December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ALSA alsaSink and NONINTERLEAVED vs INTERLEAVED
by Radivoje Jovanovic 24 Dec '10

24 Dec '10

Hello, I have a simple sound card with a stereo output. The driver works just fine when using aplay or gstreamer to play wave files. The problem occurs when I am trying to play videos through gstreamer. I use following gstremaer pipeline (just to test sound): gst-launch-0.10 filesrc location=MPEG4_MOVIE_NAME ! qtdemux name=t ! queue ! faad ! alsasink If I make my driver with following settings: .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), gstreamer comes back with a following message: Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could not get/set settings from/on resource. Additional debug info: gstalsasink.c(519): set_hwparams (): /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Unable to set hw params for playback: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... If I make my driver with following settings: .info = (SNDRV_PCM_INFO_NONINTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), gstreamer will play sound. Of course sound will not be correct since the data is interleaved. Any idea why is this happening? Here is the rest of the snd_pcm_hardware struct: .formats = SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U8, .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .period_bytes_min = 16384, .period_bytes_max = 16384*4, .periods_min = 4, .periods_max = 256, .buffer_bytes_max= 131072, Cheers Ogi

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Re: [alsa-devel] [PATCH] ASoC: Remove WM8995 write sequencer bitfield definitions
by Liam Girdwood 23 Dec '10

23 Dec '10

On Thu, 2010-12-23 at 12:04 +0000, Mark Brown wrote: > They're very verbose and extremely repetitive so bulk up the kernel more > than is ideal. If required we can readd with WRITE_SEQUENCER_n type > definitions that cover the entire register bank in a few defines. > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > sound/soc/codecs/wm8995.h | 4480 --------------------------------------------- > 1 files changed, 0 insertions(+), 4480 deletions(-) > Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH] ASoC: Remove incorrect WM8903 erratum workaround
by Mark Brown 23 Dec '10

23 Dec '10

Due to a typographical error in the erratum workaround it was never functional so just remove it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 20 ++++++-------------- sound/soc/codecs/wm8903.h | 19 ------------------- 2 files changed, 6 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d015745..4a12ac2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1013,7 +1013,7 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg, reg2; + u16 reg; switch (level) { case SND_SOC_BIAS_ON: @@ -1037,23 +1037,15 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); - /* Enable low impedence charge pump output */ - reg = snd_soc_read(codec, - WM8903_CONTROL_INTERFACE_TEST_1); - snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, - reg | WM8903_TEST_KEY); - reg2 = snd_soc_read(codec, WM8903_CHARGE_PUMP_TEST_1); - snd_soc_write(codec, WM8903_CHARGE_PUMP_TEST_1, - reg2 | WM8903_CP_SW_KELVIN_MODE_MASK); - snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, - reg); - /* By default no bypass paths are enabled so * enable Class W support. */ dev_dbg(codec->dev, "Enabling Class W\n"); - snd_soc_write(codec, WM8903_CLASS_W_0, reg | - WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V); + snd_soc_update_bits(codec, WM8903_CLASS_W_0, + WM8903_CP_DYN_FREQ | + WM8903_CP_DYN_V, + WM8903_CP_DYN_FREQ | + WM8903_CP_DYN_V); } reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index 996435e..7f9f6be 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -98,8 +98,6 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_INTERRUPT_STATUS_1_MASK 0x7A #define WM8903_INTERRUPT_POLARITY_1 0x7B #define WM8903_INTERRUPT_CONTROL 0x7E -#define WM8903_CONTROL_INTERFACE_TEST_1 0x81 -#define WM8903_CHARGE_PUMP_TEST_1 0x95 #define WM8903_CLOCK_RATE_TEST_4 0xA4 #define WM8903_ANALOGUE_OUTPUT_BIAS_0 0xAC @@ -1206,23 +1204,6 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_IRQ_POL_WIDTH 1 /* IRQ_POL */ /* - * R129 (0x81) - Control Interface Test 1 - */ -#define WM8903_USER_KEY 0x0002 /* USER_KEY */ -#define WM8903_USER_KEY_MASK 0x0002 /* USER_KEY */ -#define WM8903_USER_KEY_SHIFT 1 /* USER_KEY */ -#define WM8903_USER_KEY_WIDTH 1 /* USER_KEY */ -#define WM8903_TEST_KEY 0x0001 /* TEST_KEY */ -#define WM8903_TEST_KEY_MASK 0x0001 /* TEST_KEY */ -#define WM8903_TEST_KEY_SHIFT 0 /* TEST_KEY */ -#define WM8903_TEST_KEY_WIDTH 1 /* TEST_KEY */ - -/* - * R149 (0x95) - Charge Pump Test 1 - */ -#define WM8903_CP_SW_KELVIN_MODE_MASK 0x0006 /* CP_SW_KELVIN_MODE - [2:1] */ -#define WM8903_CP_SW_KELVIN_MODE_SHIFT 1 /* CP_SW_KELVIN_MODE - [2:1] */ -#define WM8903_CP_SW_KELVIN_MODE_WIDTH 2 /* CP_SW_KELVIN_MODE - [2:1] */ /* * R164 (0xA4) - Clock Rate Test 4 -- 1.5.4.3

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[alsa-devel] [PATCH] ASoC: mcbsp: Add McBSP support for OMAP4
by Olaya, Margarita 23 Dec '10

23 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds McBSP support for the OMAP4 CPU Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 38 +++++++++++++++++++++++++++++++++++--- sound/soc/omap/omap-mcbsp.h | 4 ++++ 2 files changed, 39 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7e84f24..69cb4d1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -102,6 +102,17 @@ static const int omap24xx_dma_reqs[][2] = { static const int omap24xx_dma_reqs[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const int omap44xx_dma_reqs[][2] = { + { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX }, + { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX }, + { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX }, + { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX }, +}; +#else +static const int omap44xx_dma_reqs[][2] = {}; +#endif + #if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, @@ -147,6 +158,21 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const unsigned long omap44xx_mcbsp_port[][2] = { + { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap44xx_mcbsp_port[][2] = {}; +#endif + static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -224,7 +250,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x()) { + if (cpu_is_omap343x() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -332,6 +358,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else if (cpu_is_omap343x()) { dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap34xx_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap44xx()) { + dma = omap44xx_dma_reqs[bus_id][substream->stream]; + port = omap44xx_mcbsp_port[bus_id][substream->stream]; } else { return -ENODEV; } @@ -498,11 +527,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr2 |= XINTM(3) | FREE; regs->spcr1 |= RINTM(3); /* RFIG and XFIG are not defined in 34xx */ - if (!cpu_is_omap34xx()) { + if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) { regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; } - if (cpu_is_omap2430() || cpu_is_omap34xx()) { + if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) { regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE; regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE; } @@ -623,6 +652,9 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (cpu_class_is_omap1()) { err = -EINVAL; break; + } else if (cpu_is_omap44xx()) { + regs->srgr2 &= ~CLKSM; + break; } err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, MCBSP_CLKS_PRCM_SRC); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ffdcc5a..110c106 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -50,6 +50,10 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 3 #endif +#if defined(CONFIG_ARCH_OMAP4) +#undef NUM_LINKS +#define NUM_LINKS 4 +#endif #if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) #undef NUM_LINKS #define NUM_LINKS 5 -- 1.7.0.4

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 23 Dec '10

23 Dec '10

The following changes since commit 9d01df063e70260d6c0aabd58dd5507db151aa51: ASoC: don't pass the string as the format arguemtn for dev_info() (2010-12-22 14:08:40 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dimitris Papastamos (1): ASoC: Add initial WM8995 driver Mark Brown (4): ASoC: Use delayed work to debounce WM8350 jack IRQs ASoC: Trace Wolfson jack detection IRQs ASoC: Add jack IRQ trace to 88pm860x driver ASoC: ifdef out trace points from modules for x86 Seungwhan Youn (1): ASoC: SAMSUNG: Debug wrong parameter sound/soc/codecs/88pm860x-codec.c | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8350.c | 71 +- sound/soc/codecs/wm8903.c | 6 + sound/soc/codecs/wm8962.c | 5 + sound/soc/codecs/wm8994.c | 5 + sound/soc/codecs/wm8995.c | 1818 ++++++++ sound/soc/codecs/wm8995.h | 8749 +++++++++++++++++++++++++++++++++++++ sound/soc/samsung/goni_wm8994.c | 2 +- 10 files changed, 10649 insertions(+), 20 deletions(-) create mode 100644 sound/soc/codecs/wm8995.c create mode 100644 sound/soc/codecs/wm8995.h

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[alsa-devel] callbacks order while capturing from Line-IN
by Manu Abraham 23 Dec '10

23 Dec '10

Hi, By now I am a bit confused. I am seeing a certain pattern in which the callbacks are invoked: - hw_params() - prepare() - trigger(); // Start DMA - pointer(); --> gets data after interrupt is handled - trigger(); // Stop DMA - prepare(); - trigger(); // Start DMA - pointer(); --> gets data after interrupt is handled .. .. and so on. Is this the right behaviour that's expected from a function ALSA capture driver ? I am trying to understand why there is Start DMA, Stop DMA, Start DMA, Stop DMA in a cyclic fashion for each of the interrupts in my setup. Thanks, Manu

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[alsa-devel] [PATCH 0/3] ASoC: tlv320dac33: Cleanup, and 32/24 bit support
by Peter Ujfalusi 22 Dec '10

22 Dec '10

Hello, the following series adds 32/24 bit support for DAC33 codec. The manual FIFO configuration code will be removed, since that was mainly used for initial debugging, and with the 24 bit support the FIFO sizes are different, so it would be pain to safe guard the manual FIFO settings for both 16, and 24 bit modes. --- Peter Ujfalusi (3): ASoC: tlv320dac33: Remove manual FIFO configuration ASoC: tlv320dac33: Some cleanup for 32/24 bit support ASoC: tlv320dac33: Add 32/24 bit audio support sound/soc/codecs/tlv320dac33.c | 214 +++++++++++----------------------------- 1 files changed, 58 insertions(+), 156 deletions(-) -- 1.7.3.4

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Re: [alsa-devel] [PATCH 08/11] ASoC: tegra: Add tegra-pcm driver
by Stephen Warren 22 Dec '10

22 Dec '10

Mark Brown wrote on Friday, December 17, 2010 3:07 PM: > On Fri, Dec 17, 2010 at 02:41:27PM -0700, Stephen Warren wrote: > > This provides an ASoC platform driver that manages Tegra's APB DMA > > controller. > > Oh, and one other thing: if your DMA controller has a free running mode > it'd be much nicer to use that than to have to continually keep feeding > it buffers. As far as I can tell, Tegra's DMA controller doesn't provide a continuous mode that's useful for audio. The DMA controller does have a continuous mode, which can either continually push the exact same buffer into the I2S FIFO, or can continually push two contiguous same-sized buffers into the I2s FIFO. In either case, I think we get an interrupt half-way through. However, I don't see a way to handle a buffer with <n> chunks, and get an interrupt for each chunk (unless n==2 of course). I suppose we could use continuous mode if we set both periods_min/max==2, but that seems a little inflexible. Finally, for reasons I don't fully understand, even when the HW is in continuous mode, the driver seems to require you to continually queue more DMA requests into the driver's request list anyway, so tegra_pcm.c would be doing just as much work, and hence is presumably no better for CPU usage and power. Perhaps this mechanism allows changing the buffer that continuous mode operates on without shutting down the DMA engine, and hence allows emulation of a larger buffer with interrupts on every chunk with more than two chunks. Or, perhaps this is intended to allow SW to update the HW registers for the next DMA half way through the current DMA, to avoid making the DMA engine wait for the ISR to program the next operation during the ISR. Perhaps Colin can comment more on this? Given all that, I'm tempted to simply stick with single-shot mode for now. -- nvpublic

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[alsa-devel] [PATCH 0/3] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 22 Dec '10

22 Dec '10

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (3): ECI: input: introduce ECI accessory input driver ECI: introducing ECI bus driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 59 +++ drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/ecibus/Kconfig | 46 ++ drivers/ecibus/Makefile | 10 + drivers/ecibus/ecibus.c | 583 ++++++++++++++++++++++++ drivers/input/misc/Kconfig | 18 + drivers/input/misc/Makefile | 2 +- drivers/input/misc/eci.c | 1002 +++++++++++++++++++++++++++++++++++++++++ include/linux/input/eci.h | 165 +++++++ 10 files changed, 1887 insertions(+), 1 deletions(-) create mode 100644 drivers/ecibus/Kconfig create mode 100644 drivers/ecibus/Makefile create mode 100644 drivers/ecibus/ecibus.c create mode 100644 drivers/input/misc/eci.c create mode 100644 include/linux/input/eci.h

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Re: [alsa-devel] format and standards issues fixes
by Takashi Iwai 22 Dec '10

22 Dec '10

Hi, At Wed, 8 Dec 2010 23:14:09 +0530, Jeffrin Jose wrote: > > hello. > > I have fixed several issues with checkpatch.pl > fixes include standards and format issues. > > i have attached the patch along with this mail. Sorry for the late reply. This was buried in my inbox. And thanks for the patch. Unfortunately, your patch doesn't look pretty correct as coding-style fixes. When you break the lines, you don't need the backslash at the end, unless it's in a macro definition. And the line-breaks are inserted in wrong places overall. But, above all, this kind of patch doesn't make much sense for such an old driver code. sound_core.c is in almost dead state, so just rewriting for coding-style is much more harm than benefit. If you'd continue on working on cleaning-up codes, it'd be better to begin with more important places. thanks, Takashi

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