driver);
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 530852385cad..2b2c7233bb5f 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -174,10 +174,8 @@ struct q6asm {
struct device *dev;
struct q6core_svc_api_info ainfo;
wait_queue_head_t mem_wait;
- struct platform_device *pcmdev;
spinlock_t slock;
struct audio_client *session[MAX_SESSIONS + 1];
- struct platform_device *pdev_dais;
};
struct audio_client {
@@ -1344,7 +1342,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
static int q6asm_probe(struct apr_device *adev)
{
struct device *dev = &adev->dev;
- struct device_node *dais_np;
struct q6asm *q6asm;
q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
@@ -1359,22 +1356,12 @@ static int q6asm_probe(struct apr_device *adev)
spin_lock_init(&q6asm->slock);
dev_set_drvdata(dev, q6asm);
- dais_np = of_get_child_by_name(dev->of_node, "dais");
- if (dais_np) {
- q6asm->pdev_dais = of_platform_device_create(dais_np,
- "q6asm-dai", dev);
- of_node_put(dais_np);
- }
-
- return 0;
+ return of_platform_populate(dev->of_node, NULL, NULL, dev);
}
static int q6asm_remove(struct apr_device *adev)
{
- struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
-
- if (q6asm->pdev_dais)
- of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+ of_platform_depopulate(&adev->dev);
return 0;
}
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 593f66b8622f..dc94c5c53788 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -8,7 +8,6 @@
#include <linux/platform_device.h>
#include <linux/of_platform.h>
#include <linux/bitops.h>
-#include <linux/component.h>
#include <linux/mutex.h>
#include <linux/of_device.h>
#include <linux/slab.h>
@@ -68,6 +67,13 @@
{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \
{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \
{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \
+ { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \
+ { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \
+ { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \
+ { mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" }, \
+ { mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" }, \
+ { mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" }, \
+ { mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" }, \
{ mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"}, \
{ mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"}, \
{ mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"}, \
@@ -122,6 +128,27 @@
SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \
id, 1, 0, msm_routing_get_audio_mixer, \
msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
+ SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX, \
+ id, 1, 0, msm_routing_get_audio_mixer, \
+ msm_routing_put_audio_mixer), \
SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0, \
id, 1, 0, msm_routing_get_audio_mixer, \
msm_routing_put_audio_mixer), \
@@ -310,7 +337,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode,
session->channels, topology, perf_mode,
session->bits_per_sample, 0, 0);
- if (!copp) {
+ if (IS_ERR_OR_NULL(copp)) {
mutex_unlock(&routing_data->lock);
return -EINVAL;
}
@@ -899,7 +926,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream,
else
path_type = ADM_PATH_LIVE_REC;
- if (be_id > AFE_MAX_PORTS)
+ if (be_id >= AFE_MAX_PORTS)
return -EINVAL;
session = &data->port_data[be_id];
@@ -949,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = {
.num_dapm_routes = ARRAY_SIZE(intercon),
};
-static int q6routing_dai_bind(struct device *dev, struct device *master,
- void *data)
+static int q6pcm_routing_probe(struct platform_device *pdev)
{
+ struct device *dev = &pdev->dev;
+
routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
if (!routing_data)
return -ENOMEM;
@@ -961,41 +989,28 @@ static int q6routing_dai_bind(struct device *dev, struct device *master,
mutex_init(&routing_data->lock);
dev_set_drvdata(dev, routing_data);
- return snd_soc_register_component(dev, &msm_soc_routing_component,
+ return devm_snd_soc_register_component(dev, &msm_soc_routing_component,
NULL, 0);
}
-static void q6routing_dai_unbind(struct device *dev, struct device *master,
- void *d)
+static int q6pcm_routing_remove(struct platform_device *pdev)
{
- struct msm_routing_data *data = dev_get_drvdata(dev);
-
- snd_soc_unregister_component(dev);
-
- kfree(data);
-
+ kfree(routing_data);
routing_data = NULL;
-}
-
-static const struct component_ops q6routing_dai_comp_ops = {
- .bind = q6routing_dai_bind,
- .unbind = q6routing_dai_unbind,
-};
-static int q6pcm_routing_probe(struct platform_device *pdev)
-{
- return component_add(&pdev->dev, &q6routing_dai_comp_ops);
-}
-
-static int q6pcm_routing_remove(struct platform_device *pdev)
-{
- component_del(&pdev->dev, &q6routing_dai_comp_ops);
return 0;
}
+static const struct of_device_id q6pcm_routing_device_id[] = {
+ { .compatible = "qcom,q6adm-routing" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id);
+
static struct platform_driver q6pcm_routing_platform_driver = {
.driver = {
.name = "q6routing",
+ .of_match_table = of_match_ptr(q6pcm_routing_device_id),
},
.probe = q6pcm_routing_probe,
.remove = q6pcm_routing_remove,
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
new file mode 100644
index 000000000000..2a781d87ee65
--- /dev/null
+++ b/sound/soc/qcom/sdm845.c
@@ -0,0 +1,285 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2018, The Linux Foundation. All rights reserved.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "common.h"
+#include "qdsp6/q6afe.h"
+
+#define DEFAULT_SAMPLE_RATE_48K 48000
+#define DEFAULT_MCLK_RATE 24576000
+#define DEFAULT_BCLK_RATE 12288000
+
+struct sdm845_snd_data {
+ struct snd_soc_card *card;
+ uint32_t pri_mi2s_clk_count;
+ uint32_t quat_tdm_clk_count;
+};
+
+static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
+
+static int sdm845_tdm_snd_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 *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ int channels, slot_width;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ slot_width = 32;
+ break;
+ default:
+ dev_err(rtd->dev, "%s: invalid param format 0x%x\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ channels = params_channels(params);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3,
+ 8, slot_width);
+ if (ret < 0) {
+ dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL,
+ channels, tdm_slot_offset);
+ if (ret < 0) {
+ dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+ } else {
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0,
+ 8, slot_width);
+ if (ret < 0) {
+ dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, channels,
+ tdm_slot_offset, 0, NULL);
+ if (ret < 0) {
+ dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+ }
+end:
+ return ret;
+}
+
+static int sdm845_snd_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 *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+
+ switch (cpu_dai->id) {
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ ret = sdm845_tdm_snd_hw_params(substream, params);
+ break;
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+ return ret;
+}
+
+static int sdm845_snd_startup(struct snd_pcm_substream *substream)
+{
+ unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ case PRIMARY_MI2S_TX:
+ if (++(data->pri_mi2s_clk_count) == 1) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_MCLK_1,
+ DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+ DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ snd_soc_dai_set_fmt(cpu_dai, fmt);
+ break;
+
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ if (++(data->quat_tdm_clk_count) == 1) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+ DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ break;
+
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+ return 0;
+}
+
+static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ case PRIMARY_MI2S_TX:
+ if (--(data->pri_mi2s_clk_count) == 0) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_MCLK_1,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ };
+ break;
+
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ if (--(data->quat_tdm_clk_count) == 0) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ break;
+
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+}
+
+static struct snd_soc_ops sdm845_be_ops = {
+ .hw_params = sdm845_snd_hw_params,
+ .startup = sdm845_snd_startup,
+ .shutdown = sdm845_snd_shutdown,
+};
+
+static int sdm845_be_hw_params_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);
+
+ rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K;
+ channels->min = channels->max = 2;
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+ return 0;
+}
+
+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;
+
+ for (i = 0; i < num_links; i++) {
+ if (link->no_pcm == 1) {
+ link->ops = &sdm845_be_ops;
+ link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
+ }
+ link++;
+ }
+}
+
+static int sdm845_snd_platform_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card;
+ struct sdm845_snd_data *data;
+ struct device *dev = &pdev->dev;
+ int ret;
+
+ card = kzalloc(sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ /* Allocate the private data */
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto data_alloc_fail;
+ }
+
+ card->dev = dev;
+ dev_set_drvdata(dev, card);
+ ret = qcom_snd_parse_of(card);
+ if (ret) {
+ dev_err(dev, "Error parsing OF data\n");
+ goto parse_dt_fail;
+ }
+
+ data->card = card;
+ snd_soc_card_set_drvdata(card, data);
+
+ sdm845_add_be_ops(card);
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(dev, "Sound card registration failed\n");
+ goto register_card_fail;
+ }
+ return ret;
+
+register_card_fail:
+ kfree(card->dai_link);
+parse_dt_fail:
+ kfree(data);
+data_alloc_fail:
+ kfree(card);
+ return ret;
+}
+
+static int sdm845_snd_platform_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ snd_soc_unregister_card(card);
+ kfree(card->dai_link);
+ kfree(data);
+ kfree(card);
+ return 0;
+}
+
+static const struct of_device_id sdm845_snd_device_id[] = {
+ { .compatible = "qcom,sdm845-sndcard" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, sdm845_snd_device_id);
+
+static struct platform_driver sdm845_snd_driver = {
+ .probe = sdm845_snd_platform_probe,
+ .remove = sdm845_snd_platform_remove,
+ .driver = {
+ .name = "msm-snd-sdm845",
+ .of_match_table = sdm845_snd_device_id,
+ },
+};
+module_platform_driver(sdm845_snd_driver);
+
+MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 05b078e7b87f..65e814d46006 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,10 +1,11 @@
# SPDX-License-Identifier: GPL-2.0
# ROCKCHIP Platform Support
snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-pcm-objs := rockchip_pcm.o
snd-soc-rockchip-pdm-objs := rockchip_pdm.o
snd-soc-rockchip-spdif-objs := rockchip_spdif.o
-obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 950823d69e9c..60d43d53a8f5 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -22,6 +22,7 @@
#include <sound/dmaengine_pcm.h>
#include "rockchip_i2s.h"
+#include "rockchip_pcm.h"
#define DRV_NAME "rockchip-i2s"
@@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
goto err_suspend;
}
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ ret = rockchip_pcm_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM\n");
return ret;
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
new file mode 100644
index 000000000000..f77538319221
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * 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 <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_pcm.h"
+
+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,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 52,
+ .buffer_bytes_max = 64 * 1024,
+ .fifo_size = 32,
+};
+
+static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
+ .pcm_hardware = &snd_rockchip_hardware,
+ .prealloc_buffer_size = 32 * 1024,
+};
+
+int rockchip_pcm_platform_register(struct device *dev)
+{
+ return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h
new file mode 100644
index 000000000000..d6c36115c60a
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.h
@@ -0,0 +1,14 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * 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 _ROCKCHIP_PCM_H
+#define _ROCKCHIP_PCM_H
+
+int rockchip_pcm_platform_register(struct device *dev);
+
+#endif
diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c
index 4db4fd56db35..881c32498808 100644
--- a/sound/soc/rockchip/rockchip_rt5645.c
+++ b/sound/soc/rockchip/rockchip_rt5645.c
@@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
if (!rk_dailink.cpu_of_node) {
dev_err(&pdev->dev,
"Property 'rockchip,i2s-controller' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_codec_of_node;
}
rk_dailink.platform_of_node = rk_dailink.cpu_of_node;
@@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
if (ret) {
dev_err(&pdev->dev,
"Soc parse card name failed %d\n", ret);
- return ret;
+ goto put_cpu_of_node;
}
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev,
"Soc register card failed %d\n", ret);
- return ret;
+ goto put_cpu_of_node;
}
return ret;
+
+put_cpu_of_node:
+ of_node_put(rk_dailink.cpu_of_node);
+ rk_dailink.cpu_of_node = NULL;
+put_codec_of_node:
+ of_node_put(rk_dailink.codec_of_node);
+ rk_dailink.codec_of_node = NULL;
+
+ return ret;
+}
+
+static int snd_rk_mc_remove(struct platform_device *pdev)
+{
+ of_node_put(rk_dailink.cpu_of_node);
+ rk_dailink.cpu_of_node = NULL;
+ of_node_put(rk_dailink.codec_of_node);
+ rk_dailink.codec_of_node = NULL;
+
+ return 0;
}
static const struct of_device_id rockchip_rt5645_of_match[] = {
@@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match);
static struct platform_driver snd_rk_mc_driver = {
.probe = snd_rk_mc_probe,
+ .remove = snd_rk_mc_remove,
.driver = {
.name = DRV_NAME,
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index f914ed45db7d..d6c62aa13041 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_channels(params)) {
case 6:
val |= MOD_DC2_EN;
+ /* fall through */
case 4:
val |= MOD_DC1_EN;
break;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 0ae0800bf3a8..dc20f0f7080a 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0
menu "SoC Audio support for Renesas SoCs"
depends on SUPERH || ARCH_RENESAS || COMPILE_TEST
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 2dc3b762fdd9..922fb6aa3ed1 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -1,16 +1,14 @@
-/*
- * SH7760 ("camelot") DMABRG audio DMA unit support
- *
- * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- * licensed under the terms outlined in the file COPYING at the root
- * of the linux kernel sources.
- *
- * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
- * trigger an interrupt when one half of the programmed transfer size
- * has been xmitted.
- *
- * FIXME: little-endian only for now
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// SH7760 ("camelot") DMABRG audio DMA unit support
+//
+// Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
+// trigger an interrupt when one half of the programmed transfer size
+// has been xmitted.
+//
+// FIXME: little-endian only for now
#include <linux/module.h>
#include <linux/gfp.h>
@@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = {
module_platform_driver(sh7760_pcm_driver);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3bae06dd121f..aa7e902f0c02 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1,16 +1,12 @@
-/*
- * Fifo-attached Serial Interface (FSI) support for SH7724
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on ssi.c
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Fifo-attached Serial Interface (FSI) support for SH7724
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on ssi.c
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
#include <linux/delay.h>
#include <linux/dma-mapping.h>
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 624aaf569fef..c2b496398e6b 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -1,13 +1,11 @@
-/*
- * Hitachi Audio Controller (AC97) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- * licensed under the terms outlined in the file COPYING at the root
- * of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable HAC output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Hitachi Audio Controller (AC97) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable HAC output pins!
/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
* the FIRST can be used since ASoC does not pass any information to the
@@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = {
module_platform_driver(hac_pcm_driver);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index ecb057ff9fbb..8739c9f60672 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -1,12 +1,8 @@
-/*
- * ALSA SoC driver for Migo-R
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ALSA SoC driver for Migo-R
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
#include <linux/clkdev.h>
#include <linux/device.h>
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 9c3d5aed99d1..5d1ff8ef26f9 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,3 @@
+# SPDX-License-Identifier: GPL-2.0
snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o
obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 4672688cac32..3a3064dda57f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -1,12 +1,9 @@
-/*
- * Helper routines for R-Car sound ADG.
- *
- * Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License. See the file "COPYING" in the main directory of this archive
- * for more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Helper routines for R-Car sound ADG.
+//
+// Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include <linux/clk-provider.h>
#include "rsnd.h"
diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c
index 5900fb535a2b..cc191cd5fb82 100644
--- a/sound/soc/sh/rcar/cmd.c
+++ b/sound/soc/sh/rcar/cmd.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car CMD support
- *
- * Copyright (C) 2015 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car CMD support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include "rsnd.h"
struct rsnd_cmd {
@@ -89,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod,
cmd_case[rsnd_mod_id(src)] << 16;
}
- dev_dbg(dev, "ctu/mix path = 0x%08x", data);
+ dev_dbg(dev, "ctu/mix path = 0x%08x\n", data);
rsnd_mod_write(mod, CMD_ROUTE_SLCT, data);
rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f237002180c0..f8425d8b44d2 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SRU/SCU/SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRU/SCU/SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
/*
* Renesas R-Car sound device structure
@@ -552,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
return priv->rdai + id;
}
+static struct snd_soc_dai_driver
+*rsnd_daidrv_get(struct rsnd_priv *priv, int id)
+{
+ if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
+ return NULL;
+
+ return priv->daidrv + id;
+}
+
#define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai)
static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai)
{
@@ -1037,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
int io_i;
rdai = rsnd_rdai_get(priv, dai_i);
- drv = priv->daidrv + dai_i;
+ drv = rsnd_daidrv_get(priv, dai_i);
io_playback = &rdai->playback;
io_capture = &rdai->capture;
@@ -1085,6 +1090,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
of_node_put(capture);
}
+ if (rsnd_ssi_is_pin_sharing(io_capture) ||
+ rsnd_ssi_is_pin_sharing(io_playback)) {
+ /* should have symmetric_rates if pin sharing */
+ drv->symmetric_rates = 1;
+ }
+
dev_dbg(dev, "%s (%s/%s)\n", rdai->name,
rsnd_io_to_mod_ssi(io_playback) ? "play" : " -- ",
rsnd_io_to_mod_ssi(io_capture) ? "capture" : " -- ");
@@ -1606,7 +1617,7 @@ static struct platform_driver rsnd_driver = {
};
module_platform_driver(rsnd_driver);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("Renesas R-Car audio driver");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
MODULE_ALIAS("platform:rcar-pcm-audio");
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 83be7d3ae0a8..6a55aa753003 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -1,12 +1,9 @@
-/*
- * ctu.c
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ctu.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include "rsnd.h"
#define CTU_NAME_SIZE 16
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index ef82b94d038b..fe63ef8600d0 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car Audio DMAC support
- *
- * Copyright (C) 2015 Renesas Electronics Corp.
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Audio DMAC support
+//
+// Copyright (C) 2015 Renesas Electronics Corp.
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include <linux/delay.h>
#include <linux/of_dma.h>
#include "rsnd.h"
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index ca1780e0b830..2b16e0ce6bc5 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car DVC support
- *
- * Copyright (C) 2014 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car DVC support
+//
+// Copyright (C) 2014 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
/*
* Playback Volume
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 25642e92dae0..0230301fe078 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car Gen1 SRU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Gen1 SRU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
/*
* #define DEBUG
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 1881b2de9126..8e3b57eaa708 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -1,12 +1,8 @@
-/*
- * mix.c
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// mix.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
/*
* CTUn MIXn
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 6d7280d2d9be..96d93330b1e1 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#ifndef RSND_H
#define RSND_H
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6c72d1a81cf5..beccfbac7581 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car SRC support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRC support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
/*
* you can enable below define if you don't need
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 6e1166ec24a0..8304e4ec9242 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
/*
* you can enable below define if you don't need
@@ -37,6 +33,7 @@
#define CHNL_4 (1 << 22) /* Channels */
#define CHNL_6 (2 << 22) /* Channels */
#define CHNL_8 (3 << 22) /* Channels */
+#define DWL_MASK (7 << 19) /* Data Word Length mask */
#define DWL_8 (0 << 19) /* Data Word Length */
#define DWL_16 (1 << 19) /* Data Word Length */
#define DWL_18 (2 << 19) /* Data Word Length */
@@ -353,21 +350,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- u32 cr_own;
- u32 cr_mode;
- u32 wsr;
+ u32 cr_own = ssi->cr_own;
+ u32 cr_mode = ssi->cr_mode;
+ u32 wsr = ssi->wsr;
int is_tdm;
- if (rsnd_ssi_is_parent(mod, io))
- return;
-
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 | SWL_32;
if (rdai->bit_clk_inv)
cr_own |= SCKP;
@@ -377,9 +371,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
cr_own |= SDTA;
if (rdai->sys_delay)
cr_own |= DEL;
+
+ /*
+ * We shouldn't exchange SWSP after running.
+ * This means, parent needs to care it.
+ */
+ if (rsnd_ssi_is_parent(mod, io))
+ goto init_end;
+
if (rsnd_io_is_play(io))
cr_own |= TRMD;
+ cr_own &= ~DWL_MASK;
switch (snd_pcm_format_width(runtime->format)) {
case 16:
cr_own |= DWL_16;
@@ -406,7 +409,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
wsr |= WS_MODE;
cr_own |= CHNL_8;
}
-
+init_end:
ssi->cr_own = cr_own;
ssi->cr_mode = cr_mode;
ssi->wsr = wsr;
@@ -470,15 +473,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
return -EIO;
}
- if (!rsnd_ssi_is_parent(mod, io))
- ssi->cr_own = 0;
-
rsnd_ssi_master_clk_stop(mod, io);
rsnd_mod_power_off(mod);
ssi->usrcnt--;
+ if (!ssi->usrcnt) {
+ ssi->cr_own = 0;
+ ssi->cr_mode = 0;
+ ssi->wsr = 0;
+ }
+
return 0;
}
@@ -1055,9 +1061,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
{
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ if (!mod)
+ return 0;
- return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE));
+ return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE));
}
static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io,
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 47bdba9fc582..016fbf5ac242 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -1,12 +1,9 @@
-/*
- * Renesas R-Car SSIU support
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU support
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
#include "rsnd.h"
#define SSIU_NAME "ssiu"
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 4a3568a9bf59..4bb4c13cf860 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -1,10 +1,8 @@
-/*
- * Generic AC97 sound support for SH7760
- *
- * (c) 2007 Manuel Lauss
- *
- * Licensed under the GPLv2.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Generic AC97 sound support for SH7760
+//
+// (c) 2007 Manuel Lauss
#include <linux/module.h>
#include <linux/moduleparam.h>
@@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void)
module_init(sh7760_ac97_init);
module_exit(sh7760_ac97_exit);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index 6088d627c0e4..63a508fdfe78 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -1,23 +1,9 @@
-/*
- * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
#ifndef SIU_H
#define SIU_H
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index ee2211635e92..f2a386fcd92e 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -1,23 +1,9 @@
-/*
- * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
#include <linux/delay.h>
#include <linux/firmware.h>
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 172909570ed5..e263757e4a69 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -1,23 +1,10 @@
-/*
- * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
+
#include <linux/delay.h>
#include <linux/dma-mapping.h>
#include <linux/dmaengine.h>
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 89ed1b107ac5..8125fa3840b6 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -1,14 +1,11 @@
-/*
- * Serial Sound Interface (I2S) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- * licensed under the terms outlined in the file COPYING at the root
- * of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable SSI output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Serial Sound Interface (I2S) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable SSI output pins!
/*
* LIMITATIONS:
@@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = {
module_platform_driver(sh4_ssi_driver);
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
index 77e7dcf969d0..d70fcd4a1adf 100644
--- a/sound/soc/sirf/sirf-usp.c
+++ b/sound/soc/sirf/sirf-usp.c
@@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, usp);
mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- base = devm_ioremap(&pdev->dev, mem_res->start,
- resource_size(mem_res));
- if (base == NULL)
- return -ENOMEM;
+ base = devm_ioremap_resource(&pdev->dev, mem_res);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
usp->regmap = devm_regmap_init_mmio(&pdev->dev, base,
&sirf_usp_regmap_config);
if (IS_ERR(usp->regmap))
diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c
index 3f424f214bca..c086786e4471 100644
--- a/sound/soc/soc-ac97.c
+++ b/sound/soc/soc-ac97.c
@@ -1,20 +1,15 @@
-/*
- * soc-ac97.c -- ALSA SoC Audio Layer AC97 support
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- * with code, comments and ideas from :-
- * Richard Purdie <richard@openedhand.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ac97.c -- ALSA SoC Audio Layer AC97 support
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+// with code, comments and ideas from :-
+// Richard Purdie <richard@openedhand.com>
#include <linux/ctype.h>
#include <linux/delay.h>
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index 3d7e1ff79139..b8e72b52db30 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -1,18 +1,8 @@
-/*
- * soc-apci.c - support for ACPI enumeration.
- *
- * Copyright (c) 2013-15, Intel Corporation.
- *
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms and conditions of the GNU General Public License,
- * version 2, as published by the Free Software Foundation.
- *
- * This program is distributed in the hope it will be useful, but WITHOUT
- * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
- * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
- * more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-apci.c - support for ACPI enumeration.
+//
+// Copyright (c) 2013-15, Intel Corporation.
#include <sound/soc-acpi.h>
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index e095115fa9f9..409d082e80d1 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -1,18 +1,12 @@
-/*
- * soc-compress.c -- ALSA SoC Compress
- *
- * Copyright (C) 2012 Intel Corp.
- *
- * Authors: Namarta Kohli <namartax.kohli@intel.com>
- * Ramesh Babu K V <ramesh.babu@linux.intel.com>
- * Vinod Koul <vinod.koul@linux.intel.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-compress.c -- ALSA SoC Compress
+//
+// Copyright (C) 2012 Intel Corp.
+//
+// Authors: Namarta Kohli <namartax.kohli@intel.com>
+// Ramesh Babu K V <ramesh.babu@linux.intel.com>
+// Vinod Koul <vinod.koul@linux.intel.com>
#include <linux/kernel.h>
#include <linux/init.h>
@@ -146,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
stream = SNDRV_PCM_STREAM_CAPTURE;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ fe->dpcm[stream].runtime = fe_substream->runtime;
+
+ ret = dpcm_path_get(fe, stream, &list);
+ if (ret < 0)
+ goto be_err;
+ else if (ret == 0)
+ dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
+ fe->dai_link->name, stream ? "capture" : "playback");
+ /* calculate valid and active FE <-> BE dpcms */
+ dpcm_process_paths(fe, stream, &list, 1);
+ fe->dpcm[stream].runtime = fe_substream->runtime;
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ 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)
+ dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+ dpcm_be_disconnect(fe, stream);
+ fe->dpcm[stream].runtime = NULL;
+ goto out;
+ }
if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) {
ret = cpu_dai->driver->cops->startup(cstream, cpu_dai);
@@ -159,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
ret = soc_compr_components_open(cstream, &component);
if (ret < 0)
- goto machine_err;
+ goto open_err;
if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) {
ret = fe->dai_link->compr_ops->startup(cstream);
@@ -170,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
}
}
- fe->dpcm[stream].runtime = fe_substream->runtime;
-
- ret = dpcm_path_get(fe, stream, &list);
- if (ret < 0)
- goto fe_err;
- else if (ret == 0)
- dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
- fe->dai_link->name, stream ? "capture" : "playback");
-
- /* calculate valid and active FE <-> BE dpcms */
- dpcm_process_paths(fe, stream, &list, 1);
-
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
- 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)
- dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
- dpcm_be_disconnect(fe, stream);
- fe->dpcm[stream].runtime = NULL;
- goto path_err;
- }
-
dpcm_clear_pending_state(fe, stream);
dpcm_path_put(&list);
@@ -207,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
return 0;
-path_err:
- dpcm_path_put(&list);
-fe_err:
- if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
- fe->dai_link->compr_ops->shutdown(cstream);
machine_err:
soc_compr_components_free(cstream, component);
-
+open_err:
if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown)
cpu_dai->driver->cops->shutdown(cstream, cpu_dai);
out:
+ dpcm_path_put(&list);
+be_err:
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
mutex_unlock(&fe->card->mutex);
return ret;
@@ -557,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ /*
+ * Create an empty hw_params for the BE as the machine driver must
+ * fix this up to match DSP decoder and ASRC configuration.
+ * I.e. machine driver fixup for compressed BE is mandatory.
+ */
+ memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
+ sizeof(struct snd_pcm_hw_params));
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ ret = dpcm_be_dai_hw_params(fe, stream);
+ if (ret < 0)
+ goto out;
+
+ ret = dpcm_be_dai_prepare(fe, stream);
+ if (ret < 0)
+ goto out;
+
if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) {
ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai);
if (ret < 0)
@@ -583,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
goto out;
}
- /*
- * Create an empty hw_params for the BE as the machine driver must
- * fix this up to match DSP decoder and ASRC configuration.
- * I.e. machine driver fixup for compressed BE is mandatory.
- */
- memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
- sizeof(struct snd_pcm_hw_params));
-
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
- ret = dpcm_be_dai_hw_params(fe, stream);
- if (ret < 0)
- goto out;
-
- ret = dpcm_be_dai_prepare(fe, stream);
- if (ret < 0)
- goto out;
-
dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4663de3cf495..9cfe10d8040c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1,26 +1,21 @@
-/*
- * soc-core.c -- ALSA SoC Audio Layer
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- * with code, comments and ideas from :-
- * Richard Purdie <richard@openedhand.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- * TODO:
- * o Add hw rules to enforce rates, etc.
- * o More testing with other codecs/machines.
- * o Add more codecs and platforms to ensure good API coverage.
- * o Support TDM on PCM and I2S
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-core.c -- ALSA SoC Audio Layer
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+// with code, comments and ideas from :-
+// Richard Purdie <richard@openedhand.com>
+//
+// TODO:
+// o Add hw rules to enforce rates, etc.
+// o More testing with other codecs/machines.
+// o Add more codecs and platforms to ensure good API coverage.
+// o Support TDM on PCM and I2S
#include <linux/module.h>
#include <linux/moduleparam.h>
@@ -533,6 +528,7 @@ int snd_soc_suspend(struct device *dev)
"ASoC: idle_bias_off CODEC on over suspend\n");
break;
}
+ /* fall through */
case SND_SOC_BIAS_OFF:
if (component->driver->suspend)
@@ -852,6 +848,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
const char *platform_name;
int i;
+ if (dai_link->ignore)
+ return 0;
+
dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name);
if (soc_is_dai_link_bound(card, dai_link)) {
@@ -1195,15 +1194,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link);
+static void soc_set_of_name_prefix(struct snd_soc_component *component)
+{
+ struct device_node *component_of_node = component->dev->of_node;
+ const char *str;
+ int ret;
+
+ if (!component_of_node && component->dev->parent)
+ component_of_node = component->dev->parent->of_node;
+
+ ret = of_property_read_string(component_of_node, "sound-name-prefix",
+ &str);
+ if (!ret)
+ component->name_prefix = str;
+}
+
static void soc_set_name_prefix(struct snd_soc_card *card,
struct snd_soc_component *component)
{
int i;
- if (card->codec_conf == NULL)
- return;
-
- for (i = 0; i < card->num_configs; i++) {
+ for (i = 0; i < card->num_configs && card->codec_conf; i++) {
struct snd_soc_codec_conf *map = &card->codec_conf[i];
struct device_node *component_of_node = component->dev->of_node;
@@ -1215,8 +1226,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
if (map->dev_name && strcmp(component->name, map->dev_name))
continue;
component->name_prefix = map->name_prefix;
- break;
+ return;
}
+
+ /*
+ * If there is no configuration table or no match in the table,
+ * check if a prefix is provided in the node
+ */
+ soc_set_of_name_prefix(component);
}
static int soc_probe_component(struct snd_soc_card *card,
@@ -1461,7 +1478,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
{
struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int i, ret;
+ struct snd_soc_rtdcom_list *rtdcom;
+ struct snd_soc_component *component;
+ int i, ret, num;
dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
card->name, rtd->num, order);
@@ -1507,9 +1526,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
soc_dpcm_debugfs_add(rtd);
#endif
+ num = rtd->num;
+
+ /*
+ * most drivers will register their PCMs using DAI link ordering but
+ * topology based drivers can use the DAI link id field to set PCM
+ * device number and then use rtd + a base offset of the BEs.
+ */
+ for_each_rtdcom(rtd, rtdcom) {
+ component = rtdcom->component;
+
+ if (!component->driver->use_dai_pcm_id)
+ continue;
+
+ if (rtd->dai_link->no_pcm)
+ num += component->driver->be_pcm_base;
+ else
+ num = rtd->dai_link->id;
+ }
+
if (cpu_dai->driver->compress_new) {
/*create compress_device"*/
- ret = cpu_dai->driver->compress_new(rtd, rtd->num);
+ ret = cpu_dai->driver->compress_new(rtd, num);
if (ret < 0) {
dev_err(card->dev, "ASoC: can't create compress %s\n",
dai_link->stream_name);
@@ -1519,7 +1557,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
if (!dai_link->params) {
/* create the pcm */
- ret = soc_new_pcm(rtd, rtd->num);
+ 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);
@@ -1846,6 +1884,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name);
#endif /* CONFIG_DMI */
+static void soc_check_tplg_fes(struct snd_soc_card *card)
+{
+ struct snd_soc_component *component;
+ const struct snd_soc_component_driver *comp_drv;
+ struct snd_soc_dai_link *dai_link;
+ int i;
+
+ list_for_each_entry(component, &component_list, list) {
+
+ /* does this component override FEs ? */
+ if (!component->driver->ignore_machine)
+ continue;
+
+ /* for this machine ? */
+ if (strcmp(component->driver->ignore_machine,
+ card->dev->driver->name))
+ continue;
+
+ /* machine matches, so override the rtd data */
+ for (i = 0; i < card->num_links; i++) {
+
+ dai_link = &card->dai_link[i];
+
+ /* ignore this FE */
+ if (dai_link->dynamic) {
+ dai_link->ignore = true;
+ continue;
+ }
+
+ dev_info(card->dev, "info: override FE DAI link %s\n",
+ card->dai_link[i].name);
+
+ /* override platform component */
+ dai_link->platform_name = component->name;
+
+ /* convert non BE into BE */
+ dai_link->no_pcm = 1;
+
+ /* override any BE fixups */
+ dai_link->be_hw_params_fixup =
+ component->driver->be_hw_params_fixup;
+
+ /* 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)
+ dai_link->stream_name = dai_link->name;
+ }
+
+ /* Inform userspace we are using alternate topology */
+ if (component->driver->topology_name_prefix) {
+
+ /* topology shortname created ? */
+ if (!card->topology_shortname_created) {
+ comp_drv = component->driver;
+
+ snprintf(card->topology_shortname, 32, "%s-%s",
+ comp_drv->topology_name_prefix,
+ card->name);
+ card->topology_shortname_created = true;
+ }
+
+ /* use topology shortname */
+ card->name = card->topology_shortname;
+ }
+ }
+}
+
static int snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
@@ -1855,6 +1961,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
mutex_lock(&client_mutex);
mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
+ /* check whether any platform is ignore machine FE and using topology */
+ 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]);
@@ -2523,6 +2632,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
/**
+ * snd_soc_dai_get_channel_map - Get DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ * 0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ * 0~num-1 uses
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot)
+{
+ if (dai->driver->ops->get_channel_map)
+ return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+ rx_num, rx_slot);
+ else
+ return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+
+/**
* snd_soc_dai_set_tristate - configure DAI system or master clock.
* @dai: DAI
* @tristate: tristate enable
@@ -3258,9 +3389,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
-static int snd_soc_of_get_slot_mask(struct device_node *np,
- const char *prop_name,
- unsigned int *mask)
+int snd_soc_of_get_slot_mask(struct device_node *np,
+ const char *prop_name,
+ unsigned int *mask)
{
u32 val;
const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
@@ -3275,6 +3406,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np,
return val;
}
+EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask);
int snd_soc_of_parse_tdm_slot(struct device_node *np,
unsigned int *tx_mask,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 229c12349803..7e96793050c9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1,27 +1,21 @@
-/*
- * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- * Features:
- * o Changes power status of internal codec blocks depending on the
- * dynamic configuration of codec internal audio paths and active
- * DACs/ADCs.
- * o Platform power domain - can support external components i.e. amps and
- * mic/headphone insertion events.
- * o Automatic Mic Bias support
- * o Jack insertion power event initiation - e.g. hp insertion will enable
- * sinks, dacs, etc
- * o Delayed power down of audio subsystem to reduce pops between a quick
- * device reopen.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//
+// Features:
+// o Changes power status of internal codec blocks depending on the
+// dynamic configuration of codec internal audio paths and active
+// DACs/ADCs.
+// o Platform power domain - can support external components i.e. amps and
+// mic/headphone insertion events.
+// o Automatic Mic Bias support
+// o Jack insertion power event initiation - e.g. hp insertion will enable
+// sinks, dacs, etc
+// o Delayed power down of audio subsystem to reduce pops between a quick
+// device reopen.
#include <linux/module.h>
#include <linux/moduleparam.h>
@@ -3662,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
struct snd_pcm_substream substream;
struct snd_pcm_hw_params *params = NULL;
struct snd_pcm_runtime *runtime = NULL;
- u64 fmt;
+ unsigned int fmt;
int ret;
if (WARN_ON(!config) ||
@@ -4073,6 +4067,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
continue;
}
+ /* let users know there is no DAI to link */
+ if (!dai_w->priv) {
+ dev_dbg(card->dev, "dai widget %s has no DAI\n",
+ dai_w->name);
+ continue;
+ }
+
dai = dai_w->priv;
/* ...find all widgets with the same stream and link them */
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 7ac745df1412..a9ea172a66a7 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -1,13 +1,8 @@
-/*
- * soc-devres.c -- ALSA SoC Audio Layer devres functions
- *
- * Copyright (C) 2013 Linaro Ltd
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-devres.c -- ALSA SoC Audio Layer devres functions
+//
+// Copyright (C) 2013 Linaro Ltd
#include <linux/module.h>
#include <linux/moduleparam.h>
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 56a541b9ff9e..52fd7af952a5 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -1,17 +1,8 @@
-/*
- * Copyright (C) 2013, Analog Devices Inc.
- * Author: Lars-Peter Clausen <lars@metafoo.de>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- * You should have received a copy of the GNU General Public License along
- * with this program; if not, write to the Free Software Foundation, Inc.,
- * 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright (C) 2013, Analog Devices Inc.
+// Author: Lars-Peter Clausen <lars@metafoo.de>
+
#include <linux/module.h>
#include <linux/init.h>
#include <linux/dmaengine.h>
@@ -197,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
case 32:
case 64:
if (addr_widths & (1 << (bits / 8)))
- hw.formats |= (1LL << i);
+ hw.formats |= pcm_format_to_bits(i);
break;
default:
/* Unsupported types */
@@ -343,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer(
static int dmaengine_copy_user(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- void *buf, unsigned long bytes)
+ void __user *buf, unsigned long bytes)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component =
@@ -359,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream,
int ret;
if (is_playback)
- if (copy_from_user(dma_ptr, (void __user *)buf, bytes))
+ if (copy_from_user(dma_ptr, buf, bytes))
return -EFAULT;
if (process) {
- ret = process(substream, channel, hwoff,
- (void __user *)buf, bytes);
+ ret = process(substream, channel, hwoff, (__force void *)buf, bytes);
if (ret < 0)
return ret;
}
if (!is_playback)
- if (copy_to_user((void __user *)buf, dma_ptr, bytes))
+ if (copy_to_user(buf, dma_ptr, bytes))
return -EFAULT;
return 0;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 026cd5347e53..1ff9175e9d5e 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -1,15 +1,10 @@
-/*
- * soc-io.c -- ASoC register I/O helpers
- *
- * Copyright 2009-2011 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-io.c -- ASoC register I/O helpers
+//
+// Copyright 2009-2011 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
#include <linux/i2c.h>
#include <linux/spi/spi.h>
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index b2b16044ae80..c7b990abdbaa 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -1,15 +1,10 @@
-/*
- * soc-jack.c -- ALSA SoC jack handling
- *
- * Copyright 2008 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-jack.c -- ALSA SoC jack handling
+//
+// Copyright 2008 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
#include <sound/jack.h>
#include <sound/soc.h>
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 7144a51ddfa9..592efb370c44 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -1,20 +1,15 @@
-/*
- * soc-ops.c -- Generic ASoC operations
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- * with code, comments and ideas from :-
- * Richard Purdie <richard@openedhand.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ops.c -- Generic ASoC operations
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+// with code, comments and ideas from :-
+// Richard Purdie <richard@openedhand.com>
#include <linux/module.h>
#include <linux/moduleparam.h>
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5e7ae47a9658..e8b98bfd4cf1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1,20 +1,14 @@
-/*
- * soc-pcm.c -- ALSA SoC PCM
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Authors: Liam Girdwood <lrg@ti.com>
- * Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-pcm.c -- ALSA SoC PCM
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Authors: Liam Girdwood <lrg@ti.com>
+// Mark Brown <broonie@opensource.wolfsonmicro.com>
#include <linux/kernel.h>
#include <linux/init.h>
@@ -448,6 +442,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
hw->rate_max = min_not_zero(hw->rate_max, rate_max);
}
+static int soc_pcm_components_close(struct snd_pcm_substream *substream,
+ struct snd_soc_component *last)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_rtdcom_list *rtdcom;
+ struct snd_soc_component *component;
+
+ for_each_rtdcom(rtd, rtdcom) {
+ component = rtdcom->component;
+
+ if (component == last)
+ break;
+
+ if (!component->driver->ops ||
+ !component->driver->ops->close)
+ continue;
+
+ component->driver->ops->close(substream);
+ }
+
+ return 0;
+}
+
/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
@@ -462,7 +479,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai;
const char *codec_dai_name = "multicodec";
- int i, ret = 0, __ret;
+ int i, ret = 0;
pinctrl_pm_select_default_state(cpu_dai->dev);
for (i = 0; i < rtd->num_codecs; i++)
@@ -486,7 +503,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
}
}
- ret = 0;
for_each_rtdcom(rtd, rtdcom) {
component = rtdcom->component;
@@ -494,16 +510,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
!component->driver->ops->open)
continue;
- __ret = component->driver->ops->open(substream);
- if (__ret < 0) {
+ ret = component->driver->ops->open(substream);
+ if (ret < 0) {
dev_err(component->dev,
"ASoC: can't open component %s: %d\n",
- component->name, __ret);
- ret = __ret;
+ component->name, ret);
+ goto component_err;
}
}
- if (ret < 0)
- goto component_err;
+ component = NULL;
for (i = 0; i < rtd->num_codecs; i++) {
codec_dai = rtd->codec_dais[i];
@@ -612,15 +627,7 @@ codec_dai_err:
}
component_err:
- for_each_rtdcom(rtd, rtdcom) {
- component = rtdcom->component;
-
- if (!component->driver->ops ||
- !component->driver->ops->close)
- continue;
-
- component->driver->ops->close(substream);
- }
+ soc_pcm_components_close(substream, component);
if (cpu_dai->driver->ops->shutdown)
cpu_dai->driver->ops->shutdown(substream, cpu_dai);
@@ -714,15 +721,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
if (rtd->dai_link->ops->shutdown)
rtd->dai_link->ops->shutdown(substream);
- for_each_rtdcom(rtd, rtdcom) {
- component = rtdcom->component;
-
- if (!component->driver->ops ||
- !component->driver->ops->close)
- continue;
-
- component->driver->ops->close(substream);
- }
+ soc_pcm_components_close(substream, NULL);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
@@ -860,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
int ret;
+ /* perform any topology hw_params fixups before DAI */
+ if (rtd->dai_link->be_hw_params_fixup) {
+ ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "ASoC: hw_params topology fixup failed %d\n",
+ ret);
+ return ret;
+ }
+ }
+
if (dai->driver->ops->hw_params) {
ret = dai->driver->ops->hw_params(substream, params, dai);
if (ret < 0) {
@@ -874,6 +885,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_component *last)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_rtdcom_list *rtdcom;
+ struct snd_soc_component *component;
+
+ for_each_rtdcom(rtd, rtdcom) {
+ component = rtdcom->component;
+
+ if (component == last)
+ break;
+
+ if (!component->driver->ops ||
+ !component->driver->ops->hw_free)
+ continue;
+
+ component->driver->ops->hw_free(substream);
+ }
+
+ return 0;
+}
+
/*
* Called by ALSA when the hardware params are set by application. This
* function can also be called multiple times and can allocate buffers
@@ -886,7 +920,7 @@ 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;
- int i, ret = 0, __ret;
+ int i, ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (rtd->dai_link->ops->hw_params) {
@@ -944,7 +978,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
goto interface_err;
- ret = 0;
for_each_rtdcom(rtd, rtdcom) {
component = rtdcom->component;
@@ -952,16 +985,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
!component->driver->ops->hw_params)
continue;
- __ret = component->driver->ops->hw_params(substream, params);
- if (__ret < 0) {
+ ret = component->driver->ops->hw_params(substream, params);
+ if (ret < 0) {
dev_err(component->dev,
"ASoC: %s hw params failed: %d\n",
- component->name, __ret);
- ret = __ret;
+ component->name, ret);
+ goto component_err;
}
}
- if (ret < 0)
- goto component_err;
+ component = NULL;
/* store the parameters for each DAIs */
cpu_dai->rate = params_rate(params);
@@ -977,15 +1009,7 @@ out:
return ret;
component_err:
- for_each_rtdcom(rtd, rtdcom) {
- component = rtdcom->component;
-
- if (!component->driver->ops ||
- !component->driver->ops->hw_free)
- continue;
-
- component->driver->ops->hw_free(substream);
- }
+ soc_pcm_components_hw_free(substream, component);
if (cpu_dai->driver->ops->hw_free)
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
@@ -1014,8 +1038,6 @@ codec_err:
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- 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;
bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
@@ -1052,15 +1074,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
rtd->dai_link->ops->hw_free(substream);
/* free any component resources */
- for_each_rtdcom(rtd, rtdcom) {
- component = rtdcom->component;
-
- if (!component->driver->ops ||
- !component->driver->ops->hw_free)
- continue;
-
- component->driver->ops->hw_free(substream);
- }
+ soc_pcm_components_hw_free(substream, NULL);
/* now free hw params for the DAIs */
for (i = 0; i < rtd->num_codecs; i++) {
@@ -1165,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
snd_pcm_sframes_t codec_delay = 0;
int i;
+ /* clearing the previous total delay */
+ runtime->delay = 0;
+
for_each_rtdcom(rtd, rtdcom) {
component = rtdcom->component;
@@ -1176,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
offset = component->driver->ops->pointer(substream);
break;
}
+ /* base delay if assigned in pointer callback */
+ delay = runtime->delay;
if (cpu_dai->driver->ops->delay)
delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
@@ -1658,29 +1677,28 @@ unwind:
}
static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
- struct snd_soc_pcm_stream *stream,
- u64 formats)
+ struct snd_soc_pcm_stream *stream)
{
runtime->hw.rate_min = stream->rate_min;
runtime->hw.rate_max = stream->rate_max;
runtime->hw.channels_min = stream->channels_min;
runtime->hw.channels_max = stream->channels_max;
if (runtime->hw.formats)
- runtime->hw.formats &= formats & stream->formats;
+ runtime->hw.formats &= stream->formats;
else
- runtime->hw.formats = formats & stream->formats;
+ runtime->hw.formats = stream->formats;
runtime->hw.rates = stream->rates;
}
-static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
+static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
+ u64 *formats)
{
struct snd_soc_pcm_runtime *fe = substream->private_data;
struct snd_soc_dpcm *dpcm;
- u64 formats = ULLONG_MAX;
int stream = substream->stream;
if (!fe->dai_link->dpcm_merged_format)
- return formats;
+ return;
/*
* It returns merged BE codec format
@@ -1694,17 +1712,132 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
int i;
for (i = 0; i < be->num_codecs; i++) {
+ /*
+ * 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))
+ continue;
+
codec_dai_drv = be->codec_dais[i]->driver;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
else
codec_stream = &codec_dai_drv->capture;
- formats &= codec_stream->formats;
+ *formats &= codec_stream->formats;
}
}
+}
+
+static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
+ unsigned int *channels_min,
+ unsigned int *channels_max)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct snd_soc_dpcm *dpcm;
+ int stream = substream->stream;
- return formats;
+ if (!fe->dai_link->dpcm_merged_chan)
+ return;
+
+ /*
+ * It returns merged BE codec channel;
+ * if FE want to use it (= dpcm_merged_chan)
+ */
+
+ list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ 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;
+ struct snd_soc_pcm_stream *codec_stream;
+ struct snd_soc_pcm_stream *cpu_stream;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_stream = &cpu_dai_drv->playback;
+ else
+ cpu_stream = &cpu_dai_drv->capture;
+
+ *channels_min = max(*channels_min, cpu_stream->channels_min);
+ *channels_max = min(*channels_max, cpu_stream->channels_max);
+
+ /*
+ * chan min/max cannot be enforced if there are multiple CODEC
+ * DAIs connected to a single CPU DAI, use CPU DAI's directly
+ */
+ if (be->num_codecs == 1) {
+ codec_dai_drv = be->codec_dais[0]->driver;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &codec_dai_drv->playback;
+ else
+ codec_stream = &codec_dai_drv->capture;
+
+ *channels_min = max(*channels_min,
+ codec_stream->channels_min);
+ *channels_max = min(*channels_max,
+ codec_stream->channels_max);
+ }
+ }
+}
+
+static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
+ unsigned int *rates,
+ unsigned int *rate_min,
+ unsigned int *rate_max)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct snd_soc_dpcm *dpcm;
+ int stream = substream->stream;
+
+ if (!fe->dai_link->dpcm_merged_rate)
+ return;
+
+ /*
+ * It returns merged BE codec channel;
+ * if FE want to use it (= dpcm_merged_chan)
+ */
+
+ list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ 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;
+ struct snd_soc_pcm_stream *codec_stream;
+ struct snd_soc_pcm_stream *cpu_stream;
+ int i;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_stream = &cpu_dai_drv->playback;
+ else
+ cpu_stream = &cpu_dai_drv->capture;
+
+ *rate_min = max(*rate_min, cpu_stream->rate_min);
+ *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++) {
+ /*
+ * 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))
+ continue;
+
+ codec_dai_drv = be->codec_dais[i]->driver;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &codec_dai_drv->playback;
+ else
+ codec_stream = &codec_dai_drv->capture;
+
+ *rate_min = max(*rate_min, codec_stream->rate_min);
+ *rate_max = min_not_zero(*rate_max,
+ codec_stream->rate_max);
+ *rates = snd_pcm_rate_mask_intersect(*rates,
+ codec_stream->rates);
+ }
+ }
}
static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
@@ -1713,12 +1846,17 @@ static void dpcm_set_fe_runtime(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_driver *cpu_dai_drv = cpu_dai->driver;
- u64 format = dpcm_runtime_base_format(substream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format);
+ dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
else
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format);
+ dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+
+ dpcm_runtime_merge_format(substream, &runtime->hw.formats);
+ dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min,
+ &runtime->hw.channels_max);
+ dpcm_runtime_merge_rate(substream, &runtime->hw.rates,
+ &runtime->hw.rate_min, &runtime->hw.rate_max);
}
static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);
@@ -2543,106 +2681,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
return ret;
}
-/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
- * any DAI links.
- */
-int soc_dpcm_runtime_update(struct snd_soc_card *card)
+static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
{
- struct snd_soc_pcm_runtime *fe;
- int old, new, paths;
+ struct snd_soc_dapm_widget_list *list;
+ int count, paths;
- mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- list_for_each_entry(fe, &card->rtd_list, list) {
- struct snd_soc_dapm_widget_list *list;
+ if (!fe->dai_link->dynamic)
+ return 0;
- /* make sure link is FE */
- if (!fe->dai_link->dynamic)
- continue;
+ /* only check active links */
+ if (!fe->cpu_dai->active)
+ return 0;
- /* only check active links */
- if (!fe->cpu_dai->active)
- continue;
+ /* DAPM sync will call this to update DSP paths */
+ dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
+ new ? "new" : "old", fe->dai_link->name);
- /* DAPM sync will call this to update DSP paths */
- dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n",
- fe->dai_link->name);
+ /* skip if FE doesn't have playback capability */
+ if (!fe->cpu_dai->driver->playback.channels_min ||
+ !fe->codec_dai->driver->playback.channels_min)
+ goto capture;
- /* skip if FE doesn't have playback capability */
- if (!fe->cpu_dai->driver->playback.channels_min
- || !fe->codec_dai->driver->playback.channels_min)
- goto capture;
-
- /* skip if FE isn't currently playing */
- if (!fe->cpu_dai->playback_active
- || !fe->codec_dai->playback_active)
- goto capture;
-
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "playback");
- mutex_unlock(&card->mutex);
- return paths;
- }
+ /* skip if FE isn't currently playing */
+ if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
+ goto capture;
- /* update any new playback paths */
- new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1);
- if (new) {
- dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
- }
+ paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+ fe->dai_link->name, "playback");
+ return paths;
+ }
- /* update any old playback paths */
- old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0);
- if (old) {
+ /* update any playback paths */
+ count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
+ if (count) {
+ if (new)
+ dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ else
dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
- }
- dpcm_path_put(&list);
+ dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+ dpcm_path_put(&list);
+
capture:
- /* skip if FE doesn't have capture capability */
- if (!fe->cpu_dai->driver->capture.channels_min
- || !fe->codec_dai->driver->capture.channels_min)
- continue;
+ /* skip if FE doesn't have capture capability */
+ if (!fe->cpu_dai->driver->capture.channels_min ||
+ !fe->codec_dai->driver->capture.channels_min)
+ return 0;
- /* skip if FE isn't currently capturing */
- if (!fe->cpu_dai->capture_active
- || !fe->codec_dai->capture_active)
- continue;
+ /* skip if FE isn't currently capturing */
+ if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
+ return 0;
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "capture");
- mutex_unlock(&card->mutex);
- return paths;
- }
+ paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+ fe->dai_link->name, "capture");
+ return paths;
+ }
- /* update any new capture paths */
- new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1);
- if (new) {
+ /* update any old capture paths */
+ count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
+ if (count) {
+ if (new)
dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
- }
-
- /* update any old capture paths */
- old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0);
- if (old) {
+ else
dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
- }
- dpcm_path_put(&list);
+ dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
+ dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
}
- mutex_unlock(&card->mutex);
+ dpcm_path_put(&list);
+
return 0;
}
+
+/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
+ * any DAI links.
+ */
+int soc_dpcm_runtime_update(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *fe;
+ int ret = 0;
+
+ mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ /* shutdown all old paths first */
+ list_for_each_entry(fe, &card->rtd_list, list) {
+ 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) {
+ ret = soc_dpcm_fe_runtime_update(fe, 1);
+ if (ret)
+ goto out;
+ }
+
+out:
+ mutex_unlock(&card->mutex);
+ return ret;
+}
int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
{
struct snd_soc_dpcm *dpcm;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 53f121a50c97..66e77e020745 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1,29 +1,24 @@
-/*
- * soc-topology.c -- ALSA SoC Topology
- *
- * Copyright (C) 2012 Texas Instruments Inc.
- * Copyright (C) 2015 Intel Corporation.
- *
- * Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
- * K, Mythri P <mythri.p.k@intel.com>
- * Prusty, Subhransu S <subhransu.s.prusty@intel.com>
- * B, Jayachandran <jayachandran.b@intel.com>
- * Abdullah, Omair M <omair.m.abdullah@intel.com>
- * Jin, Yao <yao.jin@intel.com>
- * Lin, Mengdong <mengdong.lin@intel.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- * Add support to read audio firmware topology alongside firmware text. The
- * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
- * equalizers, firmware, coefficients etc.
- *
- * This file only manages the core ALSA and ASoC components, all other bespoke
- * firmware topology data is passed to component drivers for bespoke handling.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-topology.c -- ALSA SoC Topology
+//
+// Copyright (C) 2012 Texas Instruments Inc.
+// Copyright (C) 2015 Intel Corporation.
+//
+// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
+// K, Mythri P <mythri.p.k@intel.com>
+// Prusty, Subhransu S <subhransu.s.prusty@intel.com>
+// B, Jayachandran <jayachandran.b@intel.com>
+// Abdullah, Omair M <omair.m.abdullah@intel.com>
+// Jin, Yao <yao.jin@intel.com>
+// Lin, Mengdong <mengdong.lin@intel.com>
+//
+// Add support to read audio firmware topology alongside firmware text. The
+// topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
+// equalizers, firmware, coefficients etc.
+//
+// This file only manages the core ALSA and ASoC components, all other bespoke
+// firmware topology data is passed to component drivers for bespoke handling.
#include <linux/kernel.h>
#include <linux/export.h>
@@ -259,7 +254,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg,
int ret = 0;
if (tplg->comp && tplg->ops && tplg->ops->vendor_load)
- ret = tplg->ops->vendor_load(tplg->comp, hdr);
+ ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr);
else {
dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n",
hdr->vendor_type);
@@ -291,7 +286,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
{
if (tplg->comp && tplg->ops && tplg->ops->widget_load)
- return tplg->ops->widget_load(tplg->comp, w, tplg_w);
+ return tplg->ops->widget_load(tplg->comp, tplg->index, w,
+ tplg_w);
return 0;
}
@@ -302,27 +298,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg,
struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
{
if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
- return tplg->ops->widget_ready(tplg->comp, w, tplg_w);
+ return tplg->ops->widget_ready(tplg->comp, tplg->index, w,
+ tplg_w);
return 0;
}
/* pass DAI configurations to component driver for extra initialization */
static int soc_tplg_dai_load(struct soc_tplg *tplg,
- struct snd_soc_dai_driver *dai_drv)
+ struct snd_soc_dai_driver *dai_drv,
+ struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
{
if (tplg->comp && tplg->ops && tplg->ops->dai_load)
- return tplg->ops->dai_load(tplg->comp, dai_drv);
+ return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv,
+ pcm, dai);
return 0;
}
/* pass link configurations to component driver for extra initialization */
static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
- struct snd_soc_dai_link *link)
+ struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg)
{
if (tplg->comp && tplg->ops && tplg->ops->link_load)
- return tplg->ops->link_load(tplg->comp, link);
+ return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg);
return 0;
}
@@ -643,7 +642,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr)
{
if (tplg->comp && tplg->ops && tplg->ops->control_load)
- return tplg->ops->control_load(tplg->comp, k, hdr);
+ return tplg->ops->control_load(tplg->comp, tplg->index, k,
+ hdr);
return 0;
}
@@ -1100,6 +1100,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
return 0;
}
+/* optionally pass new dynamic kcontrol to component driver. */
+static int soc_tplg_add_route(struct soc_tplg *tplg,
+ struct snd_soc_dapm_route *route)
+{
+ if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load)
+ return tplg->ops->dapm_route_load(tplg->comp, tplg->index,
+ route);
+
+ return 0;
+}
+
static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
@@ -1148,6 +1159,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
else
route.control = elem->control;
+ soc_tplg_add_route(tplg, &route);
+
/* add route, but keep going if some fail */
snd_soc_dapm_add_routes(dapm, &route, 1);
}
@@ -1702,7 +1715,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
dai_drv->compress_new = snd_soc_new_compress;
/* pass control to component driver for optional further init */
- ret = soc_tplg_dai_load(tplg, dai_drv);
+ ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
kfree(dai_drv);
@@ -1772,7 +1785,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
set_link_flags(link, pcm->flag_mask, pcm->flags);
/* pass control to component driver for optional further init */
- ret = soc_tplg_dai_link_load(tplg, link);
+ ret = soc_tplg_dai_link_load(tplg, link, NULL);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
kfree(link);
@@ -2080,7 +2093,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg,
set_link_flags(link, cfg->flag_mask, cfg->flags);
/* pass control to component driver for optional further init */
- ret = soc_tplg_dai_link_load(tplg, link);
+ ret = soc_tplg_dai_link_load(tplg, link, cfg);
if (ret < 0) {
dev_err(tplg->dev, "ASoC: physical link loading failed\n");
return ret;
@@ -2202,7 +2215,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
set_dai_flags(dai_drv, d->flag_mask, d->flags);
/* pass control to component driver for optional further init */
- ret = soc_tplg_dai_load(tplg, dai_drv);
+ ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
return ret;
@@ -2311,7 +2324,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
/* pass control to component driver for optional further init */
if (tplg->comp && tplg->ops && tplg->ops->manifest)
- return tplg->ops->manifest(tplg->comp, _manifest);
+ return tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
if (!abi_match) /* free the duplicated one */
kfree(_manifest);
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 2d9e98bd1530..e0c93496c0cd 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -1,17 +1,11 @@
-/*
- * soc-util.c -- ALSA SoC Audio Layer utility functions
- *
- * Copyright 2009 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- * Liam Girdwood <lrg@slimlogic.co.uk>
- *
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-util.c -- ALSA SoC Audio Layer utility functions
+//
+// Copyright 2009 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+// Liam Girdwood <lrg@slimlogic.co.uk>
#include <linux/platform_device.h>
#include <linux/export.h>
@@ -381,6 +375,6 @@ int __init snd_soc_util_init(void)
void __exit snd_soc_util_exit(void)
{
- platform_device_unregister(soc_dummy_dev);
platform_driver_unregister(&soc_dummy_driver);
+ platform_device_unregister(soc_dummy_dev);
}
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index d8b6936e544e..313dab2857ef 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player);
/* Stop the player */
- snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stop_xrun(player->substream);
}
ret = IRQ_HANDLED;
@@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
/* Stop the player */
- snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stop_xrun(player->substream);
ret = IRQ_HANDLED;
}
@@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
dev_err(player->dev, "Underflow recovery failed\n");
/* Stop the player */
- snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stop_xrun(player->substream);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index ee0055e60852..7b63d35ef428 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
dev_err(reader->dev, "FIFO error detected\n");
- snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stop_xrun(reader->substream);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 48f9ddd94016..9b2681397dba 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI
depends on SND_SOC
select SND_SOC_GENERIC_DMAENGINE_PCM
select REGMAP_MMIO
+ select SND_PCM_IEC958
help
Say Y if you want to enable SAI for STM32
diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c
index db73fef3e500..706ff005234f 100644
--- a/sound/soc/stm/stm32_adfsdm.c
+++ b/sound/soc/stm/stm32_adfsdm.c
@@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private)
unsigned int old_pos = priv->pos;
unsigned int cur_size = size;
- dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n",
+ dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n",
__func__, &pcm_buff[priv->pos], priv->pos, size);
if ((priv->pos + size) > buff_size) {
@@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd)
static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
- struct snd_soc_pcm_runtime *rtd;
- struct stm32_adfsdm_priv *priv;
substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
- if (substream) {
- rtd = substream->private_data;
- priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-
+ if (substream)
snd_pcm_lib_preallocate_free_for_all(pcm);
- }
}
static struct snd_soc_component_driver stm32_adfsdm_soc_platform = {
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index cfeb219e1d78..06fba9650ac4 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -96,7 +96,8 @@
* @slot_mask: rx or tx active slots mask. set at init or at runtime
* @data_size: PCM data width. corresponds to PCM substream width.
* @spdif_frm_cnt: S/PDIF playback frame counter
- * @spdif_status_bits: S/PDIF status bits
+ * @snd_aes_iec958: iec958 data
+ * @ctrl_lock: control lock
*/
struct stm32_sai_sub_data {
struct platform_device *pdev;
@@ -125,7 +126,8 @@ struct stm32_sai_sub_data {
int slot_mask;
int data_size;
unsigned int spdif_frm_cnt;
- unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES];
+ struct snd_aes_iec958 iec958;
+ struct mutex ctrl_lock; /* protect resources accessed by controls */
};
enum stm32_sai_fifo_th {
@@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg)
}
}
-static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = {
- 0, 0, 0, IEC958_AES3_CON_FS_48000,
-};
-
static const struct regmap_config stm32_sai_sub_regmap_config_f4 = {
.reg_bits = 32,
.reg_stride = 4,
@@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = {
.fast_io = true,
};
+static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+
+ return 0;
+}
+
+static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&sai->ctrl_lock);
+ memcpy(uctl->value.iec958.status, sai->iec958.status, 4);
+ mutex_unlock(&sai->ctrl_lock);
+
+ return 0;
+}
+
+static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&sai->ctrl_lock);
+ memcpy(sai->iec958.status, uctl->value.iec958.status, 4);
+ mutex_unlock(&sai->ctrl_lock);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new iec958_ctls = {
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .info = snd_pcm_iec958_info,
+ .get = snd_pcm_iec958_get,
+ .put = snd_pcm_iec958_put,
+};
+
static irqreturn_t stm32_sai_isr(int irq, void *devid)
{
struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -259,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid)
status = SNDRV_PCM_STATE_XRUN;
}
- if (status != SNDRV_PCM_STATE_RUNNING) {
- snd_pcm_stream_lock(sai->substream);
- snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN);
- snd_pcm_stream_unlock(sai->substream);
- }
+ if (status != SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_stop_xrun(sai->substream);
return IRQ_HANDLED;
}
@@ -619,6 +657,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai)
}
}
+static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai)
+{
+ unsigned char *cs = sai->iec958.status;
+
+ cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE;
+ cs[1] = IEC958_AES1_CON_GENERAL;
+ cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC;
+ cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID;
+}
+
+static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai,
+ struct snd_pcm_runtime *runtime)
+{
+ if (!runtime)
+ return;
+
+ /* Force the sample rate according to runtime rate */
+ mutex_lock(&sai->ctrl_lock);
+ switch (runtime->rate) {
+ case 22050:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_22050;
+ break;
+ case 44100:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_44100;
+ break;
+ case 88200:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_88200;
+ break;
+ case 176400:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_176400;
+ break;
+ case 24000:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_24000;
+ break;
+ case 48000:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_48000;
+ break;
+ case 96000:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_96000;
+ break;
+ case 192000:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_192000;
+ break;
+ case 32000:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_32000;
+ break;
+ default:
+ sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID;
+ break;
+ }
+ mutex_unlock(&sai->ctrl_lock);
+}
+
static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
struct snd_pcm_hw_params *params)
{
@@ -709,7 +800,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
sai->data_size = params_width(params);
- if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+ /* Rate not already set in runtime structure */
+ substream->runtime->rate = params_rate(params);
+ stm32_sai_set_iec958_status(sai, substream->runtime);
+ } else {
ret = stm32_sai_set_slots(cpu_dai);
if (ret < 0)
return ret;
@@ -789,6 +884,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
sai->substream = NULL;
}
+static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
+
+ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+ dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__);
+ return snd_ctl_add(rtd->pcm->card,
+ snd_ctl_new1(&iec958_ctls, sai));
+ }
+
+ return 0;
+}
+
static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
{
struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
@@ -809,6 +918,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
else
snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params);
+ /* Next settings are not relevant for spdif mode */
+ if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
+ return 0;
+
cr1_mask = SAI_XCR1_RX_TX;
if (STM_SAI_IS_CAPTURE(sai))
cr1 |= SAI_XCR1_RX_TX;
@@ -820,10 +933,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
sai->synco, sai->synci);
}
- if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
- memcpy(sai->spdif_status_bits, default_status_bits,
- sizeof(default_status_bits));
-
cr1_mask |= SAI_XCR1_SYNCEN_MASK;
cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync);
@@ -861,7 +970,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
/* Set channel status bit */
byte = frm_cnt >> 3;
mask = 1 << (frm_cnt - (byte << 3));
- if (sai->spdif_status_bits[byte] & mask)
+ if (sai->iec958.status[byte] & mask)
*ptr |= 0x04000000;
ptr++;
@@ -888,6 +997,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = {
static struct snd_soc_dai_driver stm32_sai_playback_dai[] = {
{
.probe = stm32_sai_dai_probe,
+ .pcm_new = stm32_sai_pcm_new,
.id = 1, /* avoid call to fmt_single_name() */
.playback = {
.channels_min = 1,
@@ -998,6 +1108,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n");
return -EINVAL;
}
+ stm32_sai_init_iec958_status(sai);
sai->spdif = true;
sai->master = true;
}
@@ -1114,6 +1225,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
sai->id = (uintptr_t)of_id->data;
sai->pdev = pdev;
+ mutex_init(&sai->ctrl_lock);
platform_set_drvdata(pdev, sai);
sai->pdata = dev_get_drvdata(pdev->dev.parent);
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index affad46bf188..682ef33afb5f 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
ret = clk_prepare_enable(ac97->clk_ac97);
if (ret) {
dev_err(&pdev->dev, "clk_enable failed: %d\n", ret);
- goto err;
+ goto err_clk_put;
}
ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops);
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 774fc6ad2026..2e561e946de2 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -173,7 +173,7 @@
/* Number of slots in frame, minus 1 */
#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16
#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7
-#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT)
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT)
/* TDM mode slot enable bitmask */
#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 5197d6b18cb6..98d87801d57a 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
- goto err;
+ goto err_put_codec_of_node;
}
tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
if (ret)
- goto err;
+ goto err_put_cpu_of_node;
ret = snd_soc_register_card(card);
if (ret) {
@@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
err_fini_utils:
tegra_asoc_utils_fini(&alc5632->util_data);
+err_put_cpu_of_node:
+ of_node_put(tegra_alc5632_dai.cpu_of_node);
+ tegra_alc5632_dai.cpu_of_node = NULL;
+ tegra_alc5632_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+ of_node_put(tegra_alc5632_dai.codec_of_node);
+ tegra_alc5632_dai.codec_of_node = NULL;
err:
return ret;
}
@@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev)
tegra_asoc_utils_fini(&machine->util_data);
+ of_node_put(tegra_alc5632_dai.cpu_of_node);
+ tegra_alc5632_dai.cpu_of_node = NULL;
+ tegra_alc5632_dai.platform_of_node = NULL;
+ of_node_put(tegra_alc5632_dai.codec_of_node);
+ tegra_alc5632_dai.codec_of_node = NULL;
+
return 0;
}
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index 0e4805c7b4ca..7081f15302cc 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing or invalid\n");
ret = -EINVAL;
- goto err;
+ goto err_put_codec_of_node;
}
tegra_rt5677_dai.platform_of_node = tegra_rt5677_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) {
@@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
err_fini_utils:
tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+ of_node_put(tegra_rt5677_dai.cpu_of_node);
+ tegra_rt5677_dai.cpu_of_node = NULL;
+ tegra_rt5677_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+ of_node_put(tegra_rt5677_dai.codec_of_node);
+ tegra_rt5677_dai.codec_of_node = NULL;
err:
return ret;
}
@@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev)
tegra_asoc_utils_fini(&machine->util_data);
+ tegra_rt5677_dai.platform_of_node = NULL;
+ of_node_put(tegra_rt5677_dai.codec_of_node);
+ tegra_rt5677_dai.codec_of_node = NULL;
+ of_node_put(tegra_rt5677_dai.cpu_of_node);
+ tegra_rt5677_dai.cpu_of_node = NULL;
+
return 0;
}
diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c
index 638cb3fc5f7b..9bcba06ba52e 100644
--- a/sound/soc/uniphier/aio-core.c
+++ b/sound/soc/uniphier/aio-core.c
@@ -265,6 +265,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
}
/**
+ * aio_port_set_ch - set channels of LPCM
+ * @sub: the AIO substream pointer, PCM substream only
+ * @ch : count of channels
+ *
+ * Set suitable slot selecting to input/output port block of AIO.
+ *
+ * This function may return error if non-PCM substream.
+ *
+ * Return: Zero if successful, otherwise a negative value on error.
+ */
+static int aio_port_set_ch(struct uniphier_aio_sub *sub)
+{
+ struct regmap *r = sub->aio->chip->regmap;
+ u32 slotsel_2ch[] = {
+ 0, 0, 0, 0, 0,
+ };
+ u32 slotsel_multi[] = {
+ OPORTMXTYSLOTCTR_SLOTSEL_SLOT0,
+ OPORTMXTYSLOTCTR_SLOTSEL_SLOT1,
+ OPORTMXTYSLOTCTR_SLOTSEL_SLOT2,
+ OPORTMXTYSLOTCTR_SLOTSEL_SLOT3,
+ OPORTMXTYSLOTCTR_SLOTSEL_SLOT4,
+ };
+ u32 mode, *slotsel;
+ int i;
+
+ switch (params_channels(&sub->params)) {
+ case 8:
+ case 6:
+ mode = OPORTMXTYSLOTCTR_MODE;
+ slotsel = slotsel_multi;
+ break;
+ case 2:
+ mode = 0;
+ slotsel = slotsel_2ch;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ for (i = 0; i < AUD_MAX_SLOTSEL; i++) {
+ regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+ OPORTMXTYSLOTCTR_MODE, mode);
+ regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+ OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]);
+ }
+
+ return 0;
+}
+
+/**
* aio_port_set_rate - set sampling rate of LPCM
* @sub: the AIO substream pointer, PCM substream only
* @rate: Sampling rate in Hz.
@@ -276,7 +327,7 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
*
* Return: Zero if successful, otherwise a negative value on error.
*/
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
+static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
{
struct regmap *r = sub->aio->chip->regmap;
struct device *dev = &sub->aio->chip->pdev->dev;
@@ -395,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
*
* Return: Zero if successful, otherwise a negative value on error.
*/
-int aio_port_set_fmt(struct uniphier_aio_sub *sub)
+static int aio_port_set_fmt(struct uniphier_aio_sub *sub)
{
struct regmap *r = sub->aio->chip->regmap;
struct device *dev = &sub->aio->chip->pdev->dev;
@@ -460,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub)
*
* Return: Zero if successful, otherwise a negative value on error.
*/
-int aio_port_set_clk(struct uniphier_aio_sub *sub)
+static int aio_port_set_clk(struct uniphier_aio_sub *sub)
{
struct uniphier_aio_chip *chip = sub->aio->chip;
struct device *dev = &sub->aio->chip->pdev->dev;
@@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
rate = params_rate(params);
}
+ ret = aio_port_set_ch(sub);
+ if (ret)
+ return ret;
+
ret = aio_port_set_rate(sub, rate);
if (ret)
return ret;
@@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol)
int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through)
{
struct regmap *r = sub->aio->chip->regmap;
- u32 v;
+ u32 memfmt, v;
if (sub->swm->dir == PORT_DIR_OUTPUT) {
- if (pass_through)
+ if (pass_through) {
v = PBOUTMXCTR0_ENDIAN_0123 |
PBOUTMXCTR0_MEMFMT_STREAM;
- else
- v = PBOUTMXCTR0_ENDIAN_3210 |
- PBOUTMXCTR0_MEMFMT_2CH;
+ } else {
+ switch (params_channels(&sub->params)) {
+ case 2:
+ memfmt = PBOUTMXCTR0_MEMFMT_2CH;
+ break;
+ case 6:
+ memfmt = PBOUTMXCTR0_MEMFMT_6CH;
+ break;
+ case 8:
+ memfmt = PBOUTMXCTR0_MEMFMT_8CH;
+ break;
+ default:
+ return -EINVAL;
+ }
+ v = PBOUTMXCTR0_ENDIAN_3210 | memfmt;
+ }
regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v);
regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0);
diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c
index 2d9b7dde2ffa..ee90e6c3937c 100644
--- a/sound/soc/uniphier/aio-cpu.c
+++ b/sound/soc/uniphier/aio-cpu.c
@@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id,
unsigned int freq_out)
{
struct uniphier_aio *aio = uniphier_priv(dai);
- struct device *dev = &aio->chip->pdev->dev;
int ret;
if (!is_valid_pll(aio->chip, pll_id))
return -EINVAL;
- if (!aio->chip->plls[pll_id].enable) {
- dev_err(dev, "PLL(%d) is not implemented\n", pll_id);
- return -ENOTSUPP;
- }
ret = aio_chip_set_pll(aio->chip, pll_id, freq_out);
if (ret < 0)
diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c
index ab04d3331be9..de962df245ba 100644
--- a/sound/soc/uniphier/aio-ld11.c
+++ b/sound/soc/uniphier/aio-ld11.c
@@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = {
.formats = SNDRV_PCM_FMTBIT_S32_LE,
.rates = SNDRV_PCM_RATE_48000,
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 8,
},
.ops = &uniphier_aio_i2s_ops,
},
diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h
index 45fdc6ae358a..734395dbcffb 100644
--- a/sound/soc/uniphier/aio-reg.h
+++ b/sound/soc/uniphier/aio-reg.h
@@ -374,6 +374,7 @@
#define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m))
#define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0)
#define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m))
+#define OPORTMXTYSLOTCTR_MODE BIT(15)
#define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8)
#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8)
#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8)
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index aa89c2f6fa24..ca6ccbae0ee8 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -141,6 +141,9 @@ enum IEC61937_PC {
#define AUD_MIN_FRAGMENT_SIZE (4 * 1024)
#define AUD_MAX_FRAGMENT_SIZE (16 * 1024)
+/* max 5 slots, 10 channels, 2 channel in 1 slot */
+#define AUD_MAX_SLOTSEL 5
+
/*
* This is a selector for virtual register map of AIO.
*
@@ -322,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id,
void aio_chip_init(struct uniphier_aio_chip *chip);
int aio_init(struct uniphier_aio_sub *sub);
void aio_port_reset(struct uniphier_aio_sub *sub);
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate);
-int aio_port_set_fmt(struct uniphier_aio_sub *sub);
-int aio_port_set_clk(struct uniphier_aio_sub *sub);
int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
const struct snd_pcm_hw_params *params);
void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable);
diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index dc955272f58b..389272eeba9a 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on)
#define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
#define ZX_TDM_FMTBIT \
- (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \
- SNDRV_PCM_FORMAT_A_LAW)
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \
+ SNDRV_PCM_FMTBIT_A_LAW)
static int zx_tdm_dai_probe(struct snd_soc_dai *dai)
{