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

Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:32AM +0100, Lucas Stach wrote: > This connection is used by the AC97 controller. Applied, thanks.

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Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:33AM +0100, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Applied, though the above isn't strictly true - a number of later AC'97 CODECs had PLLs which allowed them to use non-standard AC'97 MCLKs.

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[alsa-devel] [PATCH v2] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the comment sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm8804: Remove redundant check
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> The condition "if (!freq_in || !freq_out)" has already been tested previously, so no need to do it again. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d321a87..1704b1e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -395,9 +395,6 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, /* power down the PLL before reprogramming it */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - if (!freq_in || !freq_out) - return 0; - /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, pll_div.n | (pll_div.prescale << 4)); -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add optional soft reset workaround
by Daniel Mack 24 Dec '12

24 Dec '12

The CS4271 requires its LRCLK and MCLK to be stable before its RESET line is de-asserted. That also means that clocks cannot be changed without putting the chip back into hardware reset, which also requires a complete re-initialization of all registers. One (undocumented) workaround is to assert and de-assert the PDN bit in the MODE2 register. This patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 12 ++++++++ include/sound/cs4271.h | 15 ++++++++++ sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++ 3 files changed, 61 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9..e2cd1d7 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d..70f4535 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -20,6 +20,21 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ + + /* + * The CS4271 requires its LRCLK and MCLK to be stable before its RESET + * line is de-asserted. That also means that clocks cannot be changed + * without putting the chip back into hardware reset, which also requires + * a complete re-initialization of all registers. + * + * One (undocumented) workaround is to assert and de-assert the PDN bit + * in the MODE2 register. This workaround can be enabled with the + * following flag. + * + * Note that this is not needed in case the clocks are stable + * throughout the entire runtime of the codec. + */ + bool enable_soft_reset; }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac99238..ffa72dd 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: soc-core: Remove unused 'ret' variable
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0d4630..29046e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2996,7 +2996,6 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: atmel-soc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a couple of format type warnings. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi, Build tested on RM9200 and x86_64. ARCH_AT91 depend is pushed down to the machine drivers since they still relies on mach includes. regards Joachim Eastwood sound/soc/atmel/Kconfig | 6 +++--- sound/soc/atmel/atmel-pcm-pdc.c | 4 ++-- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 -- 4 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1b691b..3fdd87f 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 + depends on HAS_IOMEM help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 6a293c7..054ea4d 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, pr_debug("atmel-pcm: " "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", + "(dma_bytes=%zu, period_size=%zu)\n", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, int ret = 0; pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", + "dma_area = %p, dma_bytes = %zu\n", rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index e99f181..3109db7 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", (void *)buf->area, (void *)buf->addr, size); if (!buf->area) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1c76634..2755750 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -42,8 +42,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" #include "atmel_ssc_dai.h" -- 1.8.0

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[alsa-devel] [PATCH] ASoC: atmel-ssc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a truncated warning that would come when building on a 64-bit arch. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi Mark, Sending it to you since seem like you're the last one who commited changes to the driver and the driver is mostly used on ASoC. Tested on RM9200 and build tested on x86_64. regards Joachim Eastwood drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/misc/Kconfig b/drivers/misc/Kconfig index b151b7c..8f59d88 100644 --- a/drivers/misc/Kconfig +++ b/drivers/misc/Kconfig @@ -192,7 +192,7 @@ config ICS932S401 config ATMEL_SSC tristate "Device driver for Atmel SSC peripheral" - depends on AVR32 || ARCH_AT91 + depends on HAS_IOMEM ---help--- This option enables device driver support for Atmel Synchronized Serial Communication peripheral (SSC). diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 158da5a..0cee274 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -167,7 +167,7 @@ static int ssc_probe(struct platform_device *pdev) /* disable all interrupts */ clk_enable(ssc->clk); - ssc_writel(ssc->regs, IDR, ~0UL); + ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); clk_disable(ssc->clk); -- 1.8.0

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[alsa-devel] [PATCH v2] ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

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[alsa-devel] [PATCH v2] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Add adc, mic, sidetone volume ranges and appropriately added the controls. Fix the DAC HP/EP/LS/LO/HA maximum gain values. Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

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[alsa-devel] [PATCH] ASoC: wm8962: Add device tree support
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add device tree support. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8962.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8962.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 0000000..dceb3b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8fd38cb..1ebeff4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
by Mark Brown 24 Dec '12

24 Dec '12

On Fri, Dec 21, 2012 at 06:17:12PM +0800, Chuansheng Liu wrote: > > Just found some cases that some codec drivers set the bias to _STANDBY and > set idle_bias_off to 1 during probing. > It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, > there is no reason to start from _STANDBY bias with idle_bias_off == 1. Applied, thanks.

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[alsa-devel] [PATCH RFT] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
by Axel Lin 24 Dec '12

24 Dec '12

The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi M R Swami, I don't have the datasheet. Can you check if this patch is correct? Thanks, Axel sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..c0d203b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Add sgtl5000 into the Subject sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] Does it will effect the precision of DMA position?
by Zhang wei 24 Dec '12

24 Dec '12

Hello guys. In the function snd_pcm_update_hw_ptr0(): pos -= pos % runtime->min_align; after running it, the 'pos' maybe smaller than before.because there is need to aligned.so that means lost or drop some Audio data for ever ,is that true? The data playbacked is a bit of different from we send to? Best Regards zw

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[alsa-devel] [alsa-lib, dmix] Support for S24_3BE
by Vincent Vinel 24 Dec '12

24 Dec '12

Hello, The bugtracker seems to be down, so I'm posting here instead. I was trying to use software mixing with my Native Instruments USB soundcard (snd-usb-caiaq), but all attemps to use the dmix pcm ended up with the rather unhelpful error : ALSA lib pcm_dmix.c:1311:(_snd_pcm_dmix_open) Unsupported format When using the hw pcm, however, aplay was kind enough to display the list of available formats, which in my case was limited to S24_3BE. A quick check confirmed that dmix doesn't support that format, which happens to be the only format available for NI USB soundcards (as defined in snd-usb-caiaq). Is it worth writing a patch for 24bit big-endian support in dmix ? External soundcards like these ones are usually used for realtime effects processing and low-latency audio, things for which dmix proably isn't suited anyway. -- Vincent Vinel

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Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Grant Likely 24 Dec '12

24 Dec '12

On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna <padma.v(a)samsung.com> wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > .../devicetree/bindings/sound/samsung-i2s.txt | 75 +++++++ > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 233 ++++++++++++++++---- > 4 files changed, 269 insertions(+), 41 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > new file mode 100644 > index 0000000..bff47b4 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > @@ -0,0 +1,75 @@ > +* Samsung I2S controller > + > +Required SoC Specific Properties: > + > +- compatible : "samsung,samsung-i2s" Isn't that kind of redundant? :-) The format of the compatible strings should be "<vendor>,<part-number>-i2s". Please be specific about the part number that you're doing the binding for. For example; use "samsung,exynos4210-i2s" instead of "samsung,exynos-i2s". > +- reg: physical base address of the controller and length of memory mapped > + region. > + > +[PRELIMINARY: the dma channel allocation will change once there are > +official DMA bindings] > + > +- tx-dma-channel-secondary: The dma channel specifier for secondary tx > + operations. The format of the dma specifier depends on the dma > + controller. > +- tx-dma-channel: The dma channel specifier for tx operations. The format of > + the dma specifier depends on the dma controller. > +- rx-dma-channel: The dma channel specifier for rx operations. The format of > + the dma specifier depends on the dma controller. > + > +Optional SoC Specific Properties: > + > +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel > + support, this flag is enabled. > +- samsung,supports-rstclr: This flag should be set if I2S software reset bit > + control is required. When this flag is set I2S software reset bit will be > + enabled or disabled based on need. > +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, > + then this flag is enabled. > +- samsung,idma-addr: Internal DMA register base address of the audio > + sub system(used in secondary sound source). > + > +Required Board Specific Properties: > + > +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK > + interface lines. The format of the gpio specifier depends on the gpio > + controller. > + The syntax of samsung gpio specifier is > + <[phandle of the gpio controller node] > + [pin number within the gpio controller] > + [mux function] > + [flags and pull up/down] > + [drive strength]> > + > +Aliases: > + > +- All the I2S controller nodes are represented in the aliases node using > + the following format 'i2s{n}' where n is a unique number for the alias. Are the aliases actually being used now? If not, then drop this section. I don't see any references in this patch. Otherwise the binding looks good to me. g.

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[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

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[alsa-devel] [PATCH] ALSA: usb-audio: support delay calculation on capture streams
by Pierre-Louis Bossart 21 Dec '12

21 Dec '12

Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..3a38447 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -59,7 +59,12 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, /* Approximation based on number of samples per USB frame (ms), some truncation for 44.1 but the estimate is good enough */ - est_delay = subs->last_delay - (frame_diff * rate / 1000); + est_delay = frame_diff * rate / 1000; + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + est_delay = subs->last_delay - est_delay; + else + est_delay = subs->last_delay + est_delay; + if (est_delay < 0) est_delay = 0; return est_delay; @@ -78,8 +83,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); @@ -1147,6 +1151,10 @@ static void retire_capture_urb(struct snd_usb_substream *subs, int i, period_elapsed = 0; unsigned long flags; unsigned char *cp; + int current_frame_number; + + /* read frame number here, update pointer in critical section */ + current_frame_number = usb_get_current_frame_number(subs->dev); stride = runtime->frame_bits >> 3; @@ -1180,6 +1188,15 @@ static void retire_capture_urb(struct snd_usb_substream *subs, subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + /* capture delay is by construction limited to one URB, + * reset delays here + */ + runtime->delay = subs->last_delay = 0; + + /* realign last_frame_number */ + subs->last_frame_number = current_frame_number; + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ if (oldptr + bytes > runtime->buffer_size * stride) { -- 1.7.9.5

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[alsa-devel] [PATCH RFT] ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
by Axel Lin 21 Dec '12

21 Dec '12

According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ #define WM2200_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Mark, wm2200.h defines WM2200_AIF1_FMT_SHIFT to be 0, so current code looks wrong to me. I don't have the datasheet, so can you check if this patch is correct? Axel sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31d..a12fc2f 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1626,7 +1626,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ASoC: fsi: cleanup FSI driver
by Kuninori Morimoto 21 Dec '12

21 Dec '12

Hi Mark Cc Simon These patches changed platform setting to use FSI new style, and cleanup FSI driver. Kuninori Morimoto (3): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h arch/arm/mach-shmobile/board-ap4evb.c | 11 +-- arch/arm/mach-shmobile/board-mackerel.c | 11 +-- arch/sh/boards/mach-ecovec24/setup.c | 12 +-- arch/sh/boards/mach-se/7724/setup.c | 12 +-- include/sound/sh_fsi.h | 70 +------------- sound/soc/sh/fsi.c | 156 +++---------------------------- 6 files changed, 27 insertions(+), 245 deletions(-) >> Simon #2 patch modifies both FSI driver and platform. So, it needs your Acked-by Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCHv2 0/9] ALSA: CA0132: DSP Update
by Ian Minett 21 Dec '12

21 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Thanks for the comments. This resubmit should help to correct many of the points raised. For example, use of C99-style init for structures, the previous large patch is now split into several subpatches, and there are more descriptions to the code and individual patchfiles. Patch Summary: 1: Add new definitions and structures that will be used by the DSP routines 2: Init chip registers, DSP parameters and mixer settings 3: Add the new DSP mixer kcontrols and helpers. 4: Update and add PCM callbacks and enhancements. 5: Add unsol handler for DSP response and jack detection reporting 6: Add tuning controls 7/8: Code shuffle and tidy, split into two for a cleaner diff. 9: Updates to the mic sample rate and module description, and fix potential errors on init. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

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Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the driver for the Tegra 2x AC97 host controller. The AC'97 DT binding file should really be added as part of this patch. I'm not at all familiar with AC'97, but this mostly looks fine. I have a bunch of comments below though, mostly for my own understanding. > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA20_AC97 > + tristate > + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC This should also select SND_SOC_TEGRA20_DAS, just like the I2S driver. > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > +static struct tegra20_ac97 *workdata; Why not put that into the device's drvdata? Oh, is there no way to stash driver-private data into the struct snd_ac97? > +static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* reset line is not driven by DAC pad group, have to toggle GPIO */ > + gpio_set_value(workdata->reset_gpio, 0); > + udelay(2); I'm not quite sure what that implies. Might some (Tegra) HW designs use a dedicated signal from the AC'97 HW to reset the CODEC and others not? That would make the GPIO optional. Or, does the comment mean that Tegra doesn't ever have a dedicated CODEC reset signal, so we always have to use a GPIO? If so, this comment might be more appropriate inside probe() where I asked my related questions. > +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* > + * although sync line is driven by the DAC pad group warm reset using > + * the controller cmd is not working, have to toggle sync line > + * manually. > + */ > + gpio_request(workdata->sync_gpio, "codec-sync"); Hmm. There's an AC'97 command to reset the CODEC and we don't implement it? Uggh. > +static const struct snd_soc_dai_ops tegra20_ac97_dai_ops = { > + .trigger = tegra20_ac97_trigger, > +}; No .set_fmt or .hw_params? Does the AC'97 ASoC core code handle that somehow? I guess that's what soc_ac97_ops is for? > +static struct snd_soc_dai_driver tegra20_ac97_dai = { > + .name = "tegra-ac97-pcm", > + .ac97_control = 1, > + .probe = tegra20_ac97_probe, > + .playback = { > + .stream_name = "PCM Playback", Out of curiosity, why "PCM Playback" not just "Playback"? > +static bool tegra20_ac97_volatile_reg(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case TEGRA20_AC97_CMD: ... > + return true; CMD is volatile; did we put status bits in there? > +static int tegra20_ac97_platform_probe(struct platform_device *pdev) > + ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-reset-gpio", 0); > + if (gpio_is_valid(ac97->reset_gpio)) { > + ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, > + GPIOF_OUT_INIT_HIGH, "codec-reset"); Shouldn't this get the flags from the GPIO specifier, and deal with active-high/active-low resets, or is the polarity set by the AC'97 spec? > + if (ret) { > + dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); > + goto err_clk_put; > + } > + } else { > + dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); > + goto err_clk_put; > + } Is a reset GPIO necessarily required? What if a board arranged to reset the CODEC during power-on-reset, and there was no software control over the reset? > + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-sync-gpio", 0); > + if (!gpio_is_valid(ac97->sync_gpio)) { > + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); > + goto err_clk_put; > + } I don't know what this is, so I'll ask if it's strictly required too. > + ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); snd_soc_register_dai() would be marginally simpler. > + ret = clk_prepare_enable(ac97->clk_ac97); > + if (ret) { > + dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); > + goto err_asoc_utils_fini; > + } Can you do runtime PM instead? See the I2S driver for an example. > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/20/2012 01:10 PM, Lucas Stach wrote: > Am Donnerstag, den 20.12.2012, 12:44 -0700 schrieb Stephen Warren: >> On 12/19/2012 04:17 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) >>> +{ >>> + u32 readback; >>> + unsigned long timeout; >>> + >>> + /* >>> + * although sync line is driven by the DAC pad group warm reset using >>> + * the controller cmd is not working, have to toggle sync line >>> + * manually. >>> + */ >>> + gpio_request(workdata->sync_gpio, "codec-sync"); >> >> Hmm. There's an AC'97 command to reset the CODEC and we don't implement >> it? Uggh. > > As far as I could figure from the downstream register doc there is a > command implemented in the host controller that's supposed to do the > warm reset, but in my testing it didn't work. Sadly I have not received > any documentation about the host controller from NVIDIA up until now, so > I don't now if we could make this work somehow. Bug Submission ID#: > 196075 What system was that bug submitted into? It's not a regular NVIDIA internal bug number. I'll see who I can ask about this feature... >>> + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, >>> + "nvidia,codec-sync-gpio", 0); >>> + if (!gpio_is_valid(ac97->sync_gpio)) { >>> + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); >>> + goto err_clk_put; >>> + } >> >> I don't know what this is, so I'll ask if it's strictly required too. > > Until we manage to get the host controller warm reset command working > this is fixed to be the _FS line of the DAP connected to the codec, so > strictly required. I'd like to investigate whether the warm reset command can be made to work before actually committing to a DT binding that requires this GPIO to be specified; the DT binding is supposed to be basically static.

1 0

Re: [alsa-devel] [PATCH 7/7] ASoC: tegra: add machine driver using wm9712 codec
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds a simple machine driver using the Wolfson wm9712 AC97 > codec. If the AC'97 code is going to be reworked to probe the CODEC rather than manually specify it in DT, then will this machine driver become generic for any AC'97 system, and hence you could rename the DT binding to nvidia,tegra-audio-ac97? > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt > +- nvidia,audio-codec : The phandle of the WM9712 audio codec I wonder how you'll get that if you dynamically probe the AC'97 bus... > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA_WM9712 > + tristate "SoC Audio support for Tegra boards using a WM9712 codec" > + depends on SND_SOC_TEGRA I'd suggest making this depend on the Tegra AC97 module, or ARCH_TEGRA_2x_SOC instead, since it can only run on Tegra20. The AC'97 module has been removed on Tegra30 and replaced with an HD Audio controller instead. > + select SND_SOC_TEGRA20_AC97 if ARCH_TEGRA_2x_SOC That select (or a dependency as mentioned above) is always required. > + select SND_SOC_AC97_BUS Shouldn'y the AC'97 driver select that? > diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c > +static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) > + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); Hmm. I guess that potentially ties this driver to whichever CODECs actually expose a widget with that exact name... > +static int tegra_wm9712_remove(struct snd_soc_card *card) > +{ > + return 0; > +} Do you need to implement this at all if it does nothing? > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 6/7] ASoC: tegra: add ac97 host controller to device tree
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the DT bindings and a default entry for the Tegra 2x > AC97 host controller. Everything here except the .dtsi file should be part of the previous patch. > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt > +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number > + of the GPIO corresponding with the AC97 DAP _FS line Hmm, for I2S at least, FS (frame sync I assume) definitely is something that the audio controller itself drives. Are you absolutely sure that this should be represented as a GPIO here? > diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi > + tegra_ac97: ac97@70002000 { There's only 1 node named ac97, so you can leave out the unit address since it's not required to generate a unique node name. In other words just "ac97".

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Re: [alsa-devel] [PATCH 1/7] ARM: tegra: add ac97 clock
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > Add clock used by the AC97 host controller. This is going to conflict with Prashant's clock driver rework. Can you rebase it on top of that once it's checked in. That might be a couple weeks out though.

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Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Acked-by: Stephen Warren <swarren(a)nvidia.com>

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Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This connection is used by the AC97 controller. Acked-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:07 PM, prasannatsmkumar <prasannatsmkumar(a)gmail.com> wrote: > Hi ALSA people, > > I am having a Realtek ALC 883 chip. The sound volume is very low > compared to what I hear in windows - I maintain same speaker volume > and play same audio. If I keep the volume as 10% in windows I have to > keep a volume of at least 25+% in Linux. Is there any way to improve > it? I have been seeing such a situation for past 4 or 5 years (from > Fedora 9 period till today). I have tried several distributions > (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the > same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.25. > > Should I file a bug? If so in which component (kernel or pulseaudio)? > > Thanks in advance, > PrasannaKumar Resending this as this mail was bounced by alsa devel list. Please feel free to ask for information necessary to solve this issue. I have copied output of some commands that may be useful for understanding or digging this issue. dmesg | grep hda output: [ 30.648663] snd_hda_intel 0000:00:1b.0: >irq 43 for MSI/MSI-X [ 30.716741] hda_codec: ALC883: SKU not ready 0x411111f0 lspci -vvv output (only info related audio device is in mail all others are stripped): 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: Intel Corporation Device d605 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 43 Region 0: Memory at 501c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel amixer output: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 13 [42%] [-27.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 8 [26%] [0.00dB] [on] Front Right: Capture 8 [26%] [0.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [off] Front Right: Capture 0 [0%] [-12.00dB] [off] Simple mixer control '2ch-Upmix Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 255 Front Left: 255 [100%] Front Right: 255 [100%] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Rear Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Thanks and Regards, PrasannaKumar

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[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

Hi ALSA / pulseaudio people, Resending this as this mail was bounced by alsa devel list. I am having a Realtek ALC 883 chip. The sound volume is very low compared to what I hear in windows - I maintain same speaker volume and play same audio. If I keep the volume as 10% in windows I have to keep a volume of at least 25+% in Linux. Is there any way to improve it? I have been seeing such a situation for past 4 or 5 years (from Fedora 9 period till today). I have tried several distributions (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. Should I file a bug? If so in which component (kernel or pulseaudio)? Thanks in advance, PrasannaKumar

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[alsa-devel] [PATCH RFT] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
by Axel Lin 20 Dec '12

20 Dec '12

Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Rajeev, I don't have this hardware, can you test this patch? Thanks, Axel sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 20 Dec '12

20 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,6 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- 1.7.8.3 Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH RFT] ASoC: arizona: Do proper shift for setting aif1 rate
by Axel Lin 20 Dec '12

20 Dec '12

ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..38248a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 06:37:26AM +0000, Liu, Chuansheng wrote: > Meanwhile, is it useful to add one warning there for that case? > After all, in probing, set the bias to _STANDBY even idle_bias_off == 1, and calling get_runtime_sync(), it > will let the code more obscure. So giving a warning there to indicate the driver: > it is not suggested that in probing, set the bias to _STANDBY even idle_bias_off == 1. Probably, send a patch please. Like I say it is possible to start off in _STANDBY providing the driver grabs the runtime PM reference too but I can't think of any reason for doing that so the warning seems sensible.

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Re: [alsa-devel] [PATCH 2/7] ASoC: allow wolfson wm9712 codec to be instantiated using device tree
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 12:17:31AM +0100, Lucas Stach wrote: > static struct platform_driver wm9712_codec_driver = { > .driver = { > .name = "wm9712-codec", > .owner = THIS_MODULE, > + .of_match_table = wm9712_of_match, > }, No, this is really not a sane way of handling AC'97 at all. They're not platform devices, there is no sane reason why they should appear in the device tree as such. Really we ought to be just hotplugging them, AC'97 is perfectly enumerable.

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[alsa-devel] Adding wm8962 support to mx6qsabresd
by Fabio Estevam 20 Dec '12

20 Dec '12

Hi, I am working on adding audio support to mx6qsabresd board, which contains a wm8962 codec. Here is the log with debug information turned on that I am getting so far: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 200 => ffffffff wm8962 0-001a: read 201 => 0 wm8962 0-001a: read 202 => 0 wm8962 0-001a: read 203 => ffffffff wm8962 0-001a: read 204 => 8100 wm8962 0-001a: read 205 => 8100 wm8962 0-001a: DMIC not in use, disabling input: WM8962 Beep Generator as /devices/soc.0/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input0 wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 300 => 1c00 wm8962 0-001a: read 17 => 160 wm8962 0-001a: read 48 => 0 wm8962 0-001a: read 8 => ffffffff wm8962 0-001a: read 52 => 4 wm8962 0-001a: read 19 => 0 wm8962 0-001a: write a0 = 149 wm8962 0-001a: write a1 = 271 wm8962 0-001a: write a2 = 7 wm8962 0-001a: FLL configured for 24000000Hz->11289600Hz wm8962 0-001a: No audio clocks configured ... imx-wm8962 sound.14: wm8962 <-> 2028000.ssi mapping ok root@freescale /home$ ./aplay Front_Left.wav wm8962 0-001a: hw_params set BCLK 1411200Hz LRCLK 44100Hz wm8962 0-001a: wm8962->sysclk_rate = 11289600 wm8962 0-001a: wm8962->lrclk = 44100 wm8962 0-001a: Selected sysclk ratio 256 wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 3 => 100 wm8962 0-001a: write 3 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -5 wm8962 0-001a: read 29 => 100 wm8962 0-001a: write 29 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -5 wm8962 0-001a: read 2 => 100 wm8962 0-001a: write 2 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -5 wm8962 0-001a: read 28 => 100 wm8962 0-001a: write 28 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -5 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 wm8962 0-001a: DC servo complete after 1ms Playing WAVE 'Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Any idea of why register 4 and 42 read fails? wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 Any suggestion is appreciated. Thanks, Fabio Estevam

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[alsa-devel] how to check the CODEC registers vaule
by Zhang wei 20 Dec '12

20 Dec '12

Hello everyone,I am confused again : ( I read the wm8985.pdf and found the description: If the address is not recognised or the R/W bit is '1' when operating in write only mode,the WM8985 returns to the idle conditon and waits for a new start condition and valid address. Does it mean the wm8985 registers write only, can not be read? I found other kinds of codec datasheet same as it. Is it necessary to read back the value from the register ? Thanks a lot. Best Regards zw

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: core: Fix SOC_DOUBLE_RANGE() macros
by Mark Brown 19 Dec '12

19 Dec '12

Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 10 ++++++---- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++-- 2 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a27822..1e4ea78 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1738d5..d4732ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2922,7 +2922,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2946,12 +2946,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2959,7 +2961,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2979,11 +2995,13 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -2993,6 +3011,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); -- 1.7.10.4

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[alsa-devel] Newbie help with WM8962 and tinyalsa
by Randi Eckstein 19 Dec '12

19 Dec '12

If anyone has an explanation on how to use tinymix to play on a wm8962, I'd really appreciate it. Thanks, Randi Eckstein

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 19 Dec '12

19 Dec '12

On Wed, Dec 19, 2012 at 06:36:37PM +0800, Chuansheng Liu wrote: > But some devices has been set to STANDY bias directly during device probing, > such as cs42l73_probe(): > cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > Then it will cause runtime_get() not be called but laterly runtime_put() will > be called. Also found some other uppaired cases. This is just a bug in the driver, if it's idle_bias_off then it really should be starting in _OFF or at the very least starting actually in _STANDBY (including taking the runtime reference) rather than partially in _STANDBY. > So here add new flag get_runtime, and the logic will be: > 1/ when device is from off to non-off bias, runtime_get() will be called if not yet; > 2/ When device is off bias, runtime_put() will be called if runtime_get() has > been called; This is really not a good idea at all, it's just adding new special cases and making the code more obscure.

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[alsa-devel] Speaker not working; muted?
by Martin Michlmayr 19 Dec '12

19 Dec '12

I've a problem with audio on my HP 2560p laptop (using snd_hda_intel): the speaker no longer works. However, I successfully get sound on a headphone. I've attached the output of alsa-info. It would be great if someone could take a look at let me know if this is a bug in the driver or wrong settings. Background: I used ALSA successfully in the past to play audio with the in-built speaker. At some point, I used aumix (which requires OSS modules) to turn the volume to zero. I believe I turned the volume back up again. I also rebooted my laptop at some point. I don't know if it's related to using aumix or something else, but at some point the in-built speaker stopped working... I can still listen to audio using a headphone. Another observation is that my laptop has a speaker mute button. When I reboot my laptop, and the BIOS or grub is running, I see that the mute button is activated. However, pressing the button doesn't change anything... when the Linux kernel loads, the mute button is still on, but when ALSA starts the button changes colour: it seems to me that ALSA thinks that mute is off while in reality it is on. -- Martin Michlmayr http://www.cyrius.com/

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add noise gate hold time enumeration
by Mark Brown 18 Dec '12

18 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a85a68f..89c0433 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -323,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index db0c8dd..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: arizona: Correct FLL source definitions
by Mark Brown 18 Dec '12

18 Dec '12

The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 7f22b4f..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of more Haswell HDMI codec pins
by mengdong.lin＠intel.com 18 Dec '12

18 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec may report invalid connection list entries on all three pins. But our previous patch only fixes the HDMI pin, not the DP pin, so audio playbak via DP can still fail. This patch adds fixup for Haswell to set proper connection list entries for the other 2 pins including the DP pin. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 42fdf14..73d298a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1710,8 +1710,14 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, if (vendor_param == -1 || !(vendor_param & 0x02)) return; + snd_hda_override_conn_list(codec, 0x05, 3, list); + hdmi_read_pin_conn(codec, 1); + snd_hda_override_conn_list(codec, 0x06, 3, list); hdmi_read_pin_conn(codec, 1); + + snd_hda_override_conn_list(codec, 0x07, 3, list); + hdmi_read_pin_conn(codec, 1); } -- 1.7.9.5

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Re: [alsa-devel] [pulseaudio-discuss] Front mic not working properly
by Raymond Yau 18 Dec '12

18 Dec '12

>> > >> > I'm totally stuck with a sound problem I fail to debug on my own. >> > >> > My setup: >> > Asus P5Q Motherboard with Realtec ALC1200 soundchip. Stereo speakers plugged in to the rear output and headphone speaker/mic combo plugged in to the front jacks. System: Ubuntu 12.10 >> > Desired behaviour: Sound is played back on headphones AND rear speaker. Front mic should "just-work" Asus P5Q specification Realtek ALC1200, 8-channel High Definition Audio CODEC - Support Jack-Detection, Multi-streaming, and Front Panel Jack-Retasking Alt_playback is still not enabled to support multi streaming playback on realtec codecs >> > Problem 2: >> > This is the biggest problem. >> > As soon as I plug in the front mic, Inside gnome-alsamixer the volume for master, surround,center,lfe start to jump between on and off. Why is the Front mic jack controlling these values? Do you mean BIOS did not setup pin default correctly after asus-p5q model was removed? http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6f… > How can I debug this behaviour? Post the output of alsa-info.sh > I just want them to stay the same, if the mic is plugged in or not. >> > Inside the gnome volume settings, the output port starts to jump between "Analog Output" and "Analog Headphones" >> >> This look like problem in Jack detection of the headphone and Mic Jack of front audio panel >> >> alsamixer -c0 >> >> Does the playback switch change rapidly on and off in alsamixer ? >> >> Do your computer chassis have a hda front audio panel and you connect them to the front audio connector of the motherboard according to the user manual ? >> >> Does it help if you disabled the automute switch ? >> > Thanks a lot. Your input helped me to at least work around the problem. My chassis has a HDA and an AC97 connector. The HDA connector was plugged in and correctly configured in BIOS. > I now use the AC97 connector (and changed the BIOS settings) and everything is working properly. Checking the front audio panel specification of your computer chassis Setting front audio panel to AC97 in BIOS of Asus motherboard change the miscellaneous bit of pin default of front HP and Mic Jack （i.e. misc = NO_PRESENCE) The alsa driver won't enable the unsolicited event on these nodes > > Regarding your questions: Disabling automute didn't help. And yes, the playback switch toggled rapidly. http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Try disable trigger sense by codec->no_trigger_sense = 1; > > One thing I didn't test: I mistakenly plugged the rear speaker into the front channel jack (plugging the green connector into a green jack seemed so obvious). I don't know is this is the same channel as the headphone connector on the front panel... > > In AC97 mode it doesn't matter if I use the front channel jack or the rear channel jack (both at the rear of the computer) to connect my main speakers.

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[alsa-devel] [PATCH 0/2] A few fixes for Reatek codecs
by Takashi Iwai 17 Dec '12

17 Dec '12

Hi, here is a patch series for fixing some obvious errors I've overseen for Realtek codecs. It'll be queued for 3.8-rc1. Takashi

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