May 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Force AD1988_6STACK_DIG for Asus M3N-HT Deluxe
by Tony Vroon 25 May '11

25 May '11

The microphone input on the back panel (pink connector) stopped operating correctly after an upgrade from 2.6.35 to 2.6.38; the actual problem manifests itself as a lack of microphone bias voltage (VREF_HIZ) on node 0x17. With AD1988_6STACK_DIG the maximum bias voltage (VREF_80) is applied and the headset operates correctly. Signed-off-by: Tony Vroon <tony(a)linx.net> Tested-by: Doug Redlich <pbrigade(a)nxltech.net> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f1b3875..696ac25 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3159,6 +3159,7 @@ static const struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} };

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[alsa-devel] [PATCH 0/9] ASoC: sh: fsi: tidyup and cleanup FSI
by Kuninori Morimoto 25 May '11

25 May '11

Dear Mark, Liam Cc Simon, yoshii-san, Magnus These are tidyup and cleanup patches for FSI Kuninori Morimoto (9): ASoC: sh: fsi: tidyup parameter of fsi_stream_push ASoC: sh: fsi: tidyup unclear variable naming ASoC: sh: fsi: remove pm_runtime from fsi_dai_set_fmt. ASoC: sh: fsi: make sure fsi_stream_push/pop access by spin lock ASoC: sh: fsi: Add fsi_set_master_clk ASoC: sh: fsi: irq control moves to fsi_port_start/stop ASoC: sh: fsi: Add fsi_hw_startup/shutdown ASoC: sh: fsi: remove fsi_module_init/kill ASoC: sh: fsi: cleanup suspend/resume #1 and #2 are tidyup for fsi_fifo_data_ctrl. The variables and sub-functions were unclear nameing. So, it was difficult to understand the detail of calculation. Thank you for pointing it yoshii-san. #3 - #9 are clean up for RuntimePM. Current FSI driver was wrong for RuntimePM. these patches fixup for RuntimePM and clean up for suspend/resume. last patch remove noisy saved_xxx variable. Best regards -- Kuninori Morimoto

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[alsa-devel] A plea for help on mixer support for Fast Track Ultra (8R)
by Felix Homann 25 May '11

25 May '11

Hi, I need some help again by someone who's more familiar with Alsa's inner working than I am. This might be you, Daniel ;-) I've attached a patch series that would add support for the mixer in M-Audio's Fast Track Ultra (8R) devices. BUT, it will break others! It will break others in that mixer controls will show up multiple times. The reason for this seems to be my ambitiuos calling of snd_usb_create_mixer() for quirked devices in card.c. To circumvent the problem of a control showing up multiple times, I check for duplicates in add_unique_control_to_empty_mixer() and skip controls already present. This doesn't work for devices which are not using add_unique_control_to_empty_mixer(), in other words *all* devices not using my messy patch... So, probably the best thing would be to have some kind of entry in quirks-table.h to tell which device would need to be "mixerquirked" to prevent other devices to be disturbed. How can this be done? I would highly appreciate anybody coming up with a proper solution. Kind regards, Felix

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[alsa-devel] [PATCH] ASoC: dapm - fix prefix for DAPM muxes
by Liam Girdwood 24 May '11

24 May '11

Make sure DAPM muxes have a valid kcontrol name instead of NULL. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e..c5d98e3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -489,10 +489,10 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, wlist->widgets[wlistentries - 1] = w; if (!kcontrol) { - if (dapm->codec) + if (dapm->codec && dapm->codec->name_prefix) prefix = dapm->codec->name_prefix; else - prefix = NULL; + prefix = w->name; if (shared) { name = w->kcontrol_news[0].name; -- 1.7.4.1

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Re: [alsa-devel] GR-55 Driver
by Jeffrey Scott Flesher Gmail 24 May '11

24 May '11

I have read this article over a few times, trying to remove the kernel and Reiser4 instructions; and I got a little confused; although I did get it to compile and now it see's the GR-55 as a sound device; I tried to jack it to the output but got no audio; what should I do to test it? Do these instructions look correct? # Path: /home/username/dev/sound/alsa/sound-2.6 git clone git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git cd sound-2.6 git apply 0001-ALSA-snd-usb-add-quirks-for-Roland-GR-55.patch sudo cp -vi /boot/config-$(uname -r) .config sudo make oldconfig # a lot of questions, I hit enter for default make-kpkg clean # Since I'm not building the kernel I didn't do this: # But since this is the only command that will write the new kernel, # what do I do? nice fakeroot make-kpkg --initrd --append-to-version=-lockard --overlay-dir=/home/username/dev/kernel/source kernel-image kernel-headers If I was to guess; I'd change the overlay-dir to use the kernel in the sound-2.6 folder? nice fakeroot make-kpkg --initrd --append-to-version=-gr55 --overlay-dir=/home/username/dev/sound/alsa/sound-2.6/kernel kernel-image kernel-headers # is this right? cd .. ls -lh sudo dpkg -i linux-*.deb uname -r # 2.6.39-rc3-gr55+ One thing strange was that it locked up on reboot, showing rc3, but a reset with previous kernel, I saw the gr55 version so I picked it; it worked as stated above; just don't know why it didn't set this version as default, maybe a step I missed. Sorry it took so long to get back to this; I had a hard drive go bad; so I installed Ubuntu Studio 11.04; so I'm doing this against the newest kernel. Thanks From: Daniel Mack <zonque(a)gmail.com> To: Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> Subject: Re: [alsa-devel] GR-55 Driver Date: Fri, 13 May 2011 09:24:57 +0200 On Fri, May 13, 2011 at 4:30 AM, Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> wrote: > I'm running Ubuntu Studio 10.04 x64 with realtime kernel from > http://ppa.launchpad.net/abogani/ppa/ubuntu lucid (linux-realtime - > 2.6.33-29.1); my motherboard is ASUS M4A78T-E onboard VIA sound, Nvidia 460 > GTX Video card; AMD 955 X4 CPU. > Link to kernel used > https://launchpad.net/~abogani/+archive/ppa/+packages?field.name_filter=&fi… > > I searched for instructions on how to do this, with no good results; so I > must ask someone that knows how. Things are a little more complicated as modern Linux distributions use techniques like initramfs. For Ubuntu, you should follow one of the tutorials. Two I found are here: https://help.ubuntu.com/community/Kernel/Compile http://parabing.com/2011/04/28/ubuntu-natty-a-custom-kernel-is-what-you-wan… That looks comprehensive. Just don't download the ubuntu Kernel sources as described but clone the ALSA repository (which you already did). And instead of downloading the patches as described, just use "git apply" on the file I sent you. On case you want to revert the patching and return to the kernel you originally cloned, just call "git reset --hard". Hope this helps, looking forward to your results. Thanks, Daniel

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[alsa-devel] aplay can't run in background since adding pause support
by Stephen Warren 24 May '11

24 May '11

Jaroslav, In recent versions of aplay, if I try to either run them in the background, or run them in the foreground, then ^Z and bg them, they hang indefinitely: aplay foo.wav & # doesn't play anything aplay foo.wav ... ^Z bg [1]+ Stopped aplay foo.wav bg [1]+ Stopped aplay foo.wav At least, this is true under ChromeOS on ARM with kernel 2.6.38 and ALSA 1.0.24.2, or an Ubuntu Natty rootstock on ARM with the same ChromeOS kernel. This was introduced by alsa-utils commit: 3bd65336222a4d00cefc4db5e74a7a96c07ab567 aplay/arecord: Added hardware pause support (press SPACE or Enter) What happens is that the call to tcsetattr causes the process to receive signal TTOU and be stopped. Running under strace yields an infinite loop with ioctl(TCSETS) returning -ERESTARTSYS, and being retried. I believe this is related to http://lkml.org/lkml/2008/6/2/157. I'm not exactly sure how to solve this though. Another post in that thread implied that adding a signal handler for TTOU and then adjusting the tcsetattr settings might help, but I'm not convinced about that, since I believe the -ERESTARTSYS handling loop is inside the call to tcsetattr, and hence tcsetattr's parameters can't be changed. Does anyone have any ideas? Thanks. -- nvpublic

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[alsa-devel] [PATCH] ASoC: core - Allow some components to probe/remove later or earlier.
by Liam Girdwood 24 May '11

24 May '11

Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe(). Provide a method to allow some components to be probed() later and be also removed() earlier. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 4 +++ include/sound/soc.h | 8 +++++++ sound/soc/soc-core.c | 49 ++++++++++++++++++++++++++++++++++------------ 3 files changed, 48 insertions(+), 13 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1bafe95..ada1f24 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -209,6 +209,10 @@ struct snd_soc_dai_driver { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0..a2139d4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -613,6 +613,10 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; /* SoC platform interface */ @@ -637,6 +641,10 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f9667f6..321182e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1393,7 +1393,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num) +static void soc_remove_dai_link(struct snd_soc_card *card, int num, int early) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; @@ -1410,7 +1410,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed) { + if (codec_dai && codec_dai->probed && + codec_dai->driver->early_remove == early) { if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) @@ -1421,7 +1422,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the platform */ - if (platform && platform->probed) { + if (platform && platform->probed && + platform->driver->early_remove == early) { if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) @@ -1433,11 +1435,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) + if (codec && codec->probed && + codec->driver->early_remove == early) soc_remove_codec(codec); /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed) { + if (cpu_dai && cpu_dai->probed && + cpu_dai->driver->early_remove == early) { if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) @@ -1454,7 +1458,9 @@ static void soc_remove_dai_links(struct snd_soc_card *card) int i; for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_remove_dai_link(card, i, 1); /* early remove */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i, 0); /* late remove */ card->num_rtd = 0; } @@ -1596,7 +1602,7 @@ static int soc_post_component_init(struct snd_soc_card *card, return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num) +static int soc_probe_dai_link(struct snd_soc_card *card, int num, int late) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1605,7 +1611,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int ret; - dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + dev_dbg(card->dev, "probe %s dai link %d late %d\n", card->name, num, late); /* config components */ codec_dai->codec = codec; @@ -1617,7 +1623,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { + if (!cpu_dai->probed && + cpu_dai->driver->late_probe == late) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; @@ -1636,14 +1643,16 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC */ - if (!codec->probed) { + if (!codec->probed && + codec->driver->late_probe == late) { ret = soc_probe_codec(card, codec); if (ret < 0) return ret; } /* probe the platform */ - if (!platform->probed) { + if (!platform->probed && + platform->driver->late_probe == late) { if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1662,7 +1671,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC DAI */ - if (!codec_dai->probed) { + if (!codec_dai->probed && codec_dai->driver->late_probe == late) { if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { @@ -1677,6 +1686,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) list_add(&codec_dai->card_list, &card->dai_dev_list); } + /* complete DAI probe during late probe */ + if (!late) + return 0; + /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -1895,8 +1908,18 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } + /* early DAI link probe */ + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i, 0); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", + card->name, ret); + goto probe_dai_err; + } + } + /* late DAI link probe */ for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i); + ret = soc_probe_dai_link(card, i, 1); if (ret < 0) { pr_err("asoc: failed to instantiate card %s: %d\n", card->name, ret); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: dapm - add method to get AIF widget from stream
by Liam Girdwood 24 May '11

24 May '11

In preparation for ASoC DSP support. Provide a method to get the AIF widget name from the stream name. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 17 +++++++++++++++++ 2 files changed, 19 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c46e7d8..f37c293 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +const char *snd_soc_dapm_get_aif(struct snd_soc_dapm_context *dapm, + const char *stream_name, enum snd_soc_dapm_type type); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5d98e3..1b7f4ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1882,6 +1882,23 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); +const char *snd_soc_dapm_get_aif(struct snd_soc_dapm_context *dapm, + const char *stream_name, enum snd_soc_dapm_type type) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (!w->sname) + continue; + + if (w->id == type && strstr(w->sname, stream_name)) + return w->name; + } + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_aif); + /** * snd_soc_dapm_get_volsw - dapm mixer get callback * @kcontrol: mixer control -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: core - remove superfluous new line.
by Liam Girdwood 24 May '11

24 May '11

Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f051e85..f9667f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1617,7 +1617,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; -- 1.7.4.1

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[alsa-devel] Regression post-ALSA 1.0.22 for Lenovo G555 Internal mic
by Roo - Andrew Baines 24 May '11

24 May '11

Dear All, I have been told I should send an email to this list from this bug<https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5367> (0005367). There is a regression in ALSA 1.0.23 and above for Lenovo sound. The internal mic no longer works. A helpful guy 'Raymond' on the bug identified that pin-ctls is correct in ALSA 1.0.21 and 1.0.22, but incorrect in higher versions due to SND_PCI_QUIRK_VENDOR. This is based on alsa-info.sh run on systems using different versions of ALSA but on the same system. The jacksense has never actually worked but I have always managed to use hda-verb to do this for me - the main issue is the regression with the internal mic. Regards Andrew

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