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[alsa-devel] S/PDIF-Out on Asus M3A-H/HDMI with ALC1200
by Thomas Schneider 25 Dec '08

25 Dec '08

Hi, i am looking for a way to make Digital-Audio work with the external and internal S/PDIF-Output of my Motherboard Asus M3A-H/HDMI with ALC1200 Soundard. I use the latest unstable snapshot and i can´t make it work at the same time. On the external S/PDIF-Output is my Digital Receiver connected, on the internal one my GeForce 8500GT to get sound over HDMI on my LCT-TV. When i load the driver without any model-options my Soundcard is detected as Asus P5Q-EM HDMI ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:2716: hda_codec: model 'asus-p5q' is selected for config 1043:82fe (Asus P5Q-EM HDMI) [ 6.721700] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Headphone Playback Volume, skipped [ 6.721714] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Volume, skipped [ 6.721716] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Volume, skipped [ 6.721719] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Line-Out Playback Volume, skipped [ 6.721721] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave PCM Playback Volume, skipped [ 6.721729] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Switch, skipped [ 6.721731] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Switch, skipped Now i have sound on my LCD-TV over HDMI (internal S/PDIF-Out) When i load the driver with model=6stack-dig i have sound on my Digital-Receiver (external S/PDIF-Out). In the mailinglist-archive or in the doku i have seen, that the only difference between this models is the DIGOUT_NID asus-p5q = 0x10 6stack-dig = 0x06 I there a way to get both output to work at the same time so i have sound on my LCD-TV and my Digital-Receiver ? Regards Thomas

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[alsa-devel] [PATCH 0/8]ALSA: ASoc: DaVinci: cleanup
by Troy Kisky 24 Dec '08

24 Dec '08

Sorry, for the long series to do something so simple, but there were unresolved objections when I first posted this to the davinci list. Hopefully, having this split into tiny pieces will make resolving those differences easier.

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[alsa-devel] [PATCH] ALSA: hda - Add missing terminators in patch_sigmatel.c
by Herton Ronaldo Krzesinski 24 Dec '08

24 Dec '08

Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 71d3a77..35b83dc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -841,6 +841,7 @@ static struct hda_verb stac92hd83xxx_core_init[] = { /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, + {} }; static struct hda_verb stac92hd71bxx_core_init[] = { @@ -850,6 +851,7 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} }; #define HD_DISABLE_PORTF 2 @@ -1681,6 +1683,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + {} /* terminator */ }; static unsigned int ref92hd71bxx_pin_configs[11] = {

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[alsa-devel] Urgent Help Needed:ALsa Querries
by SUBHRANIL CHOUDHURY 23 Dec '08

23 Dec '08

I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, Subhranil

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[alsa-devel] [PATCH] ASoC: Mark non-connected TWL4030 pins for pandora
by Grazvydas Ignotas 23 Dec '08

23 Dec '08

Pandora has all TWL4030 output pins floating, it uses external DAC for playback. Mark those outputs as not connected using DAPM calls. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index bd91594..0ffb3de 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -180,6 +180,19 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + snd_soc_dapm_nc_pin(codec, "OUTL"), + snd_soc_dapm_nc_pin(codec, "OUTR"), + snd_soc_dapm_nc_pin(codec, "EARPIECE"), + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"), + snd_soc_dapm_nc_pin(codec, "PREDRIVER"), + snd_soc_dapm_nc_pin(codec, "HSOL"), + snd_soc_dapm_nc_pin(codec, "HSOR"), + snd_soc_dapm_nc_pin(codec, "CARKITL"), + snd_soc_dapm_nc_pin(codec, "CARKITR"), + snd_soc_dapm_nc_pin(codec, "HFL"), + snd_soc_dapm_nc_pin(codec, "HFR"), + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); if (ret < 0) -- 1.5.4.3

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[alsa-devel] [topic/asoc][PATCH] ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
by Jarkko Nikula 23 Dec '08

23 Dec '08

N810 bootloader muxes I2S pins for OMAP2420 EAC block while N810 ASoC drivers are using McBSP block so the kernel have to change configuration runtime. Author has not seen problems using kernel pin multiplexing on N810 but very many times unworking audio after forgotten to enable it and spending 15 minutes each time to figure it out again... This change makes it easier for other users as well. If problems arise, then they are better to find and fix in OMAP pin multiplexing framework. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a7b1d77..4f7f040 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -10,6 +10,7 @@ config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 select SND_OMAP_SOC_MCBSP + select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.5.6.5

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[alsa-devel] [topic/asoc][PATCH]] ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
by Jarkko Nikula 23 Dec '08

23 Dec '08

N810 bootloader muxes I2S pins for OMAP2420 EAC block while N810 ASoC drivers are using McBSP block so the kernel have to change configuration runtime. Author has not seen problems using kernel pin multiplexing on N810 but very many times unworking audio after forgotten to enable it and spending 15 minutes each time to figure it out again... This change makes it easier for other users as well. If problems arise, then they are better to find and fix in OMAP pin multiplexing framework. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a7b1d77..4f7f040 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -10,6 +10,7 @@ config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 select SND_OMAP_SOC_MCBSP + select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.5.6.5

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Re: [alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by Eliot Blennerhassett 22 Dec '08

22 Dec '08

SUBHRANIL CHOUDHURY wrote: > Eliot, > Suppose i need to change the DAC gain, ADC gain of the audio codec and > also disable and enable AGC of the codec i need to put some IOCTLs in > the driver. > So, in case of ALSA how do i implement those IOCTLs in the driver and > also expose them to the application through the ALSA library. No! For this kind of control You just need to implement standard alsa controls in your driver. BOOLEAN: AGC ON/OFF INTEGER: DAC gain. See "writing an ALSA driver" for details, and study the source of any ALSA driver. http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c843.htm -- Eliot

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[alsa-devel] [topic/asoc][PATCH] ASoC: Fix DSP formats in SSM2602 audio codec
by Jarkko Nikula 22 Dec '08

22 Dec '08

Thanks to Troy Kisky <troy.kisky(a)boundarydevices.com> for noticing. - DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2 - DSP_B has 0-bit data delay which corresponds to submode 1 - Currently driver sets them opposite so swap the submode setting Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Cliff Cai <cliff.cai(a)analog.com> --- sound/soc/codecs/ssm2602.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2325aef..cac3736 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -454,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- 1.5.6.5

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Re: [alsa-devel] [PATCH] ASoC: typo
by Liam Girdwood 22 Dec '08

22 Dec '08

On Mon, 2008-12-22 at 17:40 +0100, Roel Kluin wrote: > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> > --- > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index acd68ef..227c865 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, > prtd->dma_data = dma_data; > err = omap_request_dma(dma_data->dma_req, dma_data->name, > omap_pcm_dma_irq, substream, &prtd->dma_ch); > - if (!err & !cpu_is_omap1510()) { > + if (!err && !cpu_is_omap1510()) { > /* > * Link channel with itself so DMA doesn't need any > * reprogramming while looping the buffer Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Liam

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