February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] ASoC: omap: mcbsp: small cleanups for the sidetone code
by Peter Ujfalusi 28 Feb '12

28 Feb '12

Hello, to make the code a bit nicer to look at. This series is on top of the McBSP move changeset: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049628.h… Regards, Peter --- Peter Ujfalusi (2): ASoC: omap: mcbsp: Use uniform st_data pointer initialization ASoC: omap: mcbsp: Remove redundant checks for the st_data pointer sound/soc/omap/mcbsp.c | 25 +++++++------------------ 1 files changed, 7 insertions(+), 18 deletions(-) -- 1.7.8.4

3 6

[alsa-devel] [PATCH] ALSA: hda - Add position_fix=4 (COMBO) option
by Takashi Iwai 28 Feb '12

28 Feb '12

This patch adds a new position_fix option value, 4, as a combo mode to use LPIB for playbacks and POSBUF for captures. It's the way recommended by Intel hardware guys. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Seth, this is the patch to add the mode you mentioned. It'll be queued for 3.4 kernel. This mode isn't chosen as default yet, but if this is confirmed to work more reliably on Intel controllers, we can set these as default for them. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +++++++- Documentation/sound/alsa/HD-Audio.txt | 7 ++++++- sound/pci/hda/hda_intel.c | 10 +++++++++- 3 files changed, 22 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 936699e..9af64c5 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, + 3 = VIACOMBO, 4 = COMBO) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the @@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPIB register is more accurate than the position buffer.) + position_fix=3 is specific to VIA devices. The position + of the capture stream is checked from both LPIB and POSBUF + values. position_fix=4 is a combination mode, using LPIB + for playback and POSBUF for capture. + NB: If you get many "azx_get_response timeout" messages at loading, it's likely a problem of interrupts (e.g. ACPI irq routing). Try to boot with options like "pci=noacpi". Also, you diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 91fee3b..7813c06 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -59,7 +59,12 @@ a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. `position_fix=2` means to use the position-buffer. `position_fix=3` means to use a combination of both methods, needed -for some VIA and ATI controllers. 0 is the default value for all other +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +`position_fix=4` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might help. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e354c16..6e958bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -330,6 +330,7 @@ enum { POS_FIX_LPIB, POS_FIX_POSBUF, POS_FIX_VIACOMBO, + POS_FIX_COMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -2520,6 +2521,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: + case POS_FIX_COMBO: return fix; } @@ -2699,6 +2701,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); + /* combo mode uses LPIB for playback */ + if (chip->position_fix[0] == POS_FIX_COMBO) { + chip->position_fix[0] = POS_FIX_LPIB; + chip->position_fix[1] = POS_FIX_AUTO; + } + check_probe_mask(chip, dev); chip->single_cmd = single_cmd; -- 1.7.9

2 2

[alsa-devel] [PATCH 1/4] ASoC: imx: let SND_MXC_SOC_FIQ select FIQ
by Shawn Guo 28 Feb '12

28 Feb '12

CONFIG_FIQ is only needed when CONFIG_SND_MXC_SOC_FIQ is selected to build imx-pcm-fiq.c, so let SND_MXC_SOC_FIQ select FIQ. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 91b66ef..17810d9 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select FIQ select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to @@ -12,6 +11,7 @@ if SND_IMX_SOC config SND_MXC_SOC_FIQ tristate + select FIQ config SND_MXC_SOC_MX2 tristate -- 1.7.5.4

5 66

Re: [alsa-devel] Cranking sound coming out of speakers with the new 3.1 kernel
by Takashi Iwai 28 Feb '12

28 Feb '12

At Tue, 28 Feb 2012 16:16:48 +0100, Jan Prunk wrote: > > Hello ! > > On Sat, Jan 14, 2012 at 12:04 AM, Jonathan Nieder <jrnieder(a)gmail.com> wrote: > > Hi again, > > > > Jan Prunk wrote: > > > >> I tried booting into 3.2.0-rc7-686-pae, but the crunching sound > >> is still present, I am able to play music file properly, and > >> crunching is in the background, and also while not processing > >> any sounds, the crunching appears out of the speakers. > > > > Ok, thanks for testing. Please send a summary of this regression to > > alsa-devel(a)alsa-project.org, cc-ing Takashi Iwai <tiwai(a)suse.de> and > > either me or this bug log so we can track it. > > > > Be sure to mention: > > > > - steps to reproduce, what you expect versus what actually happens, > > and how that indicates a bug > > The sound is ok in kernel version "3.0.0-1-686-pae" but on all newer > versions there are sparkling/farting noises coming out of the speaker. > The noise is not very loud, its barely noticeable, but its repetative > and doesn't go away. And I think it happens while typing on the > keyboard and a bit later, then it dissapears for a few seconds, > or by the time that I type again. > > > - which kernel versions you tried, and what happens with each > I tried running alsa-info.sh as a "user". > > I tried 3.0.0-1-686-pae here is the debug: > http://www.alsa-project.org/db/?f=c1149b6466b2068d79c7517a8b0808d8b5c26e6b > --- > Linux vaio 3.2.0-1-686-pae #1 SMP Fri Feb 17 06:27:21 UTC 2012 i686 GNU/Linux > ii linux-image-3.2.0-1-686-pae 3.2.6-1 > http://www.alsa-project.org/db/?f=fddbfbdc4fc44555a6dc851ecdc70760da6dd840 > --- > Linux vaio 3.2.0-rc7-686-pae #1 SMP Wed Dec 28 21:26:25 UTC 2011 i686 GNU/Linux > ii linux-image-3.2.0-rc7-686-pae 3.2~rc7-1~experimental.1 > http://www.alsa-project.org/db/?f=78c8919e87dd03294484f83b5f7ae44331b5edfe > Being booted into this version for 15 minutes, it seems to not make any noises, > but I will resubmit the info later if the noises will start to appear. 3.2.0-rc7 alsa-info shows that the headphone jack is plugged, thus the speaker is muted. It's natural that the noise goes away in this state. Apart from that, I see no obvious problem in 3.2.0 output. Does the problem still happen when you disable "Auto-Mute Mode" mixer enum? % amixer -c0 set "Auto-Mute Mode" "Disabled" Also, try to pass algin_buffer_size=0 option to snd-hda-intel. I don't expect much that this will influence, but at least, it's a difference between 3.0 and 3.2. Takashi > > - alsa-info.sh output, as an attachment > > > > - if you can make a recording of the strange sound available > > somewhere online, that would be ideal > > Because the noise is barely heard, I would need to find a good mic. > to be able to record it, but I could do that in the next bug submission. > > > - whether you are able to bisect or test patches if needed > > I might try that, if there are any manuals around how to do it, > but my response might take some time. > > > [1] may have more hints. > > > > Thanks and good luck, > > Jonathan > > > > [1] http://alsa-project.org/main/index.php/Help_To_Debug_Intel_HDA > > Kind regards, > Jan Prunk > -- > Jan Prunk http://www.prunk.si > 0x00E80E86 http://pgp.prunk.si > http://AS50763.peeringdb.com >

1 0

[alsa-devel] [PATCH] ASoC: DT: Update digital microphone binding documentation to PAZ00 board.
by Leon Romanovsky 28 Feb '12

28 Feb '12

This patch updates device tree binding documentation to add digital microphone to PAZ00 board. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- .../bindings/sound/tegra-audio-alc5632.txt | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt index 300f2da..9e305bb 100644 --- a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt +++ b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt @@ -25,12 +25,14 @@ Required properties: * MIC2_P * MIC2_N * MICBIAS1 + * DMICDAT Board connectors: * Headset Stereophone * Int Spk * Headset Mic + * Digital Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller - nvidia,audio-codec : The phandle of the ALC5632 audio codec -- 1.7.3.4

2 1

[alsa-devel] [PATCH v2 0/9] imx patches when I enable imx6q sabrelite audio
by Richard Zhao 28 Feb '12

28 Feb '12

Changes in v2: - change dt peroperty/label to lower case as much as possible keep XXX-supply property upper case. - convert audmux-v2 to platform device driver - move audmux-v2 driver to sound/soc/imx [PATCH v2 1/9] ARM: mxc: make imx_dma_is_general_purpose more [PATCH v2 2/9] ARM: imx6q: add cko1 clock [PATCH v2 3/9] ARM: dts: imx6q-sabrelite: add 2p5v and 3p3v [PATCH v2 4/9] ARM: dts: imx6q-sabrelite: add sgtl5000 audio codec [PATCH v2 5/9] ARM: mxc: convert audmux-v2 to a platform driver [PATCH v2 6/9] ARM: mx31ads: add audmux device [PATCH v2 7/9] ARM: mxc: move audmux-v2 to sound/soc/imx [PATCH v2 8/9] ASoC: imx: add dt support for audmux-v2 [PATCH v2 9/9] ARM: imx6q-sabrelite: add audmux support Thanks Richard

5 42

[alsa-devel] [GIT PULL FOR v3.4] tea575x update to latest V4L2 framework requirements
by Hans Verkuil 28 Feb '12

28 Feb '12

Hi Mauro, This patch series updates the tea575x drivers to the latest V4L2 framework requirements, converts radio-maxiradio to tea575x as well and adds hwseek support to these drivers. All changes are V4L related, even though some of the patches touch sound drivers (but only the V4L parts of those sound drivers). Tested with SF16-FMR2, SF64-PCR, SF64-PCE2, SF256-PCP (Ondrej) and maxiradio (Hans). Regards, Hans The following changes since commit a3db60bcf7671cc011ab4f848cbc40ff7ab52c1e: [media] xc5000: declare firmware configuration structures as static const (2012-02-14 17:22:46 -0200) are available in the git repository at: git://linuxtv.org/hverkuil/media_tree.git radio-pci Hans Verkuil (4): tea575x-tuner: update to latest V4L2 framework requirements. tea575x: fix HW seek radio-maxiradio: use the tea575x framework. V4L2 Spec: return -EINVAL on unsupported wrap_around value. .../DocBook/media/v4l/vidioc-s-hw-freq-seek.xml | 6 +- drivers/media/radio/Kconfig | 2 +- drivers/media/radio/radio-maxiradio.c | 379 ++++---------------- drivers/media/radio/radio-sf16fmr2.c | 61 +++- include/sound/tea575x-tuner.h | 6 +- sound/i2c/other/tea575x-tuner.c | 169 ++++++--- sound/pci/es1968.c | 15 + sound/pci/fm801.c | 20 +- 8 files changed, 273 insertions(+), 385 deletions(-)

1 0

[alsa-devel] [PATCH] ASoC: omap3beagle: Add dai link for McBSP2 master configuration (CLKS source)
by Peter Ujfalusi 28 Feb '12

28 Feb '12

Create new dai link so we can choose between two different modes of operation: PCM0: twl4030 master, McBSP2 slave (current configuration) PCM1: twl4030 slave, McBSP2 slave (SRG source is from CLKS pin) This will help us tracking down, and test different setups on Beagle board. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, While tracking down an issue regported by Grazvydas Ignotas regarding to the McBSP move series (Pandora setup was not working) I have noticed that we can use the Beagle board in a similar way as Pandora uses it (McBSP master, twl4030 slave, CLKS pin is connected to twl4030 256*FS clock output). With this patch we can switch between the two modes by playing on hw:0,0 or on hw:0,1. I will add the third option later where McBSP is the master, but the clock for SRG is coming from internal source. This patch depends on the McBSP rewrite series (OMAP/ASoC: Move and merge McBSP driver under ASoC) http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049628.h… Regards, Peter sound/soc/omap/omap3beagle.c | 110 ++++++++++++++++++++++++++++++++++++++---- 1 files changed, 100 insertions(+), 10 deletions(-) diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 2830dfd..a2f9130 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -34,6 +34,7 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" +/* McBSP2 slave, TWL4030 master */ static int omap3beagle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -80,6 +81,14 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Set McBSP clock to PER_96M_FCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, + 96000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set cpu system clock\n"); + return ret; + } + return 0; } @@ -87,23 +96,104 @@ static struct snd_soc_ops omap3beagle_ops = { .hw_params = omap3beagle_hw_params, }; +/* McBSP2 master, TWL4030 slave */ +static int omap3beagle_slave_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external (CLKS) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + printk(KERN_ERR "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3beagle_slave_ops = { + .hw_params = omap3beagle_slave_hw_params, +}; + /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link omap3beagle_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .platform_name = "omap-pcm-audio", - .codec_dai_name = "twl4030-hifi", - .codec_name = "twl4030-codec", - .ops = &omap3beagle_ops, +static struct snd_soc_dai_link omap3beagle_dai[] = { + { + /* McBSP2 slave, TWL4030 master configuration */ + .name = "TWL4030 master", + .stream_name = "TWL4030 master", + .cpu_dai_name = "omap-mcbsp.2", + .platform_name = "omap-pcm-audio", + .codec_dai_name = "twl4030-hifi", + .codec_name = "twl4030-codec", + .ops = &omap3beagle_ops, + }, { + /* McBSP2 master, TWL4030 slave configuration */ + .name = "TWL4030 slave", + .stream_name = "TWL4030 slave", + .cpu_dai_name = "omap-mcbsp.2", + .platform_name = "omap-pcm-audio", + .codec_dai_name = "twl4030-hifi", + .codec_name = "twl4030-codec", + .ops = &omap3beagle_slave_ops, + }, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3beagle = { .name = "omap3beagle", .owner = THIS_MODULE, - .dai_link = &omap3beagle_dai, - .num_links = 1, + .dai_link = omap3beagle_dai, + .num_links = ARRAY_SIZE(omap3beagle_dai), }; static struct platform_device *omap3beagle_snd_device; -- 1.7.8.4

2 3

Re: [alsa-devel] alsa usb audio class 2
by Daniel Mack 28 Feb '12

28 Feb '12

On 28.02.2012 09:57, Markus Bollinger wrote: > OK, thank you Takashi. > I attached the log file of the 'lsusb -v' output Thanks for the dump. Are there any kernel messages that are generated when plugging in the device? Check with 'dmesg'. Also, from the dump I can see that some string descriptors are really broken in this device. Don't know if that causes your trouble though. Daniel

1 0

[alsa-devel] [PATCH 00/38] Realtek HD-audio auto-parser
by Takashi Iwai 28 Feb '12

28 Feb '12

Hi, for 3.4 kernel, finally I got rid of all static quirks from HD-audio Realtek codec parser. This means that the all necessary quirks have been converted to alc_fixup either the pin-config overrides, the additional verbs, or the extra init functions. Since some machines have really weird pin mappings, I introduced the new mechanism in the parser to evaluate the best DAC->pin connections, which is based on the badness calculation. You won't notice on most of machines but some machines that have fewer DACs than necessary output pins will get different mappings by this change. The whole changes can be seen in sound git tree topic/hda branch, as usual. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Now it's merged to main branch, so the snapshot tarball contains these changes, too. The rest of the thread will contain the changes in patch_realtek.c. It's just for refernces, and I omitted the other files. For the complete changes, take a look at the git tree. thanks, Takashi

2 39