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

[alsa-devel] [PATCH] ALSA: ASoC: CS4270: export de-emphasis filter as ALSA control
by Daniel Mack 29 Oct '09

29 Oct '09

The CS4270 codec features an de-emphasis filter for compensation of audio material filtered by an 50/15 uS algorithm. Not sure whether we should always enable it for 44100Hz sampling frequency, but it should at least be configurable by the user. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ca1e24a..8069023 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -520,6 +520,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("De-emphasis filter", CS4270_TRANS, 0, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), -- 1.6.5

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[alsa-devel] Digital distortion when converting Mono to Stereo?
by Daniel Mack 29 Oct '09

29 Oct '09

While testing with our ASoC enabled embedded platform, we observed a strange phenomena with Mono audio files. We used external audio test equipment for SNR/THD analysis and used a mono wav file (16bit, 44.1 KHz) which plays a simple sweep from ~10Hz to ~20KHz. As the hardware output is Stereo, an implicit conversion takes places (don't know which part of ALSA actually does it, though). Strangely, the measurement showed a phantom frequency which looks like the sweeping test signal frequency was somehow 'mirrored' at 11.025KHz (fs/4). This phantom noise becomes stronger the lower its frequency gets. So while sweeping, those two peaks were approaching each other, they met at 11KHz, and when the 'real' peak aproached 20KHz, the phantom was at something like 1KHz and had comparable level than the 'real' signal. All that was much stronger on one channel and barely there, but still measurable, on the other one. We first though about an analog filter issue, then suspected the digital input stream, but eventually it turned out that with the same test file in stereo, all these effects suddenly went away. So there must be something in the chain from aplay to the actual PXA SSP ports which adds considerable digital distortion to one channel when converting from Mono to Stereo. I didn't dig any deeper, but still wanted to let you know. Can anyone think of a possible reason for that? It doesn't harm us in our application as we will never play Mono audio eventually, but I guess it could affect others. Thanks, Daniel

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[alsa-devel] aloop / asym (ttable-bindings) Merging audio stream with SIP/Skype - .asoundrc struggle
by yvonne barrymore 28 Oct '09

28 Oct '09

(NOTE: RESENT as received an email the alsa-devel mailinglist had moved.) OBJECTIVE: merge (mplayer cli or aplay) audio stream with SIP/Skype so that music can be heard during chat. Really need help from a developer I think as alsa-users mailinglist not able to help during past 2 weeks. 2 weeks happily studying alsa/plugins, posted/searched on mailing lists and tried irc as well. Could really use a guru assist. I get the parts mostly, though obviously not how it all comes together in an .asoundrc. - aloop driver is installed. After scores of attempts now wondering: maybe having the music track playing on it's own soundcard or loopback and having that merge into the microphone? channel of sip/skype's soundcard. Like, have the music play out on the intel card 0, and loopback through aloop card 1, then merge that with asym/dmix onto sip/skype's headset card 2 ???? I kinda know what the plugs do, though not how to hook-up through .asoundrc. I've found 3 or 4 great posts on alsa-devel and alsa-users though ultimately for me just glimmers of working. Rather than glut up the list here I've only posted my aplay -l Can anyone comment on exactly what I'd put in my .asoundrc? What the flow/parts would be even? To stop stuttering (fighting for card control ? etc.) I guessed that the music playing on it's own card might be good? Very happy to post what I've got though thought I should ask first if anyone has found a best practice for this. **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] 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 1: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] 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 2: default [C-Media USB Headphone Set ], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

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[alsa-devel] A question about the soc jack status
by Joonyoung Shim 28 Oct '09

28 Oct '09

Hi, all. I have a question about soc jack. I know the soc jack reports the jack status when the jack status is change only. For example when the jack is inserted or released, the soc jack reports the jack status to user via input subsystem. But i want to know the jack status always regardless of the event to insert or release a jack. For example we can check the jack status always if the jack status is reported via sysfs file. So, is there how to know the jack status always the currently?

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[alsa-devel] [PATCH 0/2] ASoC: TWL4030: bypass magic removal and cleanup patch
by Peter Ujfalusi 28 Oct '09

28 Oct '09

Hello, Since the soc core is now handling the codec bias state based on the complete paths as well, there is no need for the bypass tracking in the codec. The second patch replaces the missleading codec_muted to apll_enabled, which is what actually is handled by it. This series applies on top of Mark's tree, for-2.6.33-pending branch at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Since the underlying twl4030 patches are not yet in the topic/asoc of Takashi's tree. --- Peter Ujfalusi (2): ASoC: TWL4030: Remove bypass tracking ASoC: TWL4030: Change codec_muted to apll_enabled sound/soc/codecs/twl4030.c | 146 ++++++++++++-------------------------------- 1 files changed, 39 insertions(+), 107 deletions(-)

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[alsa-devel] still: clicking noise on Sigmatel 9202 - how can I use patches for 2.6.24 kernel on 2.6.30?
by Christian Steffen 28 Oct '09

28 Oct '09

Hello, I still have my problem that I have a constant clicking noise coming out of my boxes with a sigmatel 9202 sound chip on intel-hda (Poulsbo) soundcard. I now found two patches for the alsa driver that can be applied on a 2.6.24 kernel. Those patches are exactly for the sigmatel 9202, so possibly they would fix my problem. http://redirectingat.com/?id=593X1004&url=http%3A%2F%2Flists.moblin.org%2Fpi permail%2Fcommits%2F2007-December%2F001232.html and http://redirectingat.com/?id=593X1004&url=http%3A%2F%2Fwww.google.com%2Furl% 3Fsa%3Dt%26source%3Dweb%26ct%3Dres%26cd%3D9%26ved%3D0CDAQFjAI%26url%3Dhttp%2 53A%252F%252Fwww.soulinfo.com%252F~hugang%252Ftmp%252Fubuntu-hardy%25252Btux onice%252Fdebian%252Fbinary-custom.d%252Flpia%252Fpatchset%252F0010-hda_sigm atel.patch%26ei%3D4AfoSs24OpeG_Abfy7yGCA%26usg%3DAFQjCNH-mrDdDcgvhNh_eM5hy4W hepUtUQ%26sig2%3D2fK3ED_DyaLSexlPlkFufQ I found both patches here: http://software.intel.com/en-us/articles/patches-for-moblin-kernel-on-menlow / I think these patches aren't yet completely integrated into alsa driver 1.0.21? I tried to manually insert them into the files of my kernel 2.6.30 source files but without any expertise this is "impossible" to me. I think this is not the easiest job but I appreciate any help! :) Thank you very much, Cheers Christian

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Re: [alsa-devel] A question about the soc jack status
by Joonyoung Shim 28 Oct '09

28 Oct '09

On 10/28/2009 4:32 PM, Naveen Krishna Ch wrote: > 2009/10/28 Joonyoung Shim <jy0922.shim(a)samsung.com> > >> Hi, all. >> >> I have a question about soc jack. I know the soc jack reports the jack >> status when the jack status is change only. For example when the jack >> is inserted or released, the soc jack reports the jack status to user >> via input subsystem. >> >> But i want to know the jack status always regardless of the event to >> insert or release a jack. For example we can check the jack status >> always if the jack status is reported via sysfs file. >> > There is a /sys/class/soc/jack/status > Dont remember the exact path. > Cat of that file will tell us right > I cannot find "jack" word under /sys directory. Can you give me the exact path? >> So, is there how to know the jack status always the currently? >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > > >

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[alsa-devel] What can change after suspend ?
by Giuliano Pochini 28 Oct '09

28 Oct '09

I'm adding suspend/resume support to echoaudio drivers. I was reading /Documentation/power/pci.txt: " pci_set_power_state(PCI_D0); pci_restore_state(); /* device's irq possibly is changed, driver should take care */ " The card does not support D[0-3] states. The driver suspends the card by turning the DSP off, without changing the PCI state. Can IRQ, memory mapped regions, whatever, change on resume ? Are pci_save_state() and pci_enable_wake() useful/required in this case ? -- Giuliano.

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[alsa-devel] Problems with snd_pcm_writei() function
by Douglas Lopes Pereira 27 Oct '09

27 Oct '09

Hi, I'm modifying an application that calls snd_pcm_writei() function from libasound2-dev (on verstion 1.0.16-2). I does work. I'm splitting my audio data into 160 bytes buffers but when buffer with different size is passed to that function I hear a noise (like 'Krrr') on my audio device. I tried to add zeroed bytes to complete the buffer but it didn't work. I tried to ignore the last buffer and it worked but with some side effects: No other operation worked properly after that. Please, can anyone help? Thanks in advance. Regards, Douglas

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Re: [alsa-devel] [Intel-gfx] Problems with HDMI audio on Intel DG45FC motherboard
by Wu Fengguang 27 Oct '09

27 Oct '09

On Tue, Oct 27, 2009 at 07:11:25AM +0800, David Härdeman wrote: > On Tue, Oct 20, 2009 at 03:43:53PM -0700, Shane W wrote: > > On Tue, Oct 20, 2009 at 09:26:00AM +0800, Wu Fengguang wrote: > > > HDMI codec/sink seem to take some time to response to the output > > > enable and new infoframe, so there are some delay. I've moved the HDMI > > > output enable command to module load time, this helps. The infoframe > > > is in theory created in run time based on the format of _each_ new > > > audio stream, so infoframe transmission has to be started/stopped > > > for each track.. > > > > I know this has come up before but can't we not start/stop > > the infoframe if the sample format on the new track is the > > same as the old. I mean playing an album, you're gonna get > > 44100, 2ch s16le 95% of the time so that would atleast > > cause the problem to appear much less. > > Tried it, still had the second-track-produces-silence bug on my receiver > so I couldn't test it properly. > > I however did manage to create a different patch which fixed my receiver > bug (yay!)...by sending "refer-to-stream-header" audio infoframes from > pcm close to pcm open...something like this quick hack: This is an important clue, thank you! :) The patch is not quite logical. Hopefully the sticky infoframe patch can help in the same way. Attached is the broken up patches, hope that helps your debugging. The last 4 patches in the series are inspired by your comments. Thanks, Fengguang > > --- linux-2.6.32-rc5/sound/pci/hda/patch_intelhdmi.c 2009-10-16 02:41:50.000000000 +0200 > +++ ../linux-2.6.32-rc5/sound/pci/hda/patch_intelhdmi.c 2009-10-26 23:39:54.000000000 +0100 > @@ -552,8 +552,16 @@ > struct snd_pcm_substream *substream) > { > struct intel_hdmi_spec *spec = codec->spec; > + struct hdmi_audio_infoframe ai = { > + .type = 0x84, > + .ver = 0x01, > + .len = 0x0a, > + .CC02_CT47 = 0x00, > + }; > > hdmi_stop_infoframe_trans(codec); > + hdmi_fill_audio_infoframe(codec, &ai); > + hdmi_start_infoframe_trans(codec); > > return snd_hda_multi_out_dig_close(codec, &spec->multiout); > } > @@ -571,6 +579,7 @@ > > hdmi_set_channel_count(codec, substream->runtime->channels); > > + hdmi_stop_infoframe_trans(codec); > hdmi_setup_audio_infoframe(codec, substream); > > return 0; > > > Not sure how/if something like that could be turned into an acceptable > patch though... > > -- > David Härdeman

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