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"

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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 12/20/2012 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.

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