August 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] omap: rx51: Remove redefinition of rx51_aic3x_data
by Jarkko Nikula 26 Aug '10

26 Aug '10

Redefinition of rx51_aic3x_data got introduced when merging upstread changes for 2.6.36 and ASoC Multi-Component Support. This must go together with multi-component commit f0fba2a. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap2/board-rx51-peripherals.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-omap2/board-rx51-peripherals.c b/arch/arm/mach-omap2/board-rx51-peripherals.c index 897d960..11ce4b2 100644 --- a/arch/arm/mach-omap2/board-rx51-peripherals.c +++ b/arch/arm/mach-omap2/board-rx51-peripherals.c @@ -710,10 +710,6 @@ static struct twl4030_platform_data rx51_twldata __initdata = { .vio = &rx51_vio, }; -static struct aic3x_pdata rx51_aic3x_data __initdata = { - .gpio_reset = 60, -}; - static struct tpa6130a2_platform_data rx51_tpa6130a2_data __initdata = { .id = TPA6130A2, .power_gpio = 98, -- 1.7.1

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[alsa-devel] How to transfer more than 2 channels between i.MX25 & AD1938 using SSI?
by Sven Zeisberg 26 Aug '10

26 Aug '10

Hi, has anybody succeeded to transfer more than 2 audio channels between i.MX25 and a CODEC throught the i.MX25 SSI Port? We have succeeded to set the CODEC (AD1938) as Master to the required mode (TDM with 8 Timeslots and 32 Bit per Timeslot). However, as soon as we start to play a wav file with aplay, we can see on the logic analyzer that the data format on the SSI lines is reset to a stereo mode (2 channels with frame sync changing level between channels). In the imx-ssi driver, I can see that .channel_max = 2 - therefore my question: does this driver support more than two channels at all? BR Sven

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[alsa-devel] EEE PC 1001P Driver Issues
by Alison Gravley 26 Aug '10

26 Aug '10

ALSA Info is at: http://www.alsa-project.org/db/?f=49ae352709904da7e3201343ec639bdaf5c8563a Issues: When plugging in headhpones, it plays audio through both them and speaker. Have to manually mute one or the other. Built in Mic is also buggy and does not work reliably, though external ones through mic jack work ok. Have had same issues regardless of distro it seems. If you need any more information, feel free to ask. -- Alison

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[alsa-devel] [PATCH] ALSA: HDA: Add Sony VAIO quirk for ALC269
by David Henningsson 26 Aug '10

26 Aug '10

The attached patch enables playback on a Sony VAIO machine. References: http://launchpadlibrarian.net/53762567/Card0.Codecs.codec.0.txt http://launchpadlibrarian.net/53762571/PciMultimedia.txt https://bugs.launchpad.net/bugs/618271 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] explicit commandline control of speakers vs headphone output
by Eric Holstege 26 Aug '10

26 Aug '10

It appears that when I have a headphone plugged in, the speakers are automatically disabled and the output sound is sent to the headphone. However, I want to explicitly control (e.g. via a script or commandline - not a GUI) whether the sound from an (arbitrary) application is going to the speakers or to the headphone even when the headphone is plugged in. (When I say "arbitrary application" that means I can't alter the code of the application itself). Is this possible? (I have heard that ALSA mutes the speakers when it detects plugged-in headphones. Is switching between them as simple as using e.g. amixer(1) to unmute the speakers and mute the headphones and vice versa, or is there more to it.)

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[alsa-devel] [PATCH 0/3] alsa-lib: UCM - Use Case Manager
by Liam Girdwood 25 Aug '10

25 Aug '10

This patch series follows on from the RFC and adds sound card hardware Use Case Management support to alsa-lib. The main features of UCM are :- o Abstraction of sound card hardware routing and controls into high level use case verbs, devices and modifiers. o Discovery of source/sink PCMs for each use case. o Discovery of hardware master volume controls for each use case. It's intended UCM will provide clients (like pulseaudio) complete control over hardware signal routing within a sound card without any knowledge of the sound card hardware. Use case definintions are stored in text files (per sound card) and are now divided into three components as descibed in the RFC. A working use case definition file can be found here :- http://git.slimlogic.co.uk/cgi-bin/cgit.cgi/alsa-ucm-conf.git/tree/cards/SD… The three components of a use case are now :- 1) Verb (or action). This is mandatory and describes (in terms of ALSA control settings) the system use case action. e.g "Phone Call", "Music Playback". 2) Device. This is also mandatory and describes the audio playback or capture device. e.g. "Handset", "Bluetooth", "Headset". 3) Modifier. This is optional and is used to modify the current use case verb. i.e. The user could be listening to music when an incoming phone call requires that a ring tone is played. In this case the use case verb would be set to "Music Playback" when the music playback started and then the "Tone Playback" modifier could be enabled to play the ring tone. The use case verb would then be changed to "Voice Call" if the call was accepted. Patch 2/3 is a little on the large side so, I've included a link if it gets bounced by the list. http://git.slimlogic.co.uk/cgi-bin/cgit.cgi/alsa-lib.git/commit/?h=ucm-upst… Liam Girdwood (3): ucm: header - ALSA Use Case Manager ucm: core - Add initial Use Case Manager support. ucm: build - add build support for Use Case Manager doc/doxygen.cfg.in | 1 + include/Makefile.am | 2 +- include/use-case.h | 427 +++++++ src/Makefile.am | 2 +- src/use-case.c | 3515 +++++++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 3945 insertions(+), 2 deletions(-) create mode 100644 include/use-case.h create mode 100644 src/use-case.c

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[alsa-devel] Loopback device conf/alias
by pl bossart 25 Aug '10

25 Aug '10

I played with the snd_aloop module that Jaroslav added a couple of days ago. It just does what I wanted and seems to work ok :-) Thanks for this patch! I would like however to expose this output as a PulseAudio profile. Using a string such as hw:%f conflicts with other analog output profiles, and the actual card index will vary depending on each hardware setup. Is there a way to have an alias or a configuration so that I could use a profile definition as aloop:%f in PulseAudio, and use arecord -Daloop for my monitoring app. By default the playback device would be 0 and the capture one 1. As long as the card index is handle automatically I am happy. I tried to play with alsa-lib .conf files, no luck (I really have no idea how this works). Any ideas on how to add an alias? Thanks for your feedback -Pierre

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[PATCH] ASoC: Add optional prefix_map to snd_soc_card for prefixing codec kcontrols
by Jarkko Nikula 25 Aug '10

25 Aug '10

This optional prefix_map allows to specify unique ALSA control name prefixes for maching codec names. This makes possible to have a sound card configuration using multiple codecs without a name collision that would occur if a codec driver tries to register a kcontrol with an existing name. Now it is possible to specify for instance "Front" and "Rear" prefixes and a sound card can have two separate controls like "Front PCM Playback Volume" and "Rear PCM Playback Volume". Those controls will then show as "Front PCM" and "Rear PCM" in ALSA mixer application. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- include/sound/soc.h | 10 ++++++++++ sound/soc/soc-core.c | 35 +++++++++++++++++++++++++++++++++-- sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++------ 3 files changed, 65 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d31e8b7..cc309af 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -423,6 +423,7 @@ struct snd_soc_ops { /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; + const char *kcontrol_prefix; int id; struct device *dev; struct snd_soc_codec_driver *driver; @@ -553,6 +554,11 @@ struct snd_soc_dai_link { struct snd_soc_ops *ops; }; +struct snd_soc_prefix_map { + const char *codec_name; + const char *kcontrol_prefix; +}; + /* SoC card */ struct snd_soc_card { const char *name; @@ -587,6 +593,10 @@ struct snd_soc_card { struct snd_soc_pcm_runtime *rtd; int num_rtd; + /* optional map of name prefixes that are associated per codec */ + struct snd_soc_prefix_map *prefix_map; + int num_prefixes; + struct work_struct deferred_resume_work; /* lists of probed devices belonging to this card */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 65352c7..b17596f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1266,6 +1266,23 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) static void rtd_release(struct device *dev) {} +static void soc_set_name_prefix(struct snd_soc_card *card, + struct snd_soc_codec *codec) +{ + int i; + + if (card->prefix_map == NULL) + return; + + for (i = 0; i < card->num_prefixes; i++) { + struct snd_soc_prefix_map *map = &card->prefix_map[i]; + if (map->codec_name && !strcmp(codec->name, map->codec_name)) { + codec->kcontrol_prefix = map->kcontrol_prefix; + break; + } + } +} + static int soc_probe_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; @@ -1306,6 +1323,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the CODEC */ if (!codec->probed) { + soc_set_name_prefix(card, codec); if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { @@ -1895,14 +1913,27 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; + char prefixed_name[44], *name; + const char *prefix = NULL; int err, i; + /* Only codec controls are prefixed */ + if (!codec->probed) + prefix = codec->kcontrol_prefix; + for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (prefix) { + snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", + prefix, control->name); + name = prefixed_name; + } else { + name = control->name; + } + err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s\n", - codec->name, control->name); + codec->name, name); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 035cab8..412f0bf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -327,6 +327,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, int i, ret = 0; size_t name_len; struct snd_soc_dapm_path *path; + char prefix[10]; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -347,6 +348,13 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, name_len = strlen(w->kcontrols[i].name) + 1; if (w->id != snd_soc_dapm_mixer_named_ctl) name_len += 1 + strlen(w->name); + if (codec->kcontrol_prefix) { + name_len += 1 + strlen(codec->kcontrol_prefix); + snprintf(prefix, sizeof(prefix), "%s ", + codec->kcontrol_prefix); + } else { + prefix[0] = '\0'; + } path->long_name = kmalloc(name_len, GFP_KERNEL); @@ -355,12 +363,12 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, switch (w->id) { default: - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + snprintf(path->long_name, name_len, "%s%s %s", + prefix, w->name, w->kcontrols[i].name); break; case snd_soc_dapm_mixer_named_ctl: - snprintf(path->long_name, name_len, "%s", - w->kcontrols[i].name); + snprintf(path->long_name, name_len, "%s%s", + prefix, w->kcontrols[i].name); break; } @@ -388,6 +396,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, { struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; + char prefixed_name[44], *name; int ret = 0; if (!w->num_kcontrols) { @@ -395,7 +404,14 @@ static int dapm_new_mux(struct snd_soc_codec *codec, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (codec->kcontrol_prefix) { + snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", + codec->kcontrol_prefix, w->name); + name = prefixed_name; + } else { + name = w->name; + } + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, name); ret = snd_ctl_add(codec->card->snd_card, kcontrol); if (ret < 0) goto err; @@ -406,7 +422,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, return ret; err: - printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); + printk(KERN_ERR "asoc: failed to add kcontrol %s\n", name); return ret; } -- 1.7.1

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[alsa-devel] need help understanding overrun/underrun
by Harsha, Priya 24 Aug '10

24 Aug '10

Hi I had modified the arecord application to call snd_pcm_writei() with the output of snd_pcm_readi(). Basically a loopback. When we execute, I see the calls return EPIPE (lot of overruns and a few underruns). Can anyone help me understand why there are underrun/overrun being printed? Is it because of the serial call of reads and writes? Is there any issue with the driver I am using? Can you suggest some ways to overcome these messages? Thanks, Harsha

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[alsa-devel] Install Infrasonic Amon on Ubuntu?
by Alberto Molina Díaz 24 Aug '10

24 Aug '10

I have a Infrasonic Amon usb audio interface, but I Can't use it in ubuntu beacuse I can't install it. The id of this sound card this is 1a4e:7021 ¿Anyone know how to install it? Thanks

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