From 6bd3c6f75e0f9baddbf1196a7e3fceabb50c7e3c Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Mon, 31 Aug 2015 17:07:12 +0200 Subject: ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device In case of unknown DT compatible device the ASRC OF node possibly acquired earlier by of_parse_phandle() has to be put before returning from probe method. Signed-off-by: Maciej Szmigiero Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed4827e..96f55ae75c71 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); - return -EINVAL; + ret = -EINVAL; + goto asrc_fail; } /* Common settings for corresponding Freescale CPU DAI driver */ -- cgit v1.2.3-58-ga151 From d76f41982f2fc88492efd96c7c3178044f32e125 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Mon, 31 Aug 2015 08:24:23 -0700 Subject: ASoC: Document snd-soc-dummy-dai purpose Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 362c69ac1d6c..53dd085d3ee2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec; SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) +/* + * The dummy CODEC is only meant to be used in situations where there is no + * actual hardware. + * + * If there is actual hardware even if it does not have a control bus + * the hardware will still have constraints like supported samplerates, etc. + * which should be modelled. And the data flow graph also should be modelled + * using DAPM. + */ static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", .playback = { -- cgit v1.2.3-58-ga151 From f1ec5ec7a94ba8138f9cbdc1e9e3b03aee29c4df Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 9 Sep 2015 12:10:31 +0100 Subject: ASoC: intel: Fix SSP port configuration after RTD3 resume. Currently the SSP port settings are being clobbered as part of the DSP RTD3 restore logic. make sure we save the correct params and restore them at resume. The FW sadly does not save SSP settings as part of the PM context. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f95f271aab0c..5bc98550c81d 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -302,6 +302,10 @@ struct sst_hsw { struct sst_hsw_ipc_dx_reply dx; void *dx_context; dma_addr_t dx_context_paddr; + enum sst_hsw_device_id dx_dev; + enum sst_hsw_device_mclk dx_mclk; + enum sst_hsw_device_mode dx_mode; + u32 dx_clock_divider; /* boot */ wait_queue_head_t boot_wait; @@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, trace_ipc_request("set device config", dev); - config.ssp_interface = dev; - config.clock_frequency = mclk; - config.mode = mode; - config.clock_divider = clock_divider; + hsw->dx_dev = config.ssp_interface = dev; + hsw->dx_mclk = config.clock_frequency = mclk; + hsw->dx_mode = config.mode = mode; + hsw->dx_clock_divider = config.clock_divider = clock_divider; if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) config.channels = 4; else @@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) return -EIO; } - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); + /* Set ADSP SSP port settings - sadly the FW does not store SSP port + settings as part of the PM context. */ + ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk, + hsw->dx_mode, hsw->dx_clock_divider); if (ret < 0) dev_err(dev, "error: SSP re-initialization failed\n"); -- cgit v1.2.3-58-ga151 From 723831927e8813b5b336d383174f686ad708bf10 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 14 Sep 2015 16:06:48 +0300 Subject: ASoC: davinci-mcasp: Revise the FIFO threshold calculation The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value for the refining should meet this requirement as well. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index add6bb99661d..1260b315a96c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; - int active_serializers, numevt, n; + int active_serializers, numevt; u32 reg; /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) @@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * The number of words for numevt need to be in steps of active * serializers. */ - n = numevt % active_serializers; - if (n) - numevt += (active_serializers - n); + numevt = (numevt / active_serializers) * active_serializers; + while (period_words % numevt && numevt > 0) numevt -= active_serializers; if (numevt <= 0) -- cgit v1.2.3-58-ga151 From 295c3405a8bbd69ee8c8eb6580f30b0b8739b33a Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 9 Sep 2015 21:27:42 +0300 Subject: ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver The TX and RX direction share the same bit clock and frame sync, so the samplerate must be the same to both directions. Signed-off-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1260b315a96c..2c5a2c5444ab 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1298,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .ops = &davinci_mcasp_dai_ops, .symmetric_samplebits = 1, + .symmetric_rates = 1, }, { .name = "davinci-mcasp.1", -- cgit v1.2.3-58-ga151 From ab1fffe3a73c694d698645451ba61255ec4ba5e6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 18 Sep 2015 15:02:50 +0300 Subject: ASoC: davinci-mcasp: Fix devm_kasprintf format string The '\n' at the end of the format string is not needed. It adds an extra line break when doing cat /proc/interrupts Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 2c5a2c5444ab..7d45d98a861f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "common"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, @@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, @@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "tx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, -- cgit v1.2.3-58-ga151 From 5b64c173cdea21105eb4794487b3d593f0a2e6c3 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Wed, 16 Sep 2015 10:13:19 +0100 Subject: ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8ec6fb208ea0..37c5cd4d0e59 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- cgit v1.2.3-58-ga151