August 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 2/2] ASoC: DaVinci: Add a DAI format to McASP driver
by Chaithrika U S 11 Aug '09

11 Aug '09

The patch adds a DAI format: Codec bit clock master and frame sync slave, to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch has been separated out from the first patch of this series. In the previous version, this was part of the McASP driver FIFO support patch. Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e672f43..eca22d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -443,6 +443,16 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 1.5.6

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[alsa-devel] hda-emu codec information files
by Ozan Çağlayan 11 Aug '09

11 Aug '09

Hi, I'm able to collect a lot of alsa-info from a bunch of netbook/laptop and desktop boards. I can send them as a patch series if it will help debug problems Thanks Ozan

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 11 Aug '09

11 Aug '09

The following changes since commit e0c48a18f76415d3ec1ac28e5f7a0df10e2f3d8c: Mark Brown (1): ASoC: Drop unneeded declaration of removed wm8731 SPI write function are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Daniel Ribeiro (1): ASoC: change set_tdm_slot api to allow slot_width override. Mark Brown (2): Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 ASoC: Update WM9081 for tdm_slot() API change Randy Dunlap (1): ASoC: fix I2C build errors include/sound/soc-dai.h | 5 ++- sound/soc/codecs/wm9081.c | 63 ++++++++++++++++++++++++++------------------ sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++----- sound/soc/soc-cache.c | 4 +- sound/soc/soc-core.c | 9 ++++-- 6 files changed, 69 insertions(+), 41 deletions(-)

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[alsa-devel] hard freezes with hdsp with cardbus/pci adapter card
by Tim Blechmann 11 Aug '09

11 Aug '09

hi all, i have some issues with playback on one of my machines with different kernels, having my rme hammerfall hdsp connected via a cardbus/pci adapter card. after some time (between a few seconds and several minutes) of flawless playback, the system completely freezes, magic sysrq keys are not functional either ... the freeze occurs with rt and non-rt kernels, with jack running with and without realtime scheduling. i have the suspicion, that it is related to the combination of the hdsp with the cardbus controller of the adapter card ... the interesting part of lspci is: 06:01.0 CardBus bridge: ENE Technology Inc CB1410 Cardbus Controller (rev 01) Subsystem: ENE Technology Inc CB1410 Cardbus Controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at fb600000 (32-bit, non-prefetchable) [size=4K] Bus: primary=06, secondary=07, subordinate=0a, sec-latency=176 Memory window 0: c4000000-c7fff000 (prefetchable) Memory window 1: c8000000-cbfff000 I/O window 0: 0000e000-0000e0ff I/O window 1: 0000e400-0000e4ff BridgeCtl: Parity+ SERR+ ISA- VGA- MAbort- >Reset- 16bInt- PostWrite+ 16-bit legacy interface ports at 0001 Kernel driver in use: yenta_cardbus Kernel modules: yenta_socket 07:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 0f) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255 Interrupt: pin A routed to IRQ 17 Region 0: Memory at c8000000 (32-bit, non-prefetchable) [size=64K] Kernel driver in use: RME Hammerfall DSP Kernel modules: snd-hdsp the irq 17 is used by these two devices only. i am not sure, how to proceed debugging this, since i cannot get any useful debugging information ... thnx, tim -- tim(a)klingt.org http://tim.klingt.org A paranoid is a man who knows a little of what's going on. William S. Burroughs

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[alsa-devel] (no subject)
by Chaithrika U S 11 Aug '09

11 Aug '09

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

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[alsa-devel] [PATCH 1/5] ASoC: DaVinci: i2s: don't bounce through rtd to get dai
by Troy Kisky 11 Aug '09

11 Aug '09

dai is a parameter to the functions, so use it instead of looking it up. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 14 +++++--------- 1 files changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e5cd97b..2a56fb7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,9 +353,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; @@ -425,8 +424,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, static int davinci_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) { @@ -439,8 +437,7 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int ret = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) @@ -466,8 +463,7 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); } -- 1.5.6.3

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[alsa-devel] Automatic mic-selection on Realtek and IDT/STAC codecs
by Takashi Iwai 11 Aug '09

11 Aug '09

Hi, recently I added the code for HD-audio Realtek and IDT/Sigmatel codecs to support the automatic capture-source selection via mic jack plugging. This feature is enabled for machines with one external and one internal mic, and only for auto-BIOS parser mode in the case of Realtek codecs. So, don't be surprised if you see neither "Capture Source" nor "Input Source" mixer elements after updates. Just plugging the mic jack will switch the input source from the internal to the external mic. If you encounter any problems regarding this change, please let me know. thanks, Takashi

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[alsa-devel] NULL pointer dereference in ALSA triggered through saa7134-alsa
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, I've finally succesfully compiled and linked saa7134-alsa driver using an external alsa-driver and its Module.symvers file. Everything seems okay, no undefined symbol or something else: - An installed 2.6.30.4 kernel which only builds and brings soundcore and sound_firmware, - Latest alsa-driver built externally and installed, - Latest saa7134-alsa, cx88-alsa, etc. code from linus-2.6 (seen that they don't affected by some API/ABI changes) patched on top of the alsa-driver tarball, I'm copying Module.symvers file generated after building the alsa-driver into each directory for getting rid of undefined snd_*() symbols. When I plug a saa7134 PCI card and reboot, the -alsa driver is automatically probed but it immediately oopses. I wonder if this could be related to my weird-but-apparently-successful compilation experience above? [ 6.876120] Linux video capture interface: v2.00 [ 7.089550] saa7130/34: v4l2 driver version 0.2.15 loaded [ 7.089612] saa7134 0000:03:05.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20 [ 7.089622] saa7133[0]: found at 0000:03:05.0, rev: 209, irq: 20, latency: 64, mmio: 0xfdbff000 [ 7.089631] saa7133[0]: subsystem: 1461:f11d, board: Avermedia PCI pure analog (M135A) [card=149,autodetected] [ 7.089659] saa7133[0]: board init: gpio is 40000 [ 7.089744] input: saa7134 IR (Avermedia PCI pure as /devices/pci0000:00/0000:00:14.4/0000:03:05.0/input/input6 [ 7.089789] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 7.230265] saa7133[0]: i2c eeprom 00: 61 14 1d f1 54 20 1c 00 43 43 a9 1c 55 d2 b2 92 [ 7.230274] saa7133[0]: i2c eeprom 10: ff ff ff ff ff 20 ff ff ff ff ff ff ff ff ff ff [ 7.230282] saa7133[0]: i2c eeprom 20: 01 40 01 32 32 01 01 43 88 ff 00 56 ff ff ff ff [ 7.230289] saa7133[0]: i2c eeprom 30: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230296] saa7133[0]: i2c eeprom 40: ff 22 00 c0 96 ff 03 30 15 00 ff ff ff ff ff ff [ 7.230302] saa7133[0]: i2c eeprom 50: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230309] saa7133[0]: i2c eeprom 60: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230316] saa7133[0]: i2c eeprom 70: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230323] saa7133[0]: i2c eeprom 80: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230330] saa7133[0]: i2c eeprom 90: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230337] saa7133[0]: i2c eeprom a0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230344] saa7133[0]: i2c eeprom b0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230350] saa7133[0]: i2c eeprom c0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230357] saa7133[0]: i2c eeprom d0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230364] saa7133[0]: i2c eeprom e0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230371] saa7133[0]: i2c eeprom f0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.412095] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 7.512683] hda_codec: ALC883: BIOS auto-probing. [ 7.512886] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:14.2/input/input7 [ 7.516862] HDA Intel 0000:01:05.2: PCI INT B -> GSI 19 (level, low) -> IRQ 19 [ 7.689107] tuner 1-004b: chip found @ 0x96 (saa7133[0]) [ 7.724034] tda829x 1-004b: setting tuner address to 60 [ 7.800011] tda829x 1-004b: type set to tda8290+75a [ 8.294455] usblp0: USB Bidirectional printer dev 2 if 0 alt 0 proto 2 vid 0x04E8 pid 0x325B [ 8.294500] usbcore: registered new interface driver usblp [ 10.814379] saa7133[0]: dsp access error [ 10.845083] saa7133[0]: registered device video0 [v4l2] [ 10.845111] saa7133[0]: registered device vbi0 [ 10.845130] saa7133[0]: registered device radio0 [ 10.950359] EXT3 FS on sda5, internal journal [ 10.996203] saa7134 ALSA driver for DMA sound loaded [ 10.996217] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 10.996247] saa7133[0]/alsa: saa7133[0] at 0xfdbff000 irq 20 registered as card -1 <-- /* That's interesting? */ [ 10.996349] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********** 1st null ptr deref [ 10.996962] IP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] *pde = 00000000· [ 10.997255] Oops: 0000 [#1] SMP· [ 10.997255] last sysfs file: /sys/module/saa7134/initstate [ 10.997255] Modules linked in: saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_seq_m idi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore snd_pa ge_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 10.997255]· [ 10.997255] Pid: 1141, comm: modprobe.bin Not tainted (2.6.30.4-126 #3) Unknow [ 10.997255] EIP: 0060:[<f8834cd4>] EFLAGS: 00010246 CPU: 1 [ 10.997255] EIP is at snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] EAX: 00000002 EBX: f65c0e0c ECX: f6b2fe98 EDX: 00000000 [ 10.997255] ESI: f65c0c00 EDI: 00000018 EBP: f6b2febc ESP: f6b2fe98 [ 10.997255] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 [ 10.997255] Process modprobe.bin (pid: 1141, ti=f6b2e000 task=f6a2caa0 task.ti=f6b2e000) [ 10.997255] Stack: [ 10.997255] 00000003 00000000 c030c2e9 f6b2feb8 f87ca221 00000000 f65c0e0c f6921400 [ 10.997255] 00000018 f6b2feec f882da8f 00000001 f6921438 f88358e4 436d6370 63304432 [ 10.997255] f65c0c00 f87cf9fe f69b42a0 f65c0d70 f65c0c00 f6b2fefc f87ce0bb f65c0e0c [ 10.997255] Call Trace: [ 10.997255] [<c030c2e9>] ? device_create_file+0x13/0x15 [ 10.997255] [<f87ca221>] ? snd_add_device_sysfs_file+0x67/0x6e [snd] [ 10.997255] [<f882da8f>] ? snd_pcm_dev_register+0x176/0x1de [snd_pcm] [ 10.997255] [<f87ce0bb>] ? snd_device_register_all+0x27/0x44 [snd] [ 10.997255] [<f87cadce>] ? snd_card_register+0x4c/0x131 [snd] [ 10.997255] [<f8d17687>] ? alsa_device_init+0x200/0x24b [saa7134_alsa] [ 10.997255] [<f8d1773d>] ? saa7134_alsa_init+0x6b/0x93 [saa7134_alsa] [ 10.997255] [<c0101137>] ? do_one_initcall+0x4a/0x115 [ 10.997255] [<f8d176d2>] ? saa7134_alsa_init+0x0/0x93 [saa7134_alsa] [ 10.997255] [<c0145205>] ? __blocking_notifier_call_chain+0x40/0x4c [ 10.997255] [<c015302d>] ? sys_init_module+0x87/0x18b [ 10.997255] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 11.066012] Code: 00 00 5b 5d c3 55 89 e5 57 56 53 89 c3 83 ec 18 8b 53 30 c7 45 e0 00 00 00 00 8d 4d dc c7 45 dc 03 00 00 00 8b 30 83 e2 01 8b 06 <8b> 00 89 45 e4 8b 46 0c 89 45 e8 8b 43 0c 01 c0 09 c2 8d 45 f0· [ 11.066012] EIP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] SS:ESP 0068:f6b2fe98 [ 11.066012] CR2: 0000000000000002 [ 11.066318] ---[ end trace 000b75f3b1903e42 ]--- [ 14.079154] r8169: eth0: link up [ 14.095472] r8169: eth0: link up [ 14.335980] r8169: eth0: link up [ 17.074440] NET: Registered protocol family 10 [ 17.074879] lo: Disabled Privacy Extensions [ 20.520592] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********* 2nd nullptr deref [ 20.535353] IP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.535353] *pde = 00000000· [ 20.559932] Oops: 0000 [#2] SMP· [ 20.559932] last sysfs file: /sys/devices/pci0000:00/0000:00:14.4/0000:03:05.0/sound/card2/pcmC2D0c/pcm_class [ 20.586971] Modules linked in: ipv6 saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_ seq_midi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore s nd_page_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 20.606741]· [ 20.606741] Pid: 1367, comm: hald Tainted: G D (2.6.30.4-126 #3) Unknow[ 20.704561] Process hald (pid: 1367, ti=f660e000 task=f6420c70 task.ti=f660e000) [ 20.704561] Stack: [ 20.704561] f65c0e0c f65c0c00 f6ada3f0 f642fe00 f6919ac0 f660fe34 f882e15a f65c0e0c [ 20.704561] 00000200 f6ada3f0 f6ada3c0 f6667000 f660fe3c f882e254 f660fe70 f87cb9e9 [ 20.704561] f680db00 0160fe70 ffffffff f64cb870 f6919ac0 00001000 00000020 00000020 [ 20.704561] Call Trace: [ 20.704561] [<f882e15a>] ? snd_pcm_proc_info_read+0x80/0x16c [snd_pcm] [ 20.704561] [<f882e254>] ? snd_pcm_stream_proc_info_read+0xe/0x10 [snd_pcm] [ 20.704561] [<f87cb9e9>] ? snd_info_entry_open+0x26b/0x2de [snd] [ 20.704561] [<c01df157>] ? proc_reg_open+0xb6/0x122 [ 20.704561] [<f87cb6ca>] ? snd_info_entry_release+0x0/0xa7 [snd] [ 20.704561] [<c01ac441>] ? __dentry_open+0x119/0x207 [ 20.704561] [<c01ac5c9>] ? nameidata_to_filp+0x2c/0x43 [ 20.704561] [<c01df0a1>] ? proc_reg_open+0x0/0x122 [ 20.704561] [<c01b656b>] ? do_filp_open+0x3ae/0x6c8 [ 20.704561] [<c019ff47>] ? free_pages_and_swap_cache+0x44/0x57 [ 20.704561] [<c01b6937>] ? getname+0x20/0xb7 [ 20.704561] [<c01be23c>] ? alloc_fd+0x55/0xbe [ 20.704561] [<c01ac253>] ? do_sys_open+0x44/0xba [ 20.704561] [<c01ac30b>] ? sys_open+0x1e/0x26 [ 20.704561] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 20.704561] Code: 00 00 00 89 e5 57 56 53 89 d3 83 ec 08 89 df 89 45 ec 8b 00 8b 55 ec 89 45 f0 31 c0 8b 72 04 f3 ab 8b 55 f0 b9 40 00 00 00 8b 02 <8b> 00 89 43 0c 8b 42 0c 8b 55 ec 89 03 8b 42 30 89 43 08 8b 42· [ 20.704561] EIP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] SS:ESP 0068:f660fe04 [ 21.199052] CR2: 0000000000000002 [ 21.222000] ---[ end trace 000b75f3b1903e43 ]--- [ 20.606741] EIP: 0060:[<f8830abf>] EFLAGS: 00010246 CPU: 1 [ 20.704561] EIP is at snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.704561] EAX: 00000002 EBX: f642fe00 ECX: 00000040 EDX: f65c0c00 [ 20.704561] ESI: 00000000 EDI: f642ff20 EBP: f660fe18 ESP: f660fe04 [ 20.704561] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 The first dereference is happening at: (gdb) list *snd_pcm_timer_init+0x26 0x7cf8 is in snd_pcm_timer_init (/var/pisi/module-alsa-driver-1.0.20_20090808-49/work/alsa-driver/acore/../alsa-kernel/core/pcm_timer.c:131). 126 struct snd_timer_id tid; 127 struct snd_timer *timer; 128 129 tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; 130 tid.dev_class = SNDRV_TIMER_CLASS_PCM; 131 tid.card = substream->pcm->card->number; <---------- 132 tid.device = substream->pcm->device; 133 tid.subdevice = (substream->number << 1) | (substream->stream & 1); 134 if (snd_timer_new(substream->pcm->card, "PCM", &tid, &timer) < 0) 135 return;at the following place: Can it be related to a race condition between the sound card and the saa7134-alsa interface?

2 7

[alsa-devel] issue at 44.1 Khz capturing
by Harsha, Priya 10 Aug '09

10 Aug '09

Hi, I am using arecord for capturing data. When I try to record at 8 or 48 KHz, it's perfectly fine. But when I try to record at 44.1 Khz, there are spikes in the recorded data for every buffer. When I tried to tap the data captured at the firmware/hardware end, it's smooth but the same data when is checked from the user space after being recorded, it has spikes. On further analysis, looks like for every buffer recorded there is either 1 sample recorded extra or 1 sample dropped. But when I printed out the values passed from the pointer function, they are exactly at the intervals of 5513 bytes. Can anyone give me any clue as to what is going wrong? I am using indirect_* functions in the driver code. Is there any buffer size round off that is giving this error? Can I handle something in the driver to avoid this? Thanks, Harsha.

3 4

[alsa-devel] Implementing sync start
by Jon Smirl 10 Aug '09

10 Aug '09

I'm implementing sync start as Takashi described... --------------- If the hardware supports the simultaneous triggering of multiple streams, add the sync support to the driver. Namely, - Add SNDRV_PCM_INFO_SYNC_START flag to PCM info field - Call snd_pcm_set_sync(substream) at each open callback - In the trigger callback, check all bundled streams via snd_pcm_group_for_each_entry(). Mark all the streams but the current one via snd_pcm_trigger_done(). Then start all streams at once. Ditto for stop and pause. ------------- Is there a simple way to test this with something like a mono sine wav file? A aplay mono.wav and then use and ALSA conf to copy the sine to all channels and then sync start them. ----------- brutefit has this routine for sync start, does this look correct? int bfio_synch_start(void) { snd_pcm_status_t *status; int err, n; if (base_handle == NULL) { return 0; } /* FIXME: the SND_PCM_STATE_RUNNING code would not be needed if the bfio_write autostart hack was not there */ snd_pcm_status_alloca(&status); if (link_handles) { if ((err = snd_pcm_status(base_handle, status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { return 0; } if ((err = snd_pcm_start(base_handle)) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } return 0; } FOR_IN_AND_OUT { for (n = 0; n < n_handles[IO]; n++) { if ((err = snd_pcm_status(handles[IO][n], status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { continue; } if ((err = snd_pcm_start(handles[IO][n])) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } } } return 0; } -- Jon Smirl jonsmirl(a)gmail.com

3 12

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 10 Aug '09

10 Aug '09

The following changes since commit b7b8f9bf0cd73b90561e8123fd5ec28f4539c419: Janusz Krzysztofik (1): TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Chaithrika U S (1): ASoC: DaVinci: Support Audio on DA830 EVM Mark Brown (3): ASoC: Convert WM8776 to use factored out register cache code ASoC: Define more formats for the AC97 CODECs ASoC: Drop unneeded declaration of removed wm8731 SPI write function Uwe Kleine-König (1): ASoC: s3c2443-ac97: convert semaphore to mutex include/sound/soc-dai.h | 8 ++++- sound/soc/codecs/wm8731.c | 3 -- sound/soc/codecs/wm8776.c | 59 +++++++------------------------------ sound/soc/davinci/Kconfig | 11 +++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++ sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++--- 7 files changed, 68 insertions(+), 57 deletions(-)

2 1

[alsa-devel] load_for_all_cards bug
by Kevin Goodsell 10 Aug '09

10 Aug '09

(I originally tried to send this via the moderator without subscribing to the list, but it never went through and I never received a moderator response. Apparently this list is set up as moderated, but nobody is actually moderating it.) Summary: load_for_all_cards fails with existing configuration elements In snd_config_hook_load_for_all_cards, the first call to snd_config_search attempts to locate an existing configuration node with the name of the driver. Typically none is found, and everything is good. However, if such a node is located, the next line assumes it is a leaf node with type 'string' and calls snd_config_get_string to fetch the string value. If this fails, the entire hook is abandoned. Because of this, setting something like the following in asoundrc: cards.<driver name>.foo 0 is sufficient to disable the entire card-specific configuration. As a concrete example, I have a HDA-Intel sound card. dmix.conf includes a way to set period_size, period_time, and periods by using configuration elements of the form cards.<driver name>.pcm.dmix.<var>. In ~/.asoundrc I add cards.HDA-Intel.pcm.dmix.period_size 1024 This will cause HDA-Intel.conf to fail to load, and the pcm defined in default.conf will fail to find the device-specific pcm cards.HDA-Intel.pcm.default, and fall back on the default pcm using plughw. By attempting to configure dmix, I have disabled it. Fixing this may just be a matter of changing the line in snd_config_hook_load_for_all_cards from: if (snd_config_search(root, fdriver, &n) >= 0) { to: if (snd_config_search(root, fdriver, &n) >= 0 && n->type != SND_CONFIG_TYPE_COMPOUND) { I have not tested this. -Kevin

2 1

[alsa-devel] [RFC][PATCH 0/3] Enhance ASoC on OMAP1510 by implementing DMA channel linking
by Janusz Krzysztofik 10 Aug '09

10 Aug '09

Current implementation of ASoC OMAP PCM makes use of DMA channel self linking feature for all machines but OMAP1510. This is caused by channel linking not supported on OMAP1510 in current OMAP DMA implementation. Instead, DMA interrputs are counted as periods, and after all periods in a buffer has been detected, DMA transfer is started again for a new buffer content. The period counter is also used as a software replacement for a hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. In my attempts to debug random PCM hangups, I was wondering what could happen if a single interrupt was missing, ie. the period index was not incremented for a period. It seems that the transfer would just hang after the current buffer processed, as DMA would not be started again, waiting for a missing period. Furthermore, software calculated PCM pointer accuracy can suffer from missing DMA interrupts. Even if OMAP1510 DMA hardware does not support channel linking in a way like newer models do, it has some other features that could be used for the purpose. Interrupt status flags, passed to OMAP PCM interrupt handler routine, can be used to get more accurate DMA progress counter software calculation. Even if these enhancements did not help me in eliminating random PCM hangups completely, I find them worth of sharing. There are three patches in the series: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 [RFC][PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

1 0

[alsa-devel] alsamixer -a basic
by Elimar Riesebieter 10 Aug '09

10 Aug '09

alsa-lib 1.0.20 running alsamixer -a basic gives: ALSA lib python.c:991:(alsa_mixer_simple_finit) Unable to find python module '/usr/lib/alsa-lib/smixer/python/main.py' How do I build smixer/python/main.py ? Elimar -- On the keyboard of life you have always to keep a finger at the escape key;-)

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#539454: libasound2: Division by zero in pcm_rate.c when period_size->min == 0
by Elimar Riesebieter 10 Aug '09

10 Aug '09

forwarded 539454 alsa-devel(a)alsa-project.org tags 539454 + patch thanks Hi, this is Debian bug 539454. It would be better to fix it in general. Patching the Debian Package isn't a reasonable solution. Thanks Elimar * John Lindgren [090731 22:29 -0400] > Package: libasound2 > Version: 1.0.20-2 > Severity: important > Tags: patch > > > System: Dell Inspiron 1200 > Card: Intel ICH6 > Chip: SigmaTel STAC9752,53 > > > Steps to reproduce: > > 1. Disable dmix: mv /usr/share/alsa/cards/ICH4.conf /usr/share/alsa/cards/ICH4.conf.0 > 2. Restart ALSA: /etc/rcS.d/S50alsa-utils restart > 3. Clear Audacity configuration: rm -r ~/.audacity* > 4. Run Audacity 1.3.7-3 in Valgrind: valgrind --tool=none audacity > 5. Choose "OK" in the "Audacity First Run" dialog. > > > Valgrind output: > > ==26237== Process terminating with default action of signal 8 (SIGFPE) > ==26237== Integer divide by zero at address 0x64D1CD56 > ==26237== at 0x4CAD600: snd_pcm_rate_hw_refine_cchange (pcm_rate.c:189) > ==26237== by 0x4C97B2E: snd1_pcm_hw_refine_slave (pcm_params.c:2182) > ==26237== by 0x4CAD7D5: snd_pcm_rate_hw_refine (pcm_rate.c:222) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4CA116B: snd1_pcm_generic_hw_refine (pcm_generic.c:98) > ==26237== by 0x4C97B0D: snd1_pcm_hw_refine_slave (pcm_params.c:2158) > ==26237== by 0x4CC0E25: snd_pcm_lfloat_hw_refine (pcm_lfloat.c:261) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4C99068: sndrv_pcm_hw_params (pcm_params.c:2290) > ==26237== by 0x4CB209C: snd_pcm_plug_hw_params (pcm_plug.c:1045) > ==26237== by 0x4C9921D: sndrv_pcm_hw_params (pcm_params.c:2299) > ==26237== by 0x4C947B0: snd_pcm_hw_params (pcm.c:822) > > > Diagnosis: > > pcm_rate.c:189 reads: > > if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > When using the ICH4 driver without dmix, period_size->min == 0. Handle > this by changing pcm_rate.c:189 to: > > if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > With this change, Audacity starts without problems. Patch against native 1.0.20: diff -Naurd a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c --- a/src/pcm/pcm_rate.c 2009-05-06 09:07:23.000000000 +0200 +++ b/src/pcm/pcm_rate.c 2009-08-06 21:34:23.000000000 +0200 @@ -186,7 +186,7 @@ if (!snd_interval_checkempty(period_size) && period_size->openmin && period_size->openmax && period_size->min + 1 == period_size->max) { - if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { snd_interval_set_value(period_size, period_size->min); } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { snd_interval_set_value(period_size, period_size->max); -- Experience is something you don't get until just after you need it!

2 1

[alsa-devel] [BUG] NULL pointer dereference in patch_sigmatel.c
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, One of our users is having a NULL ptr dereference upon loading the snd_hda_intel module with 20090624's snapshot. There's only one commit after that date in patch_sigmatel.c so I didn't tell him to try with the latest snapshot but if you think that the bug may be related to another part of the ALSA codebase, I can make him try the latest snapshot. I'm attaching the alsa-info output on a working system with 1.0.18a. The kernel version in subject is 2.6.25.20. Thanks, Ozan ------------------------ First the BUG's trace: ------------------------ BUG: unable to handle kernel NULL pointer dereference at 00000074 IP: [<f8cdb83a>] :snd_hda_codec_idt:stac92xx_add_jack+0x84/0x9e *pde = 00000000 Oops: 0002 [#1] SMP Modules linked in: snd_hda_codec_idt snd_hda_intel(+) snd_hda_codec aes_i586 aes_generic ipv6 af_packet bridge bnep rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace cpufreq_conservative ndiswrapper vboxdrv pcmcia snd_hwdep rtc_cmos rtc_core yenta_socket tpm_infineon snd_seq_dummy rtc_lib tpm snd_seq_oss rsrc_nonstatic tpm_bios tifm_7xx1 iTCO_wdt snd_seq_midi_event tifm_core i2c_i801 snd_seq iTCO_vendor_support pcmcia_core r5u870 arc4 joydev snd_seq_device snd_pcm_oss nvidia(P) battery sg usbcam snd_mixer_oss ecb ac videobuf_dma_sg videobuf_core sony_laptop snd_pcm uvcvideo iwl4965 video compat_ioctl32 snd_timer output hci_usb iwlcore videodev thermal bluetooth snd processor v4l1_compat rfkill button led_class soundcore firmware_class snd_page_alloc mac80211 intel_agp i2c_core sky2 cfg80211 agpgart ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix uhci_hcd pata_acpi ehci_hcd usbcore ohci1394 ieee1394 ata_generic libata scsi_mod dock Pid: 1977, comm: modprobe Tainted: P (2.6.25.20-114 #1) EIP: 0060:[<f8cdb83a>] EFLAGS: 00210206 CPU: 0 EIP is at stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] EAX: 00000000 EBX: f8c7973e ECX: 00000004 EDX: f8cdeb1a ESI: 03211020 EDI: f675c600 EBP: f68d7d20 ESP: f68d7cd4 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process modprobe (pid: 1977, ti=f68d6000 task=f69ece60 task.ti=f68d6000) Stack: f68d7cf4 f8cdeb0a f8c796d5 f8c7973e f8c79793 00000001 000a69e0 f8c79793 4f205048 61207475 78452074 654c2074 4a207466 006b6361 f6898800 f6898800 00000000 f6898800 f6898800 f68d7d4c f8cdbac0 f65ef000 f6970000 00000001 Call Trace: [<f8cdbac0>] ? stac92xx_build_controls+0x26c/0x2d0 [snd_hda_codec_idt] [<f8c73aa9>] ? snd_hda_codec_build_controls+0x31/0x3d [snd_hda_codec] [<f8c750d3>] ? snd_hda_build_controls+0x18/0x67 [snd_hda_codec] [<f8c69138>] ? azx_probe+0x79e/0x840 [snd_hda_intel] [<f8c68857>] ? azx_send_cmd+0x0/0x101 [snd_hda_intel] [<f8c686a6>] ? azx_get_response+0x0/0x1b1 [snd_hda_intel] [<f8c67ece>] ? azx_attach_pcm_stream+0x0/0x15c [snd_hda_intel] [<f8c67b84>] ? azx_bus_reset+0x0/0x56 [snd_hda_intel] [<f8c67a40>] ? azx_power_notify+0x0/0x57 [snd_hda_intel] [<c01e7a37>] ? pci_device_probe+0x39/0x59 [<c024395f>] ? driver_probe_device+0xa0/0x136 [<c0243a50>] ? __driver_attach+0x5b/0x91 [<c024333c>] ? bus_for_each_dev+0x3b/0x63 [<c0243804>] ? driver_attach+0x14/0x16 [<c02439f5>] ? __driver_attach+0x0/0x91 [<c0242d3a>] ? bus_add_driver+0x9d/0x1ba [<c0243bc4>] ? driver_register+0x47/0xa7 [<c0168681>] ? __vunmap+0x93/0x9b [<c01e7bec>] ? __pci_register_driver+0x35/0x61 [<f8afd017>] ? alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [<c0141f9c>] ? sys_init_module+0x18ad/0x19ca [<c0109c77>] ? do_syscall_trace+0x138/0x17f [<c0104a2e>] ? syscall_call+0x7/0xb ======================= Code: f9 ff 89 45 d0 89 f0 e8 29 68 f9 ff 89 c3 89 f0 e8 32 68 f9 ff ff 75 d0 53 50 68 0a eb cd f8 8d 45 d4 50 e8 45 25 50 c7 8b 47 08 <89> 78 74 8b 47 08 83 c4 14 c7 40 78 7f a2 cd f8 31 c0 8d 65 f4 EIP: [<f8cdb83a>] stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] SS:ESP 0068:f68d7cd4 ---[ end trace 4a628a5b4fc302e7 ]--- alsa-info output: ------------------- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Sun Feb 8 19:15:27 EET 2009 !!Linux Distribution !!------------------ Pardus 2008.2 Canis aureus !!Kernel Information !!------------------ Kernel release: 2.6.25.20-114 Operating System: GNU/Linux Architecture: i686 Processor: Intel(R) Core(TM)2 Duo CPU T8100 @ 2.10GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18a Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 21 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) 09:04.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 104d:9008 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=vaio !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : vaio,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d1e00 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=5, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x07 Processing caps: benign=0, ncoeff=0 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x0d 0x0d] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x80: HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03a15030: [Jack] Mic at Ext Left Conn = 1/8, Color = Red DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Delay: 4 samples Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d 0x14* 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=0, val=127 Connection: 4 0x02* 0x03 0x04 0x05 Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Delay: 4 samples Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Şub 8 2009 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Şub 8 2009 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Şub 8 2009 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Şub 8 19:14 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Şub 8 19:15 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Şub 8 2009 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Şub 8 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # PulseAudio plugin configuration # Let's create a virtual device "pulse" for mixer and PCM pcm.pulse { type pulse hint { description "PulseAudio Sound Server" } } ctl.pulse { type pulse hint { description "PulseAudio Sound Server" } } # Let's make it the default! pcm.!default { type pulse hint { description "Default" } } ctl.!default { type pulse hint { description "Default" } } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfc300000 irq 21' Mixer name : 'SigmaTel STAC9872AK' Components : 'HDA:83847662,104d1e00,00100201 HDA:14f12c06,104d1700,00100000' Controls : 10 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 255 Mono: Capture [off] Front Left: Playback 0 [0%] [-51.00dB] Front Right: Playback 0 [0%] [-51.00dB] Simple mixer control 'Mic Jack',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 13 [87%] [19.50dB] [on] Front Right: Capture 13 [87%] [19.50dB] [on] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 127 value.1 127 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Speaker Playback Volume' value.0 127 value.1 127 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 13 value.1 13 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Mic Jack' comment.item.1 'Internal Mic' comment.item.2 PCM iface MIXER name 'Capture Source' value 'Internal Mic' } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 127 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.10 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module aes_i586 aes_generic ipv6 bridge bnep af_packet rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace ndiswrapper r5u870 usbcam videobuf_dma_sg snd_hda_intel snd_seq_dummy snd_seq_oss snd_seq_midi_event videobuf_core nvidia snd_seq uvcvideo snd_seq_device snd_pcm_oss arc4 compat_ioctl32 ecb snd_mixer_oss intel_agp snd_pcm iwl4965 snd_timer snd_page_alloc snd_hwdep snd pcmcia hci_usb rtc_cmos bluetooth yenta_socket rsrc_nonstatic tpm_infineon pcmcia_core tifm_7xx1 sky2 battery videodev iTCO_wdt agpgart ac thermal processor iwlcore rfkill led_class firmware_class i2c_i801 mac80211 video output tpm tpm_bios rtc_core rtc_lib joydev iTCO_vendor_support tifm_core sony_laptop v4l1_compat i2c_core sg button soundcore cfg80211 ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix ohci1394 ieee1394 uhci_hcd pata_acpi ata_generic libata scsi_mod dock ehci_hcd usbcore

3 24

[alsa-devel] [PATCH 1/2] ASoC: DaVinci: McASP driver enhacements
by Chaithrika U S 10 Aug '09

10 Aug '09

On DA830/OMAPL137 SoC, the McASP peripheral has internal FIFO support. It provides additional data buffering. It also provides tolerance to vairiation in host/DMA controller response times. The read and write FIFO sizes are 256 bytes each. If FIFO is enabled, the DMA events from McASP are sent to the FIFO which inturn sends DMA requests to the host CPU according to the thresholds programmed. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber= sprufm1&fileType=pdf This patch adds support for FIFO configuration.The platform member has a version field which differentiates the McASP on different SoCs. The patch also adds another DAI format to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 113 ++++++++++++++++++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 5 ++ sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + 4 files changed, 115 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f0c0347..eca22d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -102,6 +102,11 @@ /* Receive Buffer for Serializer n */ #define DAVINCI_MCASP_RXBUF_REG 0x280 +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -276,6 +281,13 @@ */ #define TXDATADMADIS BIT(0) +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + #define DAVINCI_MCASP_NUM_SERIALIZER 16 static inline void mcasp_set_bits(void __iomem *reg, u32 val) @@ -345,6 +357,9 @@ static void mcasp_start_rx(struct davinci_audio_dev *dev) static void mcasp_start_tx(struct davinci_audio_dev *dev) { + u8 offset = 0, i; + u32 cnt; + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); @@ -353,6 +368,19 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); } @@ -362,6 +390,13 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) mcasp_start_tx(dev); else mcasp_start_rx(dev); + + /* enable FIFO */ + if (dev->txnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -382,6 +417,13 @@ static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) mcasp_stop_tx(dev); else mcasp_stop_rx(dev); + + /* disable FIFO */ + if (dev->txnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -401,7 +443,16 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -505,6 +556,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) { int i; + u8 tx_ser = 0; + u8 rx_ser = 0; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -525,12 +578,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) + if (dev->serial_dir[i] == TX_MODE) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); - else if (dev->serial_dir[i] == RX_MODE) + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } @@ -543,6 +621,8 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ @@ -622,8 +702,16 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; int word_length; + u8 numevt; davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + numevt = dev->txnumevt; + else + numevt = dev->rxnumevt; + + if (!numevt) + numevt = 1; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -650,6 +738,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); return -EINVAL; } + + if (dev->version == MCASP_VERSION_2) { + dma_params->data_type *= numevt; + dma_params->acnt = 4 * numevt; + } else + dma_params->acnt = dma_params->data_type; + davinci_config_channel_size(dev, word_length); return 0; @@ -778,6 +873,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; dma_data[count].name = "I2S PCM Stereo out"; dma_data[count].eventq_no = pdata->eventq_no; @@ -807,9 +905,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } dma_data[count].channel = res->start; - davinci_mcasp_dai[pdev->id].private_data = dev; - davinci_mcasp_dai[pdev->id].dev = &pdev->dev; - ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) goto err_release_region; @@ -827,12 +925,13 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; - snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); - dev = davinci_mcasp_dai[pdev->id].private_data; + snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]); + dev = davinci_mcasp_dai[pdata->op_mode].private_data; clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 36b7104..554354c 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -50,6 +50,11 @@ struct davinci_audio_dev { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; }; #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 8fd0c3c..1e2dfd0 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; unsigned int data_type; + unsigned short acnt; unsigned int count; period_size = snd_pcm_lib_period_bytes(substream); @@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } + acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); edma_set_src_index(lch, src_bidx, 0); edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index eb4287f..63d9625 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -19,6 +19,7 @@ struct davinci_pcm_dma_params { char *name; /* stream identifier */ int channel; /* sync dma channel ID */ + unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ -- 1.5.6

4 4

[alsa-devel] ASoC: PCM formats for AC97
by Jon Smirl 09 Aug '09

09 Aug '09

Tell me if this is right and I'll do a patch.... ----------------------------------------------------------- The current mpc5200 code has: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S32_BE, I think I have this backwards, the ac97 driver can take 8/16/24/32 it's the Bestcomm program that doesn't know about anything except 32b. So if we wrote new Bestcomm programs that expanded bytes from memory out into 32b DMA register writes, we could support 8/16b formats. The correct formats would be: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, After implementing new Bestcomm programs.... static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, ----------------------------------------------------------- A related issue in soc-dai.h: #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) shouldn't this be #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16 |\ SNDRV_PCM_FMTBIT_S32) The first form says the AC97 hardware can take both endians, but it doesn't take both endians it takes the endian matching the machine it is attached to. Don't the standard AC97 formats also include 8 and 24b? -- Jon Smirl jonsmirl(a)gmail.com

3 6

[alsa-devel] Testers wanted: New SB X-Fi driver
by Frashman 09 Aug '09

09 Aug '09

Hi Takashi, first of all, thank you for developing this driver. I tested it today. It works, so far. But two annoying things. As you mentioned, I got problems, if the onboard sound modules are loaded. Then i get no sound out auf my xfi. If I disable the loading of snd-hda-intel modules while booting, then it works. The second problem is, that SIDE and SURROUND connectors are interchanged. I'm absolutly sure, it works fine under windows and the connector colors are also correct. If I run speaker-test -c8 the SIDE Speakers sounds come out of my REAR (SURROUND) Speakers, while nothing is coming out of my rear speakers, if i'm using speaker-test -c6 Here lspci: $ lspci 00:00.0 Host bridge: ATI Technologies Inc RD780 Northbridge only dual slot PCI-e_GFX and HT1 K8 part 00:02.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (external gfx0 port A) 00:04.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port A) 00:0a.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port F) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:00.0 VGA compatible controller: ATI Technologies Inc RV770 [Radeon HD 4850] 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 02:00.0 Audio device: Creative Labs X-Fi Titanium series [EMU20k2] (rev 03) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) 04:06.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 04:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 04:0e.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23 IEEE-1394a-2000 Controller (PHY/Link) Thank you so far, Frashman

7 33

[alsa-devel] Auzentech X-Fi Prelude with alsa drivers
by Adrian Patzak 09 Aug '09

09 Aug '09

Dear sirs, I've got a Auzentech X-Fi Prelude and I installed the alsa unstable snapshot drivers. Stereo sound for my headphones and ditital output are working fine but my microphone isn't recording anything. I was checking the mixer settings but i wasn't able to find something wrong. Are the x-fi drivers just not designed to fit with the Auzentech card or do i have a local configuration problem? The output of alsa-info.sh script is attached. Thanking you in anticipation for any answer. upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Fri Aug 7 10:48:57 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!DMI Information !!--------------- Manufacturer: FUJITSU SIEMENS Product Name: AMD690VM-FMH !!Kernel Information !!------------------ Kernel release: 2.6.28-14-generic Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_ctxfi !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [XFi ]: SB-XFi - Creative X-Fi Creative X-Fi 20K1 Unknown !!PCI Soundcards installed in the system !!-------------------------------------- 03:01.0 Multimedia audio controller: Creative Labs SB X-Fi !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 03:01.0 0401: 1102:0005 Subsystem: 415a:0034 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_ctxfi enable : Y,Y,Y,Y,Y,Y,Y,Y id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 multiple : 2 reference_rate : 48000 use_system_timer : N !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 7 08:13 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 24 Aug 7 12:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 7 12:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 17 Aug 7 08:13 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 18 Aug 7 08:13 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 19 Aug 7 08:13 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 20 Aug 7 08:13 /dev/snd/pcmC0D4p crw-rw----+ 1 root audio 116, 33 Aug 7 08:13 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY E: core-util.c: Home directory /home/adrian not ours. **** List of PLAYBACK Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 1: ctxfi [Surround] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 2: ctxfi [Center/LFE] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 3: ctxfi [Side] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 4: ctxfi [IEC958 Non-audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD E: core-util.c: Home directory /home/adrian not ours. **** List of CAPTURE Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [XFi] Card hw:0 'XFi'/'Creative X-Fi 20K1 Unknown' Mixer name : '20K1' Components : '' Controls : 30 Simple ctrls : 11 Simple mixer control 'Master',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Simple mixer control 'PCM',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Front',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Center/LFE',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Side',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Line-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Simple mixer control 'Digit-IO',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'S/PDIF-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'S/PDIF-out',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.XFi { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 256 value.1 256 } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 256 value.1 256 } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Playback Volume' value.0 256 value.1 256 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Playback Volume' value.0 256 value.1 256 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Playback Volume' value.0 256 value.1 256 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-out Playback Volume' value.0 256 value.1 256 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 256 value.1 256 } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 256 value.1 256 } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Center/LFE Playback Volume' value.0 256 value.1 256 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 256 value.1 256 } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Capture Volume' value.0 256 value.1 256 } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Capture Volume' value.0 256 value.1 256 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Capture Volume' value.0 256 value.1 256 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Capture Volume' value.0 256 value.1 256 } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Capture Volume' value.0 256 value.1 256 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Capture Switch' value false } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Capture Switch' value false } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Capture Switch' value true } control.19 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Capture Switch' value false } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Playback Switch' value false } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-out Playback Switch' value true } control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Playback Switch' value false } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Front Playback Switch' value true } control.24 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Surround Playback Switch' value false } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center/LFE Playback Switch' value false } control.26 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Side Playback Switch' value false } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Digit-IO Playback Switch' value true } control.28 { comment.access read comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Mask' value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.29 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.30 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback PCM Stream' value '0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat binfmt_misc ppdev bridge stp bnep video output input_polldev sbp2 lp parport joydev snd_ctxfi snd_pcm snd_timer snd usbhid i2c_piix4 soundcore snd_page_alloc psmouse serio_raw usblp pcspkr nvidia usb_storage ohci1394 ieee1394 r8169 mii fbcon tileblit font bitblit softcursor !!ALSA/HDA dmesg !!------------------

1 0

[alsa-devel] Problems with gain controls
by James Courtier-Dutton 09 Aug '09

09 Aug '09

Hi, Does anyone know how to disable pulse from making any changes to the alsa hardware mixer controls? I would prefer pulse to use software gain controls, and not touch the alsamixer levels. pulse seems to do a very bizarre thing. The single master gain control within pulse changes both the alsamixer PCM and Master controls. Kind Regards James

2 3

[alsa-devel] Typo in soc/atmel/sam9g20_wm8731.c
by Pedro I. Sanchez 08 Aug '09

08 Aug '09

Hello, This is a patch to fix what I believe is a typo in this file. --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 173a239..baa42f3 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -341,7 +341,7 @@ static int __init at91sam9g20ek_init(void) } pllb = clk_get(NULL, "pllb"); - if (IS_ERR(mclk)) { + if (IS_ERR(pllb)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk;

1 0

[alsa-devel] [PATCH 2/2] ASoC: DaVinci: Support Audio on DA830 EVM
by Chaithrika U S 08 Aug '09

08 Aug '09

Add support for audio on DA830 EVM- here McASP1 is interfaced to TLV320AIC3106 codec. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/Kconfig | 11 +++++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++++++++++++++ 3 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..677a538 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -41,3 +41,14 @@ config SND_DAVINCI_SOC_SFFSDR help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAPL137 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + + help + Say Y if you want to add support for SoC audio on TI + DA830/OMAPL137 EVM + diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..5e2195f 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a5a9212 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,6 +54,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_davinci_da830_evm()) + sysclk = 24576000; + else return -EINVAL; @@ -162,6 +165,14 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { @@ -179,6 +190,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +static struct snd_soc_card da830_snd_soc_card = { + .name = "DA830 EVM", + .dai_link = &da830_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +209,11 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +static struct aic3x_setup_data da830_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -205,6 +228,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { .codec_data = &dm6467_evm_aic3x_setup, }; +/* evm audio subsystem */ +static struct snd_soc_device da830_evm_snd_devdata = { + .card = &da830_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da830_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +252,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + } else if (machine_is_davinci_da830_evm()) { + evm_snd_dev_data = &da830_evm_snd_devdata; + index = 1; } else return -EINVAL; -- 1.5.6

2 1

[alsa-devel] [PATCH 2/6] s3c2443-ac97: convert semaphore to mutex
by Uwe Kleine-König 08 Aug '09

08 Aug '09

This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig): CC [M] sound/soc/s3c24xx/s3c2443-ac97.o sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX' sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read': sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function) sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.) sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write': sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Timur Tabi <timur(a)freescale.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5d..bf16f20 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
by Candelaria Villareal, Jorge 07 Aug '09

07 Aug '09

Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6 line, controlled by register INTBR_PMBR1. Machine driver takes care of enabling gpio line through i2c and codec driver manipulates the line during headset ramp up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c51594d..f7e5b74 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,9 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +#define TWL4030_INTBR_PMBR1 0x0D +#define EXTMUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -280,6 +284,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -312,6 +317,10 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), + TWL4030_MODULE_INTBR); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- 1.6.0.4

3 6