February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] regulator: wm8350: Don't specify consumer supplies with struct device
by Mark Brown 02 Feb '12

02 Feb '12

Very, very deprecated. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/wm8350-regulator.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/wm8350-regulator.c b/drivers/regulator/wm8350-regulator.c index 6894009..4f46067 100644 --- a/drivers/regulator/wm8350-regulator.c +++ b/drivers/regulator/wm8350-regulator.c @@ -1544,7 +1544,7 @@ int wm8350_register_led(struct wm8350 *wm8350, int lednum, int dcdc, int isink, return -ENOMEM; } - led->isink_consumer.dev = &pdev->dev; + led->isink_consumer.dev_name = dev_name(&pdev->dev); led->isink_consumer.supply = "led_isink"; led->isink_init.num_consumer_supplies = 1; led->isink_init.consumer_supplies = &led->isink_consumer; @@ -1559,7 +1559,7 @@ int wm8350_register_led(struct wm8350 *wm8350, int lednum, int dcdc, int isink, return ret; } - led->dcdc_consumer.dev = &pdev->dev; + led->dcdc_consumer.dev_name = dev_name(&pdev->dev); led->dcdc_consumer.supply = "led_vcc"; led->dcdc_init.num_consumer_supplies = 1; led->dcdc_init.consumer_supplies = &led->dcdc_consumer; -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH] ASoC: pcm: Improve error logging
by Mark Brown 02 Feb '12

02 Feb '12

Use the standard logging macros and use dev_ variants where we can, also reporting error codes whenever we report an error. These changes (the error codes in particular) make it noticeably easier to figure out what went wrong just from the basic dmesg output. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 39 ++++++++++++++++++++++----------------- 1 files changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fabd33e..ef28dd5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -123,8 +123,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "can't open interface %s: %d\n", + cpu_dai->name, ret); goto out; } } @@ -132,7 +132,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->open) { ret = platform->driver->ops->open(substream); if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + dev_err(platform->dev, "can't open platform %s: %d\n", + platform->name, ret); goto platform_err; } } @@ -140,8 +141,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't open codec %s: %d\n", + codec_dai->name, ret); goto codec_dai_err; } } @@ -149,7 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + pr_err("asoc: %s startup failed: %d\n", + rtd->dai_link->name, ret); goto machine_err; } } @@ -413,7 +415,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); + pr_err("asoc: machine prepare error: %d\n", ret); goto out; } } @@ -421,7 +423,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->prepare) { ret = platform->driver->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); + dev_err(platform->dev, "platform prepare error: %d\n", + ret); goto out; } } @@ -429,7 +432,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); + dev_err(codec_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -437,7 +441,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + dev_err(cpu_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -484,7 +489,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); + pr_err("asoc: machine hw_params failed: %d\n", ret); goto out; } } @@ -492,8 +497,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (codec_dai->driver->ops->hw_params) { ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't set %s hw params: %d\n", + codec_dai->name, ret); goto codec_err; } } @@ -501,8 +506,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->driver->ops->hw_params) { ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "%s hw params failed: %d\n", + cpu_dai->name, ret); goto interface_err; } } @@ -510,8 +515,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); + dev_err(platform->dev, "%s hw params failed: %d\n", + platform->name, ret); goto platform_err; } } -- 1.7.9.rc1

2 1

[alsa-devel] Some questions related to ALSA based duplex audio processing
by public-hk 02 Feb '12

02 Feb '12

Hi, I have just started to develop a duplex audio processing application based on ALSA. My development goals are - of course - maximum stability as well as lowest possible delay (latency). So, here is what I did on my Ubuntu laptop: 1) Setup of device/soundcard "hw:0,0" for processing at a samplerate of 44.1 kHz, buffersize 128 samples. 2) I created one input PCM device and one output PCM device handle 3) I use the function "snd_async_add_pcm_handler" to install a callback function (ASYNC mode), one for input, one for output. 4) I use "snd_pcm_link" to synchronize both pcm handles. 5) I use the mmap'ed area access My motivation to work the way I do is to maximize the control of processing behavior and to minimize the latency. E.g., I have understood from the docs that the snd_pcm_read/snd_pcm_write functions do nothing else than addressing the memory mapped areas. As a conclusion, I realize this access myself to have more control about it. And by using the async callbacks, I do not have to deal with blocking or non-blocking read functions or polling related issues. So far, I have realized the playback part and noticed some aspects which are not clear to me from the ALSA documentation: a) What is the most efficient way to realize duplex audio processing, the async way with mmap'ed read/write that I follow? b) At first, I created the pcm handles as follows: "snd_pcm_open (.., SND_PCM_ASYNC);" When starting to process audio (call of "snd_pcm_start"), I realized that the registered callback functions do not get called. I had to change this to "snd_pcm_open (.., 0);". Is that really intended, it seems contradictory? c) The function "snd_pcm_info_get_sync" is supposed to return a description of the synchronization behavior of a soundcard. In my case, I called this function for two soundcards (one USB and the laptop integrated soundcard). In both cases, the returned content is all zeros. Should not this be different for both devices? d) By default, it seems that the callbacks for audio frames come within a thread with normal priority. On Windows OS, I am used to increase the thread priority whenever starting my threads, but it is commonly not a good idea to increase the process priority. In the ALSA latency-example, the PROCESS priority is increased (which can be done only with superuser priv.). What is the recommended way in Linux to achieve lower latencies? In the future I will have to deal with synchronization of input and output since the samples arrive/leave my application in two independent callback functions. So what will I do: Once input samples are available I will store these in one buffer, and these samples will be output the next time when there is space in the output ring buffer. In order to minimize the latency, I would need to know more about the intended exact behavior of the ALSA lib which I did not find in the documentation: I) If linking input and output with "snd_pcm_link", I have understood that the changes of state for input and output PCM handle will occur synchronously. That is, when doing the operation such as "snd_pcm_prepare" on one of the handles, both handles will be affected. However, what does this means for audio processing on the frame level: If I use two callbacks for signal processing for input and output respectively installed based on the function "snd_async_add_pcm_handler", will these callbacks occur simultaneously? On Windows OS (speaking of ASIO sound) there is one callback in which input and output is handled simultaneously. Can I somehow setup ALSA to have a similar behavor? Thank you for any assistance and best regards HK

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Re: [alsa-devel] [PATCH v4 1/5] OMAP4: hwmod: Add names for DMIC memory address space
by Peter Ujfalusi 02 Feb '12

02 Feb '12

Hi Paul, On 01/25/2012 11:23 PM, Paul Walmsley wrote: > Okay, thanks, queued for 3.3-rc. It seams this patch did not made it to 3.3-rc2. We still have broken audio out of box on SDP4430/Blaze. Can you make sure that this will be queued for rc3? Thank you, Péter

2 2

[alsa-devel] [PATCH] ASoC: wm_hubs: Correct line input to line output 2 paths
by Mark Brown 02 Feb '12

02 Feb '12

The second line output mixer has the controls for the line input bypasses in the opposite order. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 57514dd2..b08d04c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: Use dai_fmt in afeb9260 machine driver
by Axel Lin 01 Feb '12

01 Feb '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/snd-soc-afeb9260.c | 23 ++--------------------- 1 files changed, 2 insertions(+), 21 deletions(-) diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 4ca667d..cb0130c 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -46,29 +46,8 @@ static int afeb9260_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; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S| - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return err; - } - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return err; - } - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -129,6 +108,8 @@ static struct snd_soc_dai_link afeb9260_dai = { .platform_name = "atmel_pcm-audio", .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &afeb9260_ops, }; -- 1.7.5.4

2 2

[alsa-devel] [PATCH 2/3] ARM: ux500: Add DMA-channels for MSP
by Ola Lilja 01 Feb '12

01 Feb '12

Add DMA-channels for MSP0, MSP1, MSP2 and MSP3. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/devices-db8500.c | 6 ++++++ arch/arm/mach-ux500/include/mach/hardware.h | 2 ++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ux500/devices-db8500.c b/arch/arm/mach-ux500/devices-db8500.c index a7c6cdc..6e66d37 100644 --- a/arch/arm/mach-ux500/devices-db8500.c +++ b/arch/arm/mach-ux500/devices-db8500.c @@ -101,6 +101,9 @@ static const dma_addr_t dma40_tx_map[DB8500_DMA_NR_DEV] = { [DB8500_DMA_DEV41_SD_MM3_TX] = -1, [DB8500_DMA_DEV42_SD_MM4_TX] = -1, [DB8500_DMA_DEV43_SD_MM5_TX] = -1, + [DB8500_DMA_DEV14_MSP2_TX] = U8500_MSP2_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV30_MSP1_TX] = U8500_MSP1_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV31_MSP0_TX_SLIM0_CH0_TX] = U8500_MSP0_BASE + MSP_TX_RX_REG_OFFSET, }; /* Mapping between source event lines and physical device address */ @@ -133,6 +136,9 @@ static const dma_addr_t dma40_rx_map[DB8500_DMA_NR_DEV] = { [DB8500_DMA_DEV41_SD_MM3_RX] = -1, [DB8500_DMA_DEV42_SD_MM4_RX] = -1, [DB8500_DMA_DEV43_SD_MM5_RX] = -1, + [DB8500_DMA_DEV14_MSP2_RX] = U8500_MSP2_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV30_MSP3_RX] = U8500_MSP3_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV31_MSP0_RX_SLIM0_CH0_RX] = U8500_MSP0_BASE + MSP_TX_RX_REG_OFFSET, }; /* Reserved event lines for memcpy only */ diff --git a/arch/arm/mach-ux500/include/mach/hardware.h b/arch/arm/mach-ux500/include/mach/hardware.h index b6ba26a..d93d6db 100644 --- a/arch/arm/mach-ux500/include/mach/hardware.h +++ b/arch/arm/mach-ux500/include/mach/hardware.h @@ -30,6 +30,8 @@ #include <mach/db8500-regs.h> #include <mach/db5500-regs.h> +#define MSP_TX_RX_REG_OFFSET 0 + #ifndef __ASSEMBLY__ #include <mach/id.h> -- 1.7.8.3

2 1

[alsa-devel] [PATCH 1/3] ARM: ux500: Add placeholder for clk_set_parent
by Ola Lilja 01 Feb '12

01 Feb '12

Calling clk_set_parent (e.g. from Ux500 ASoC-driver) generates build-errors. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/clock.c | 7 +++++++ arch/arm/mach-ux500/clock.h | 1 + 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ux500/clock.c b/arch/arm/mach-ux500/clock.c index 7379075..ec35f0a 100644 --- a/arch/arm/mach-ux500/clock.c +++ b/arch/arm/mach-ux500/clock.c @@ -223,6 +223,13 @@ int clk_set_rate(struct clk *clk, unsigned long rate) } EXPORT_SYMBOL(clk_set_rate); +int clk_set_parent(struct clk *clk, struct clk *parent) +{ + /*TODO*/ + return -ENOSYS; +} +EXPORT_SYMBOL(clk_set_parent); + static void clk_prcmu_enable(struct clk *clk) { void __iomem *cg_set_reg = __io_address(U8500_PRCMU_BASE) diff --git a/arch/arm/mach-ux500/clock.h b/arch/arm/mach-ux500/clock.h index 0744907..d776ada 100644 --- a/arch/arm/mach-ux500/clock.h +++ b/arch/arm/mach-ux500/clock.h @@ -21,6 +21,7 @@ struct clkops { void (*enable) (struct clk *); void (*disable) (struct clk *); unsigned long (*get_rate) (struct clk *); + int (*set_parent)(struct clk *, struct clk *); }; /** -- 1.7.8.3

2 1

[alsa-devel] [PATCH] ALSA: E-mu 1616m PCIe support
by Sergey Korolev 01 Feb '12

01 Feb '12

diff -urdN linux-2.6.36.orig/sound/pci/emu10k1/emu10k1_main.c linux-2.6.36/sound/pci/emu10k1/emu10k1_main.c --- linux-2.6.36.orig/sound/pci/emu10k1/emu10k1_main.c 2010-10-20 22:30:22.000000000 +0200 +++ linux-2.6.36/sound/pci/emu10k1/emu10k1_main.c 2011-01-21 17:48:11.440000537 +0100 @@ -1415,6 +1415,14 @@ .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ + /* Tested 21th Jan 2011. */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, + .driver = "Audigy2", .name = "E-mu 1010 PCIe", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James(a)superbug.co.uk 8th July 2005. */ /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,

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[alsa-devel] [PATCH] ASoC: blackfin: Use dai_fmt
by Lars-Peter Clausen 01 Feb '12

01 Feb '12

Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing this manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ad1836.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad193x.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad73311.c | 29 ++++------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 20 ++++---------------- sound/soc/blackfin/bfin-eval-adau1373.c | 13 ++----------- sound/soc/blackfin/bfin-eval-adau1701.c | 16 +++++----------- sound/soc/blackfin/bfin-eval-adav80x.c | 13 ++----------- 7 files changed, 27 insertions(+), 98 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 60962ce..d542d40 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -40,20 +40,8 @@ static int bf5xx_ad1836_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; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | 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_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; /* set cpu DAI channel mapping */ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), @@ -68,6 +56,9 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { .hw_params = bf5xx_ad1836_hw_params, }; +#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { { .name = "ad1836", @@ -77,6 +68,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, { .name = "ad1836", @@ -86,6 +78,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 2d8d82d..0e55e9f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,18 +60,6 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, break; } - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | 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_DSP_A | - SND_SOC_DAIFMT_IB_IF | 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, 0, clk, SND_SOC_CLOCK_IN); @@ -92,6 +80,9 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, return 0; } +#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_ops bf5xx_ad193x_ops = { .hw_params = bf5xx_ad193x_hw_params, }; @@ -105,6 +96,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, { .name = "ad193x", @@ -114,6 +106,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 8e49508..61cc91d 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -145,29 +145,8 @@ static int bf5xx_probe(struct snd_soc_card *card) return 0; } -static int bf5xx_ad73311_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; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - - -static struct snd_soc_ops bf5xx_ad73311_ops = { - .hw_params = bf5xx_ad73311_hw_params, -}; +#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { { @@ -177,7 +156,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, { .name = "ad73311", @@ -186,7 +165,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 0303032..df3ac73 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -49,7 +49,6 @@ static int bf5xx_ssm2602_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 clk = 0; int ret = 0; @@ -75,21 +74,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, break; } - /* - * CODEC is master for BCLK and LRC in this configuration. - */ - - /* 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; - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -102,6 +86,10 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = { .hw_params = bf5xx_ssm2602_hw_params, }; +/* CODEC is master for BCLK and LRC in this configuration. */ +#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { { .name = "ssm2602", diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 26b271c..f3adbdb 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -67,21 +67,10 @@ static int bfin_eval_adau1373_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; int ret; int pll_rate; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - 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) - return ret; - switch (params_rate(params)) { case 48000: case 8000: @@ -143,6 +132,8 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = { .codec_name = "adau1373.0-001a", .ops = &bfin_eval_adau1373_ops, .init = bfin_eval_adau1373_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card bfin_eval_adau1373 = { diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index c0064fa..b0531fc 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -37,20 +37,9 @@ static int bfin_eval_adau1701_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; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - 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) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, SND_SOC_CLOCK_IN); @@ -61,6 +50,9 @@ static struct snd_soc_ops bfin_eval_adau1701_ops = { .hw_params = bfin_eval_adau1701_hw_params, }; +#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { { .name = "adau1701", @@ -70,6 +62,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, { .name = "adau1701", @@ -79,6 +72,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 4ef079f..84b0998 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -34,20 +34,9 @@ static int bfin_eval_adav80x_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; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - 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) - return ret; - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, 27000000, params_rate(params) * 256); if (ret) @@ -88,6 +77,8 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { .platform_name = "bfin-i2s-pcm-audio", .init = bfin_eval_adav80x_codec_init, .ops = &bfin_eval_adav80x_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }, }; -- 1.7.8.3

2 1