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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 12 Mar '09

12 Mar '09

The following changes since commit 3c081384007332ab95c8d8753f47719d6646bcb2: Mark Brown (1): Merge branch 's3c-iis-header' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 David Brownell (1): ASoC: buildfix for OSK Eric Miao (1): ASoC: make ops a pointer in 'struct snd_soc_dai' Mark Brown (6): ASoC: Add initial driver for the WM8400 CODEC ASoC: Remove version display from WM8580 driver ASoC: Fix formats for s3c24xx-i2s register prints ASoC: Merge dai_ops factor out [ARM] Revert futher extraneous changes from the S3C header move Merge branch 's3c-iis-header' into for-2.6.30 arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/plat-s3c24xx/clock-dclk.c | 1 - include/linux/mfd/wm8400-audio.h | 1 + include/sound/soc-dai.h | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 33 +- sound/soc/au1x/psc-ac97.c | 10 +- sound/soc/au1x/psc-i2s.c | 12 +- sound/soc/blackfin/bf5xx-i2s.c | 14 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ak4104.c | 10 +- sound/soc/codecs/ak4535.c | 14 +- sound/soc/codecs/cs4270.c | 14 +- sound/soc/codecs/ssm2602.c | 20 +- sound/soc/codecs/tlv320aic23.c | 18 +- sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/uda134x.c | 18 +- sound/soc/codecs/uda1380.c | 42 +- sound/soc/codecs/wm8350.c | 20 +- sound/soc/codecs/wm8400.c | 1481 +++++++++++++++++++++++++++++++ sound/soc/codecs/wm8400.h | 62 ++ sound/soc/codecs/wm8510.c | 16 +- sound/soc/codecs/wm8580.c | 34 +- sound/soc/codecs/wm8728.c | 12 +- sound/soc/codecs/wm8731.c | 18 +- sound/soc/codecs/wm8750.c | 14 +- sound/soc/codecs/wm8753.c | 90 +- sound/soc/codecs/wm8900.c | 16 +- sound/soc/codecs/wm8903.c | 18 +- sound/soc/codecs/wm8971.c | 14 +- sound/soc/codecs/wm8990.c | 18 +- sound/soc/codecs/wm9705.c | 8 +- sound/soc/codecs/wm9712.c | 14 +- sound/soc/codecs/wm9713.c | 40 +- sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/fsl/fsl_ssi.c | 18 +- sound/soc/fsl/mpc5200_psc_i2s.c | 20 +- sound/soc/omap/omap-mcbsp.c | 20 +- sound/soc/omap/osk5912.c | 12 +- sound/soc/pxa/pxa-ssp.c | 65 +- sound/soc/pxa/pxa2xx-ac97.c | 13 +- sound/soc/pxa/pxa2xx-i2s.c | 18 +- sound/soc/s3c24xx/s3c2412-i2s.c | 8 +- sound/soc/s3c24xx/s3c2443-ac97.c | 18 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 32 +- sound/soc/sh/ssi.c | 30 +- sound/soc/soc-core.c | 102 ++- 49 files changed, 2047 insertions(+), 450 deletions(-) create mode 100644 sound/soc/codecs/wm8400.c create mode 100644 sound/soc/codecs/wm8400.h

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[alsa-devel] Problem compiling soc - imx31 and wm8753
by Marcelo Utikawa da Fonseca 11 Mar '09

11 Mar '09

Hi! I am new here and do not know if I am posting correctly, please understand! :-) I knew about this list in the wolfsonmicro website and I hope someone can help me... I am facing a problem while compiling the sound support for imx31ads board. I got the kernel from wolfsonmicro using the following command: git clone git://opensource.wolfsonmicro.com/linux-2.6-asoc linux-2.6-asoc After this I just copy the arch/arm/configs/mx31ads_defconfig to .config and run: make ARCH=arm CROSS_COMPILE=arm-none-linux-gnueabi- menuconfig I selected Alsa support, SoC, Support for i.MX CPU and Support for i.MX31 - WM8753. After this I run: make ARCH=arm CROSS_COMPILE=arm-none-linux-gnueabi- zImage and I receive this error message: CC sound/soc/imx/imx-ssi.o In file included from sound/soc/imx/imx-ssi.c:29: include/sound/driver.h:1:2: warning: #warning "This file is deprecated" sound/soc/imx/imx-ssi.c:34:26: error: asm/arch/dma.h: No such file or directory sound/soc/imx/imx-ssi.c:35:28: error: asm/arch/clock.h: No such file or directory sound/soc/imx/imx-ssi.c:37:26: error: asm/hardware.h: No such file or directory In file included from sound/soc/imx/imx-ssi.c:39: sound/soc/imx/imx-ssi.h:10:31: error: asm/arch/hardware.h: No such file or directory In file included from sound/soc/imx/imx-ssi.c:39: sound/soc/imx/imx-ssi.h:213: error: array type has incomplete element type In file included from sound/soc/imx/imx-ssi.c:40: sound/soc/imx/imx31-pcm.h:62: error: expected specifier-qualifier-list before 'dma_channel_params' sound/soc/imx/imx31-pcm.h:65: error: array type has incomplete element type sound/soc/imx/imx-ssi.c:72: error: unknown field 'params' specified in initializer sound/soc/imx/imx-ssi.c:72: error: extra brace group at end of initializer sound/soc/imx/imx-ssi.c:72: error: (near initialization for 'imx_ssi1_pcm_stereo_out0') sound/soc/imx/imx-ssi.c:74: error: 'emi_2_per' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:76: error: 'SSI1_BASE_ADDR' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:77: error: 'DMA_REQ_SSI1_TX1' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:78: error: 'SSI' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:79: warning: excess elements in struct initializer sound/soc/imx/imx-ssi.c:79: warning: (near initialization for 'imx_ssi1_pcm_stereo_out0') sound/soc/imx/imx-ssi.c:84: error: unknown field 'params' specified in initializer sound/soc/imx/imx-ssi.c:84: error: extra brace group at end of initializer sound/soc/imx/imx-ssi.c:84: error: (near initialization for 'imx_ssi1_pcm_stereo_out1') ... ... ... sound/soc/imx/imx-ssi.c:836: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:836: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:837: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:837: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:838: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:838: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:841: warning: type defaults to 'int' in declaration of 'imx_ssi_pcm_dai' make[3]: *** [sound/soc/imx/imx-ssi.o] Error 1 make[2]: *** [sound/soc/imx] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I did not change nothing, just configure and compile... My gcc version is 4.1.2 from Freescale. What can be wrong? Could be this version of gcc a problem? I already tried to correct the missing include files but the other errors are still there after the corrections... Thanks in advance, Marcelo Utikawa da Fonseca --------------------------------------------- Tecnequip Tecnologia em Equipamentos Endereço/Address: Rua Juno, 527 Cidade/City: São Paulo Estado/State: SP País/Country: Brasil CEP/Postal Code: 03445-000 Fone/Phone: 55-11-20937199 FAX: 55-11-29412289

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[alsa-devel] snd-soc-pxa-ssp I2C/FIFO issues
by pHilipp Zabel 11 Mar '09

11 Mar '09

Ok, I'm not sure how to handle this correctly. Magician needs stereo S16_LE input data from alsa, DAIFMT_DSP_A on the wire with a frame size of 16 bit (one frame for each channel) and thus 2-byte DMA transfers to the FIFO (currently called 'mono' DMA params). Zylonite's current code uses 32 bit frame size and 4-byte DMA transfers to emulate I2S (S16_LE,stereo). Network mode seems to work, so it could be changed to use two 16 bit slots instead with 2-byte DMA transfers (which would make Magician happy). But... Daniel's board needs very strange settings because network mode doesn't work, so I guess he can't use 2-byte DMA transfers. Is that correct? If we can't find a way to have all machines use the mono DMA params, there has to be some kind of interface to set the width of the data in the FIFO independently from the Alsa format. How should that interface look like? regards Philipp

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[PATCH] pxa-ssp: switch from network mode to PSP
by Daniel Mack 11 Mar '09

11 Mar '09

This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/pxa-ssp.c | 44 +++++++++++++++++++++++++++++++++----------- 1 files changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 52d97c4..3cde686 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- 1.6.2

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[PATCH] pxa-ssp: switch from network mode to PSP
by Daniel Mack 11 Mar '09

11 Mar '09

This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/pxa-ssp.c | 44 +++++++++++++++++++++++++++++++++----------- 1 files changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 52d97c4..3cde686 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- 1.6.2

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 11 Mar '09

11 Mar '09

The following changes since commit 0a4e1c9069b3cc2a9e5b57898ac9be94314e8efa: Takashi Iwai (1): Merge branch 'for-2.6.30' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (1): ASoC: Fix Samsung S3C2412_IISMOD_SDF_{MSB,LSB} definitions Daniel Mack (1): ASoC: don't touch pxa-ssp registers when stream is running Hugo Villeneuve (2): ALSA: ASoC: Davinci: Replaced DAI format RIGHT_J by DSP_B for SFFSDR ALSA: ASoC: Davinci: Updated sffsdr_hw_params() function to new format Mark Brown (3): ASoC: Convert PXA AC97 driver to probe with the platform device Merge commit 'takashi/topic/asoc' into for-2.6.30 ASoC: Fix up merge with the ARM tree arch/arm/mach-shark/include/mach/io.h | 1 + arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 4 +- sound/soc/davinci/davinci-sffsdr.c | 20 ++++++----- sound/soc/pxa/pxa-ssp.c | 11 ++++++ sound/soc/pxa/pxa2xx-ac97.c | 38 ++++++++++++++++++-- 5 files changed, 59 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH] ASoC: Complete Beagleboard support
by Steve Sakoman 11 Mar '09

11 Mar '09

Commit dc06102a0c8b5aa0dd7f9a40ce241e793c252a87 in the asoc tree did not include the necessary Kconfig and Makefile changes. This patch completes the support for Beagleboard Signed-off-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/Kconfig | 10 ++++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4f7f040..ccd8973 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 76fedd9..0c9e4ac 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.5.6.3

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[alsa-devel] [patch 0/9] S3C24XX/S3C64XX updates
by Ben Dooks 11 Mar '09

11 Mar '09

This is a set of patches for 2.6.30 to add more s3c24xx machine support and support for the S3C64XX series of SoC. It would be very useful to get the header moves merged as soon as possible, to help clear out the old ARM include directories (and make it easier for the merges from other s3c branches that need to take place in the merge window). As such, they're only moves and should not affect the actual build. -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

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[alsa-devel] [PATCH] ALSA: hda - Add model quirk for Sony VAIO VGN-NR21E
by Ozan Caglayan 11 Mar '09

11 Mar '09

One of our users reported that its laptop doesn't mute the internal speakers when a jack is inserted into headphones without model=hippo. Here's the alsa-info output for reference: http://www.alsa-project.org/db/?f=8dae188ec2db110b9e6804c60f4659f0cb606c62 Signed-off-by: Ozan Caglayan <ozan(a)pardus.org.tr> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b794cba..6e79770 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10874,6 +10874,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), + SND_PCI_QUIRK(0x104d, 0x902d, "Sony VAIO VGN-NR21E", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", -- 1.6.2

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[alsa-devel] Accuracy of substream->ops->pointer?
by Timur Tabi 11 Mar '09

11 Mar '09

During capture, my hardware can take up to two milliseconds for the DMA to start transferring data to the buffer. This is because the internal FIFO needs to fill up before DMA starts. Today, my .trigger function (during a START request) just waits until the FIFO is full before returning control back to ALSA. This seems to work, except that I wait by using a busy-loop. As you can imagine, a 2ms busy loop is not a good idea. An alternative idea is to have my .pointer function return 0 if DMA has not yet started. This appears to work, but I want to make sure it's a good idea. My concern is that I might confuse ALSA because it is expecting some data in the buffer after capture has started. Is my approach correct? -- Timur Tabi Linux kernel developer at Freescale

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