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[alsa-devel] [PATCH] alsa: hdsp - allow proc reporting with disconnected io box
by Tim Blechmann 12 Aug '09

12 Aug '09

the hdsp driver refuses to report any information via the proc interface, if the io box is not connected. with this patch, the content of the control and status registers is printed before the iobox check. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 37 ++++++++++++++++++++----------------- 1 files changed, 20 insertions(+), 17 deletions(-)

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[alsa-devel] [PATCH] alsa-info: Use sysfs if available instead of dmidecode
by Ozan Çağlayan 12 Aug '09

12 Aug '09

Using sysfs for acquiring DMI data requires no root privileges. Use it if available instead of dmidecode. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 0ab1ddf..8f401c0 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -417,7 +417,12 @@ ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) -if [ -x $DMIDECODE ]; then +#Check for DMI data +if [ -d /sys/class/dmi/id ]; then + # No root privileges are required when using sysfs method + DMI_SYSTEM_MANUFACTURER=$(cat /sys/class/dmi/id/sys_vendor 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$(cat /sys/class/dmi/id/product_name 2>/dev/null) +elif [ -x $DMIDECODE ]; then DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) fi -- 1.6.3.4

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[alsa-devel] [PATCH] add output/input widgets in ad1938 to make dac/adc dynamic PM work
by Barry Song 12 Aug '09

12 Aug '09

According to the function dapm_dac_check_power() in sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any output widget as sink. And according to dapm_adc_check_power(), adc power can't be on/off stand-alone without any input widget as source. So we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC to hope their power can be managed dynamically. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index aafda74..9b8cc5c 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -99,11 +99,23 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), }; static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, }; /* -- 1.5.6.3

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[alsa-devel] [PATCHv2 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 12 Aug '09

12 Aug '09

Hello guys, Here is version 2 of this series. There were lots of little changes and also one significant change. Here is a changelog: - I tried to avoid additional ifdefs in mcbsp code - The WAKEUPEN_ALL macro was removed - I added a REVISIT prefix for the additional delay on mcbsp start - New added constants are now following mcbsp.h style - While reading DMA operation mode of mcbsp, now it prints also some help text to explain the meaning of each value (even though you still write numbers). Besides, I re-wrote the code for ASoC part, now the threshold and dma op mode usage are un-glued from mcbsp code. I tried to do not leave omap-pcm.c dirty, but, there was still one needed callback for setting the mcbsp threshold, as that needs to be done before DMA is started. In this series I've left THRESHOLD mode as default for mcbsp 2 only. But as I saw in previous comments, this is still an open issue :-) Comments, as usual are welcome. Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 375 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 67 +++++- sound/soc/omap/omap-pcm.c | 33 +++- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 517 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ALSA: hda - fix noise issue when recording from digital mic with alc268
by Herton Ronaldo Krzesinski 12 Aug '09

12 Aug '09

With auto config model of alc268 realtek codec, it allows to select any of possible available digital microphone inputs when only one is available. For example, when only digital mic in nid 0x12 is available, on second input source it will allow you to select unavailable digital mic in nid 0x13. The problem is that selecting unavailable digital mic creates a source of noise when recording (I'm not sure if this happens on all machines with alc268 and only one digital mic input, but testing on a quanta uw1 netbook a lot of noise is introduced in recording from digital mic 0x12/first input source, when you select the unavailable digital mic 0x13 for capture source 0x24 in the second input source in mixer). Then to avoid noise when recording from digital mic with auto model in this case, prevent a digital mic input source to be selected if microphone is not available. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++----------------- 1 files changed, 36 insertions(+), 26 deletions(-) I'm attaching the codec info too here of the machine with the problem. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac3207e..7ff2934 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12232,32 +12232,40 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { struct hda_input_mux *imux = &spec->private_imux[0]; - int i, idx1; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch(cfg->input_pins[i]) { - case 0x18: - idx1 = 0; /* Mic 1 */ - break; - case 0x19: - idx1 = 1; /* Mic 2 */ - break; - case 0x1a: - idx1 = 2; /* Line In */ - break; - case 0x1c: - idx1 = 3; /* CD */ - break; - case 0x12: - case 0x13: - idx1 = 6; /* digital mics */ - break; - default: - continue; + int i, idx1, dmic_nid; + + dmic_nid = 0x12; + while (dmic_nid <= 0x13) { + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch (cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + case 0x12: + case 0x13: + if (cfg->input_pins[i] != dmic_nid) + continue; + idx1 = 6; /* digital mics */ + break; + default: + continue; + } + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; } - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux++; + dmic_nid++; } return 0; } @@ -12355,7 +12363,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); - spec->num_mux_defs = 1; + spec->num_mux_defs = 2; spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); @@ -12665,6 +12673,8 @@ static int patch_alc268(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], 0, AC_VERB_SET_CONNECT_SEL, + i < spec->num_mux_defs ? + spec->input_mux[i].items[0].index : spec->input_mux->items[0].index); } -- 1.6.4

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[alsa-devel] [PATCH] ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
by Chaithrika U S 11 Aug '09

11 Aug '09

There is one instance of McASP on DA850/OMAP-L138 SoC. This is connected to TLV320AIC3106 codec for audio playback and capture. This patch adds audio support on this platform. Some of the structure prefix names which are common for DA830/OMAP-L137 EVM and DA850/OMAP-L138 EVM have been renamed to da8xx from da830. 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 | 13 +++++++++++-- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++------ 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 677a538..4dfd4ad 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -43,12 +43,21 @@ config SND_DAVINCI_SOC_SFFSDR Lyrtech SFFSDR board. config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAPL137 EVM" + tristate "SoC Audio support for DA830/OMAP-L137 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 + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 5e2195f..a6939d7 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -14,4 +14,5 @@ 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_DA850_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 a5a9212..46c1b0c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,7 +54,8 @@ 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()) + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) sysclk = 24576000; else @@ -165,7 +166,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; -static struct snd_soc_dai_link da830_evm_dai = { +static struct snd_soc_dai_link da8xx_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], @@ -191,8 +192,15 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { }; static struct snd_soc_card da830_snd_soc_card = { - .name = "DA830 EVM", - .dai_link = &da830_evm_dai, + .name = "DA830/OMAP-L137 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .dai_link = &da8xx_evm_dai, .platform = &davinci_soc_platform, .num_links = 1, }; @@ -209,7 +217,7 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; -static struct aic3x_setup_data da830_evm_aic3x_setup = { +static struct aic3x_setup_data da8xx_evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x18, }; @@ -232,7 +240,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { 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, + .codec_data = &da8xx_evm_aic3x_setup, +}; + +static struct snd_soc_device da850_evm_snd_devdata = { + .card = &da850_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da8xx_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; @@ -255,6 +269,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_evm_snd_devdata; index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.5.6

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[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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