From 5a1b8a80daa15e5b1920735133b80dd46975765f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 30 Dec 2014 16:10:32 +0200 Subject: ASoC: davinci-mcasp: Support for combined tx/rx interrupt line Some SoC, like da850/OMAP-L138 uses one common interrupt request for TX/RX events. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4f9c5d..95eef582331b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) return IRQ_RETVAL(handled_mask); } +static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + irqreturn_t ret = IRQ_NONE; + + if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]) + ret = davinci_mcasp_tx_irq_handler(irq, data); + + if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]) + ret |= davinci_mcasp_rx_irq_handler(irq, data); + + return ret; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -1441,6 +1455,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "common"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_common_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "common IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", -- cgit From cd2ee8ad55a1728f291a760cece9ceac11462b9e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 5 Jan 2015 10:18:25 +0100 Subject: ASoC: cs42l73: Constify struct regmap_config and snd_soc_codec_driver The regmap_config struct may be const because it is not modified by the driver and regmap_init() accepts pointer to const. Make struct snd_soc_codec_driver const as well (snd_soc_register_codec() accepts pointer to const). Signed-off-by: Krzysztof Kozlowski Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7c55537c69cf..8ecedba79606 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1347,7 +1347,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .probe = cs42l73_probe, .set_bias_level = cs42l73_set_bias_level, .suspend_bias_off = true, @@ -1361,7 +1361,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .num_controls = ARRAY_SIZE(cs42l73_snd_controls), }; -static struct regmap_config cs42l73_regmap = { +static const struct regmap_config cs42l73_regmap = { .reg_bits = 8, .val_bits = 8, -- cgit From 059d3dc6e5e8d4a62803eb3ad01b34fa3f714675 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:10 +0100 Subject: ASoC: sam9g20_wm8731: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 30 ++---------------------------- 1 file changed, 2 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 66b66d0e7514..98ca6341147b 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -64,33 +64,6 @@ static struct clk *mclk; -static int at91sam9g20ek_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; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops at91sam9g20ek_ops = { - .hw_params = at91sam9g20ek_hw_params, -}; - static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -173,7 +146,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .init = at91sam9g20ek_wm8731_init, .platform_name = "at91rm9200_ssc.0", .codec_name = "wm8731.0-001b", - .ops = &at91sam9g20ek_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card snd_soc_at91sam9g20ek = { -- cgit From b4508d0f95fa4aaed889549e31391641d675d4bb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:11 +0100 Subject: ASoC: db1200: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/au1x/db1200.c | 19 +++---------------- 1 file changed, 3 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a747ac0b399f..c75995f2779c 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -91,27 +91,12 @@ static int db1200_i2s_startup(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; - int ret; /* WM8731 has its own 12MHz crystal */ snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, 12000000, SND_SOC_CLOCK_IN); - /* codec is bitclock and lrclk master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - goto out; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - goto out; - - ret = 0; -out: - return ret; + return 0; } static struct snd_soc_ops db1200_i2s_wm8731_ops = { @@ -125,6 +110,8 @@ static struct snd_soc_dai_link db1200_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.1", .platform_name = "au1xpsc-pcm.1", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; -- cgit From 4de59dbd28565ad1074d2bcf294a0a4b41c64e07 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:12 +0100 Subject: ASoC: eureka-tlv320: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9ce70fc67b09..8c9e9006dd84 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,25 +42,6 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - /* fsl_ssi lacks the set_fmt ops. */ - if (ret && ret != -ENOTSUPP) { - dev_err(cpu_dai->dev, - "Failed to set the cpu dai format.\n"); - return ret; - } - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - dev_err(cpu_dai->dev, - "Failed to set the codec format.\n"); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { @@ -91,6 +72,8 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &eukrea_tlv320_snd_ops, }; -- cgit From dea53bd36811d6b117a497eb91b2e4a54ef73548 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:13 +0100 Subject: ASoC: mx27vis-aci32x4: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/mx27vis-aic32x4.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index b1ced7b8d80c..198eeb3f3f7a 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -55,16 +55,6 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - u32 dai_format; - - dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 25000000, SND_SOC_CLOCK_OUT); @@ -164,6 +154,8 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .platform_name = "imx-ssi.0", .codec_name = "tlv320aic32x4.0-0018", .cpu_dai_name = "imx-ssi.0", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &mx27vis_aic32x4_snd_ops, }; -- cgit From 734890a6c99ee81b935164963c03fc8bbf3f821b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:14 +0100 Subject: ASoC: wm1133-ev1: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 804749a6c61e..d072bd13db09 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -87,7 +87,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int rate = params_rate(params); unsigned int channels = params_channels(params); - u32 dai_format; /* find the correct audio parameters */ for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { @@ -104,15 +103,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, /* codec FLL input is 14.75 MHz from MCLK */ snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); - dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); - /* TODO: The SSI driver should figure this out for us */ switch (channels) { case 2: @@ -244,6 +234,8 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, .symmetric_rates = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card wm1133_ev1 = { -- cgit From bc6a5d649bda56c579853bfe710db88bf38da522 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:15 +0100 Subject: ASoC: mxs-sgtl5000: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 27 ++++++--------------------- 1 file changed, 6 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 6f1916b71815..6e6fce6a14ba 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -36,7 +36,7 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate = params_rate(params); - u32 dai_format, mclk; + u32 mclk; int ret; /* sgtl5000 does not support 512*rate when in 96000 fs */ @@ -65,26 +65,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, return ret; } - /* set codec to slave mode */ - dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, dai_format); - if (ret) { - dev_err(codec_dai->dev, "Failed to set dai format to %08x\n", - dai_format); - return ret; - } - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); - if (ret) { - dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n", - dai_format); - return ret; - } - return 0; } @@ -92,17 +72,22 @@ static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { .hw_params = mxs_sgtl5000_hw_params, }; +#define MXS_SGTL5000_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS) + static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { { .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", + .dai_fmt = MXS_SGTL5000_DAI_FMT, .ops = &mxs_sgtl5000_hifi_ops, .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", + .dai_fmt = MXS_SGTL5000_DAI_FMT, .ops = &mxs_sgtl5000_hifi_ops, .capture_only = true, }, -- cgit From 3aa273e3f65455066aa7338845e11910ba2bae12 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:16 +0100 Subject: ASoC: ams-delta: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 18 +++--------------- 1 file changed, 3 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 4c6afb75eea6..706613077c15 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -412,21 +412,7 @@ static struct tty_ldisc_ops cx81801_ops = { * over the modem port. */ -static int ams_delta_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* Set cpu DAI configuration */ - return snd_soc_dai_set_fmt(rtd->cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -} - -static struct snd_soc_ops ams_delta_ops = { - .hw_params = ams_delta_hw_params, -}; +static struct snd_soc_ops ams_delta_ops; /* Digital mute implemented using modem/CPU multiplexer. @@ -546,6 +532,8 @@ static struct snd_soc_dai_link ams_delta_dai_link = { .platform_name = "omap-mcbsp.1", .codec_name = "cx20442-codec", .ops = &ams_delta_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; /* Audio card driver */ -- cgit From a8bd0ee5587148cd1a23302ef37bf3f236fe6705 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:17 +0100 Subject: ASoC: raumfeld: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/raumfeld.c | 35 ++++++++--------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 083706595495..552b763005ed 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -88,7 +88,7 @@ static int raumfeld_cs4270_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 fmt, clk = 0; + unsigned int clk = 0; int ret = 0; switch (params_rate(params)) { @@ -112,15 +112,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - /* setup the CODEC DAI */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); if (ret < 0) return ret; @@ -130,10 +121,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); if (ret < 0) return ret; @@ -169,9 +156,8 @@ static int raumfeld_ak4104_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; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int fmt, ret = 0, clk = 0; + int ret = 0, clk = 0; switch (params_rate(params)) { case 44100: @@ -194,22 +180,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; - - /* setup the CODEC DAI */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* setup the CPU DAI */ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); if (ret < 0) return ret; @@ -233,6 +208,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ .codec_name = "cs4270.0-0048", \ + .dai_fmt = SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS, \ .ops = &raumfeld_cs4270_ops, \ } @@ -243,6 +221,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .cpu_dai_name = "pxa-ssp-dai.1", \ .codec_dai_name = "ak4104-hifi", \ .platform_name = "pxa-pcm-audio", \ + .dai_fmt = SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS, \ .ops = &raumfeld_ak4104_ops, \ .codec_name = "spi0.0", \ } -- cgit From 132c30969b9346a628f597804d2343b6df493693 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:18 +0100 Subject: ASoC: zylonite: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 23bf991e95d5..8f301c72ee5e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -130,16 +130,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - return 0; } @@ -172,6 +162,8 @@ static struct snd_soc_dai_link zylonite_dai[] = { .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", .codec_dai_name = "wm9713-voice", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &zylonite_voice_ops, }, }; -- cgit From 0af76f918adb8487a911bc136a6f1b12c46cd056 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:19 +0100 Subject: ASoC: goni_wm8994: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/goni_wm8994.c | 23 ++++------------------- 1 file changed, 4 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 3b527dcfc0aa..fad56b9e7369 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -136,22 +136,9 @@ static int goni_hifi_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 = 24000000; int ret = 0; - /* set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, params_rate(params) * 256); @@ -182,12 +169,6 @@ static int goni_voice_hw_params(struct snd_pcm_substream *substream, if (params_rate(params) != 8000) return -EINVAL; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, params_rate(params) * 256); @@ -234,6 +215,8 @@ static struct snd_soc_dai_link goni_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-i2s.0", .codec_name = "wm8994-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = goni_wm8994_init, .ops = &goni_hifi_ops, }, { @@ -242,6 +225,8 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-aif2", .codec_name = "wm8994-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &goni_voice_ops, }, }; -- cgit From 5cc10b9b77c234e89e7c4aca56d8a3c571111955 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:20 +0100 Subject: ASoC: h1940_uda1380: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index f2d7980d7ddc..59b044255b78 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -76,7 +76,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int div; int ret; unsigned int rate = params_rate(params); @@ -95,18 +94,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* select clock source */ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, SND_SOC_CLOCK_OUT); @@ -207,6 +194,8 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { .init = h1940_uda1380_init, .platform_name = "s3c24xx-iis", .codec_name = "uda1380-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &h1940_ops, }, }; -- cgit From 72303cafcb5cc9fd1227d0edd458f62bef1a397e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:21 +0100 Subject: ASoC: jive_wm8750: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/jive_wm8750.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index b5f6abd9d221..6c3b359bb4c1 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -61,20 +61,6 @@ static int jive_hw_params(struct snd_pcm_substream *substream, s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), s3c_i2sv2_get_clock(cpu_dai)); - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); @@ -121,6 +107,8 @@ static struct snd_soc_dai_link jive_dai = { .platform_name = "s3c2412-i2s", .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &jive_ops, }; -- cgit From e16514d931635640906ab2810c98f6a9047cb87a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:22 +0100 Subject: ASoC: neo1973_wm8753: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 9b4a09f14b6c..65602b935377 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -70,20 +70,6 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); @@ -151,13 +137,6 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); @@ -300,6 +279,8 @@ static struct snd_soc_dai_link neo1973_dai[] = { .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .codec_name = "wm8753.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -309,6 +290,8 @@ static struct snd_soc_dai_link neo1973_dai[] = { .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &neo1973_voice_ops, }, }; -- cgit From 5f0acedddf533c086edf29eefdc743dc1e756b5d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:23 +0100 Subject: ASoC: rx1950_uda1380: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 37688ebbb2b4..873f2cb4bebe 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -89,6 +89,8 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = { .init = rx1950_uda1380_init, .platform_name = "s3c24xx-iis", .codec_name = "uda1380-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &rx1950_ops, }, }; @@ -154,7 +156,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int div; int ret; unsigned int rate = params_rate(params); @@ -181,18 +182,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* select clock source */ ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source, rate, SND_SOC_CLOCK_OUT); -- cgit From d27e51dade482acdf67021c12d692041a35bea77 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:24 +0100 Subject: ASoC: s3c24xx_simtec: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_simtec.c | 20 ++------------------ 1 file changed, 2 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 2c015f62ead6..dcc008d1e1ab 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -169,24 +169,6 @@ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set the CODEC as the bus clock master, I2S */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - /* Set the CODEC as the bus clock master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { @@ -320,6 +302,8 @@ int simtec_audio_core_probe(struct platform_device *pdev, int ret; card->dai_link->ops = &simtec_snd_ops; + card->dai_link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; pdata = pdev->dev.platform_data; if (!pdata) { -- cgit From 517b9a2a90676aeeefc7e2b8508fe4260e02c432 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:25 +0100 Subject: ASoC: s3c24xx_uda134x: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_uda134x.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 9c6f7db56f60..50849e137fc0 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -173,16 +173,6 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -223,6 +213,8 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &s3c24xx_uda134x_ops, .platform_name = "s3c24xx-iis", }; -- cgit From 10756c2770d7f4758e36d2a9e37ff2637001d610 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:26 +0100 Subject: ASoC: smartq_wm8987: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smartq_wm8987.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 9b0ffacab790..8291d2a5f152 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -56,20 +56,6 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* Use PCLK for I2S signal generation */ ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, 0, SND_SOC_CLOCK_IN); @@ -199,6 +185,8 @@ static struct snd_soc_dai_link smartq_dai[] = { .platform_name = "samsung-i2s.0", .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &smartq_hifi_ops, }, }; -- cgit From 7a57ca2362a55c34b96d90539dd8b7cb6c1c09a1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:27 +0100 Subject: ASoC: smdk_wm8580: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8580.c | 21 ++++++--------------- 1 file changed, 6 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index b1a519f83b29..17a2f717ec02 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -32,7 +32,6 @@ static int smdk_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; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int pll_out; int bfs, rfs, ret; @@ -77,20 +76,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, } pll_out = params_rate(params) * rfs; - /* Set the Codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* Set the AP DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* Set WM8580 to drive MCLK from its PLLA */ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, WM8580_CLKSRC_PLLA); @@ -168,6 +153,9 @@ enum { SEC_PLAYBACK, }; +#define SMDK_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link smdk_dai[] = { [PRI_PLAYBACK] = { /* Primary Playback i/f */ .name = "WM8580 PAIF RX", @@ -176,6 +164,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-i2s.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_ops, }, [PRI_CAPTURE] = { /* Primary Capture i/f */ @@ -185,6 +174,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-i2s.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -195,6 +185,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-i2s-sec", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_ops, }, }; -- cgit From 6a5794f2c653e78127123b3cd266b3c4d4c50606 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:28 +0100 Subject: ASoC: smdk_wm8570pcm: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8580pcm.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 05c609c62de9..6deec5234c92 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -62,20 +62,6 @@ static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream, rfs = mclk_freq / params_rate(params) / 2; - /* Set the codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* Set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - if (mclk_freq == xtal_freq) { ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, mclk_freq, SND_SOC_CLOCK_IN); @@ -121,6 +107,9 @@ static struct snd_soc_ops smdk_wm8580_pcm_ops = { .hw_params = smdk_wm8580_pcm_hw_params, }; +#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \ + SND_SOC_DAIFMT_CBS_CFS) + static struct snd_soc_dai_link smdk_dai[] = { { .name = "WM8580 PAIF PCM RX", @@ -129,6 +118,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -137,6 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-pcm.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_wm8580_pcm_ops, }, }; -- cgit From fce20bf879dd5ce309c593c91bbb7b5e6ffd22e7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:29 +0100 Subject: ASoC: smdk_wm8994pcm: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994pcm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index c470e8eed6e1..b1c89ec2d999 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -68,20 +68,6 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, mclk_freq = params_rate(params) * rfs; - /* Set the codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* Set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, mclk_freq, SND_SOC_CLOCK_IN); if (ret < 0) @@ -118,6 +104,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-pcm.0", .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &smdk_wm8994_pcm_ops, }, }; -- cgit From 9cb2a25ce68aec15d5d940f5d3040bd66ca688c7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Jan 2015 17:16:30 +0100 Subject: ASoC: migor: Use static DAI format setup Set the dai_fmt field in the dai_link struct instead of manually calling snd_soc_dai_fmt(). This makes the code cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index c58c2529f103..82f582344fe7 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -63,16 +63,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - codec_freq = rate * 512; /* * This propagates the parent frequency change to children and @@ -144,6 +134,8 @@ static struct snd_soc_dai_link migor_dai = { .codec_dai_name = "wm8978-hifi", .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", + .dai_fmt = SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFS, .ops = &migor_dai_ops, }; -- cgit From ce64c8b9cf5be2a93508af4667110dbe90904557 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Jan 2015 15:17:20 +0100 Subject: ASoC: Add helper function for changing the DAI link format For some setups it is necessary to change the DAI link format at runtime. This patch factors out the code that does the initial static DAI link format configuration into a separate helper function which can be used board drivers as well. This allows board drivers that have to change the DAI link format at runtime to reuse it instead of having to manually change the format on all DAIs. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 125 ++++++++++++++++++++++++++++----------------------- 1 file changed, 69 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 985052b3fbed..80ea358d3ad7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1427,11 +1427,75 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) return 0; } +/** + * snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime + * @rtd: The runtime for which the DAI link format should be changed + * @dai_fmt: The new DAI link format + * + * This function updates the DAI link format for all DAIs connected to the DAI + * link for the specified runtime. + * + * Note: For setups with a static format set the dai_fmt field in the + * corresponding snd_dai_link struct instead of using this function. + * + * Returns 0 on success, otherwise a negative error code. + */ +int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, + unsigned int dai_fmt) +{ + struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int i; + int ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = codec_dais[i]; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(codec_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + } + + /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ + if (cpu_dai->codec) { + unsigned int inv_dai_fmt; + + inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; + } + + dai_fmt = inv_dai_fmt; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(cpu_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; - struct snd_soc_dai_link *dai_link; - int ret, i, order, dai_fmt; + int ret, i, order; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); @@ -1542,60 +1606,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); for (i = 0; i < card->num_links; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - dai_link = &card->dai_link[i]; - dai_fmt = dai_link->dai_fmt; - - if (dai_fmt) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = codec_dais[j]; - - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(codec_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } - } - - /* If this is a regular CPU link there will be a platform */ - if (dai_fmt && - (dai_link->platform_name || dai_link->platform_of_node)) { - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } else if (dai_fmt) { - /* Flip the polarity for the "CPU" end */ - dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - switch (dai_link->dai_fmt & - SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case SND_SOC_DAIFMT_CBM_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case SND_SOC_DAIFMT_CBS_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } + if (card->dai_link[i].dai_fmt) + snd_soc_runtime_set_dai_fmt(&card->rtd[i], + card->dai_link[i].dai_fmt); } snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), -- cgit From dc1b36b24990b9d87c6de88819c112c4db91dbc8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Jan 2015 15:17:21 +0100 Subject: ASoC: omap-twl4030: Use snd_soc_runtime_set_dai_fmt() Use snd_soc_runtime_set_dai_fmt() to configure the format for the DAI link rather than configuring each DAI individually. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-twl4030.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 5e551c762b7a..fb1f6bb87cd4 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -53,11 +53,7 @@ static int omap_twl4030_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; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_card *card = rtd->card; unsigned int fmt; - int ret; switch (params_channels(params)) { case 2: /* Stereo I2S mode */ @@ -74,21 +70,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - dev_err(card->dev, "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - dev_err(card->dev, "can't set cpu DAI configuration\n"); - return ret; - } - - return 0; + return snd_soc_runtime_set_dai_fmt(rtd, fmt); } static struct snd_soc_ops omap_twl4030_ops = { -- cgit From c853679a51b9bb76a3e06e64a0b4c6276f391921 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Jan 2015 15:17:22 +0100 Subject: ASoC: mop500_ab8500: Use snd_soc_runtime_set_dai_fmt() Use snd_soc_runtime_set_dai_fmt() to configure the format for the DAI link rather than configuring each DAI individually. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index be4f1ac7cd5e..aa65370db82a 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -290,21 +290,9 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_GATED; } - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - dev_err(dev, - "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", - __func__, ret); - return ret; - } - - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - dev_err(dev, - "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", - __func__, ret); + ret = snd_soc_runtime_set_dai_fmt(rtd, fmt); + if (ret) return ret; - } /* Setup TDM-slots */ -- cgit From ddaca25aa4dade234163290f6a9e091e7f57b36a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 8 Jan 2015 12:23:05 +0100 Subject: ASoC: Export snd_soc_runtime_set_dai_fmt() Export snd_soc_runtime_set_dai_fmt() so it can be used in modules. Fixes: ce64c8b9cf5b ("ASoC: Add helper function for changing the DAI link format") Reported-by: kbuild test robot Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 80ea358d3ad7..d342ee2ce28b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1491,6 +1491,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); static int snd_soc_instantiate_card(struct snd_soc_card *card) { -- cgit From f9a7ba326938f03b9305af8d31c360fce10cd4df Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 28 Jan 2015 22:30:01 +0100 Subject: ASoC: davinci: fix DM365_EVM codec selection An earlier bug fix of mine made the SND_DM365_VOICE_CODEC symbol tristate to avoid creating an undefined reference from the davinci-vcif.c driver to the davinci_soc_platform_register function that may be in a module. However, this may now lead to a different error on randconfig kernels: "warning: SND_DM365_VOICE_CODEC creates inconsistent choice state" This happens because we now have a choice statement with one bool and one tristate option, and the latter might not support being set to 'y' because of dependencies. This new change turns the other option into 'tristate' as well, which avoids the problem. Signed-off-by: Arnd Bergmann Fixes: 19926c6de0c3 ("ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is") Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8e948c63f3d9..2b81ca418d2a 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -58,13 +58,12 @@ choice depends on MACH_DAVINCI_DM365_EVM config SND_DM365_AIC3X_CODEC - bool "Audio Codec - AIC3101" + tristate "Audio Codec - AIC3101" help Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC tristate "Voice Codec - CQ93VC" - depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC -- cgit From 2f91ab86ec2500590c9dcfcc7f089d3255d72404 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 2 Feb 2015 14:49:41 +0200 Subject: ASoC: davinci-evm: Do not include edma headers The machine driver has no business with the underlying dma. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 158cb3d1db70..d1a8b6e3ac3b 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include @@ -25,8 +24,6 @@ #include #include -#include - #include "davinci-pcm.h" #include "davinci-i2s.h" -- cgit From 67d35e5bc375b03ed3ad88b7dc4469ed5d3d69e9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 2 Feb 2015 14:49:42 +0200 Subject: ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h There is no need for them to be included. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index d1a8b6e3ac3b..b6bb5947a8a8 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -24,9 +24,6 @@ #include #include -#include "davinci-pcm.h" -#include "davinci-i2s.h" - struct snd_soc_card_drvdata_davinci { struct clk *mclk; unsigned sysclk; -- cgit From 8f511ffb6b82f6914779a47039cccf91e953aa1f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 2 Feb 2015 14:38:32 +0200 Subject: ASoC: davicni-mcasp: Mark the common irq line as shared On DA830 devices McASP0,1 and 2 shares a single combined interrupt request line. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95eef582331b..a5e4fe0359ce 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1461,7 +1461,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, - IRQF_ONESHOT, irq_name, mcasp); + IRQF_ONESHOT | IRQF_SHARED, + irq_name, mcasp); if (ret) { dev_err(&pdev->dev, "common IRQ request failed\n"); goto err; -- cgit From caa1d794cca96c16e4a2f81697f3d15a588c4b02 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 2 Feb 2015 14:38:33 +0200 Subject: ASoC: davinci-mcasp: Fix DIT only McASP instance support One of the McASP instances in DM646x line of DMSoC only supports DIT mode. This means that the given IP does not have support for rx and all the rx related resources are missing, like irq and DMA request. The driver should not fail if any or all of the RX resource is missing when the op_mode is set to DIT mode. Since RX is not possible in DIT mode, we can just ignore the rx resources when the McASP is used in DIT mode. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 72 +++++++++++++++++++++------------------ 1 file changed, 39 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a5e4fe0359ce..de3b155a5011 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1327,16 +1327,19 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( pdata->tx_dma_channel = dma_spec.args[0]; - ret = of_property_match_string(np, "dma-names", "rx"); - if (ret < 0) - goto nodata; + /* RX is not valid in DIT mode */ + if (pdata->op_mode != DAVINCI_MCASP_DIT_MODE) { + ret = of_property_match_string(np, "dma-names", "rx"); + if (ret < 0) + goto nodata; - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, - &dma_spec); - if (ret < 0) - goto nodata; + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; - pdata->rx_dma_channel = dma_spec.args[0]; + pdata->rx_dma_channel = dma_spec.args[0]; + } ret = of_property_read_u32(np, "tx-num-evt", &val); if (ret >= 0) @@ -1532,19 +1535,34 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->filter_data = &dma_params->channel; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; - if (dat) - dma_params->dma_addr = dat->start; - else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + /* RX is not valid in DIT mode */ + if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_capture; + if (dat) + dma_params->dma_addr = dat->start; + else + dma_params->dma_addr = mem->start + pdata->rx_dma_offset; + + /* Unconditional dmaengine stuff */ + dma_data->addr = dma_params->dma_addr; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (res) + dma_params->channel = res->start; + else + dma_params->channel = pdata->rx_dma_channel; + + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "rx"; + else + dma_data->filter_data = &dma_params->channel; + } if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; @@ -1554,18 +1572,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res) - dma_params->channel = res->start; - else - dma_params->channel = pdata->rx_dma_channel; - - /* dmaengine filter data for DT and non-DT boot */ - if (pdev->dev.of_node) - dma_data->filter_data = "rx"; - else - dma_data->filter_data = &dma_params->channel; - dev_set_drvdata(&pdev->dev, mcasp); mcasp_reparent_fck(pdev); -- cgit