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[alsa-devel] Analog Loopback controls disabled for IDT/STAC codecs
by Takashi Iwai 02 Mar '09

02 Mar '09

Hi, I disabled now "Analog Loopback" controls for IDT/STAC codecs on sound git tree as default since these are rather confusing for users, more harmful than useful. But the feature is still available. It can be enabled via sysfs "hints". For example, # echo "loopback = yes" > /sys/class/sound/hwC0D0/hints # echo 1 > /sys/class/sound/hwC0D0/reconfig will take the mixer elements back. Also, hp_detect and gpio-bits can be changed similarly on the fly. If you find any problems with this change, please let me know. thanks, Takashi

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[alsa-devel] [PATCH] cs4270: add support for slave mode configurations
by Daniel Mack 02 Mar '09

02 Mar '09

Added support for scenarios where the Cirrus CS4270 audio codec is slave to the bitclk and lrclk. Mixed setups are unsupported. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/codecs/cs4270.c | 24 +++++++++++++++++++++++- 1 files changed, 23 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index cd4a9ee..339e0f6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { u8 reg_cache[CS4270_NUMREGS]; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ + unsigned int slave_mode; }; /** @@ -247,6 +248,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, struct cs4270_private *cs4270 = codec->private_data; int ret = 0; + /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -257,6 +259,21 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, ret = -EINVAL; } + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4270->slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4270->slave_mode = 0; + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* unsupported - cs4270 can eigther be slave or master to + * both the bitclk and the lrclk. */ + default: + ret = -EINVAL; + } + return ret; } @@ -399,7 +416,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); - reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + reg |= cs4270_mode_ratios[i].mclk; + + if (cs4270->slave_mode) + reg |= CS4270_MODE_SLAVE; + else + reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { -- 1.6.1.3

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[alsa-devel] Support for fujitsu T421x
by John Kha 02 Mar '09

02 Mar '09

I have a Fujitsu Lifebook T4210 (very similar to the T4215) and wish to work on adding support for these models in patch_sigmatel.c. I have ran the alsa-info script, once after a fresh boot, and once after changing the model a few times: Virgin: http://www.alsa-project.org/db/?f=e76430dbf61b0e2c1ff199c9d1c85051a0341508 With Model Changed: http://www.alsa-project.org/db/?f=cdd45e343c143052e18f9706e65eef8865696606 After trying various models, I have been unable to get the audio to output to the correct pin groups. I have managed to use hda_analyzer.py to change some of my hardware settings, and have an idea of what verbs i need to send. What I need is some information on the architecture of the driver, or an idea of where to edit things. I would just say the config here and ask someone else to edit patch_sigmatel.c, but this is only the first of the issues I want to fix, so some knowledge of how to edit myself would be useful. -- John C. Kha United States Navy

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[alsa-devel] arecord channel limitation
by Tim Blechmann 02 Mar '09

02 Mar '09

hi all, is there any reason, why arecord limits the channel count to 32? best, tim -- tim(a)klingt.org http://tim.klingt.org Desperation is the raw material of drastic change. Only those who can leave behind everything they have ever believed in can hope to escape. William S. Burroughs

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[alsa-devel] Status of the alsa sbxfi driver
by Brandon Ehle 02 Mar '09

02 Mar '09

I merged the topic/sbxfi branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git into my latest 2.6.29-rc6 build. I've done a few quick tests and I have not run into any issues so far (except the driver has an excessive amount of kprintfs). Is this the latest version of the sbxfi driver? _From following the alsa-devel list and the git tree above, there doesn't seem to be any changes since Oct 30 or so. What work needs to be done to get this driver into a state where it could be pushed to the main alsa git and eventually to Linus' tree?

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[alsa-devel] [PATCH] Arcam AV Control Plugin Update
by Peter Stokes 02 Mar '09

02 Mar '09

arcam_av: Refactor event callback to facilitate addition of "numid" optimisation Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk>

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[alsa-devel] [PATCH] opl3sa2 fix irq releasing and short name of card
by Krzysztof Helt 02 Mar '09

02 Mar '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Two simple fixes: 1. Use the same pointer for the free_irq() and the request_irq() calls. 2. A short name of card is appended with '2' or '3' character depending on a detected chip. Remove the '2' character from the short name. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The short name without the patch is "OPL3-SA23" for opl3-sa3. sound/isa/opl3sa2.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b..9a53ab4 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -613,7 +613,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -626,7 +626,7 @@ static struct snd_card *snd_opl3sa2_card_new(int dev) if (card == NULL) return NULL; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; -- 1.5.2.2 ---------------------------------------------------------------------- 169 stacji TV! Naziemne i satelitarne! Sprawdz program TV w kilka sekund >> http://link.interia.pl/f2050

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[alsa-devel] pointer callback question
by Tim Blechmann 28 Feb '09

28 Feb '09

hi all, one question about the pointer callback. the docs say: > This callback is called when the PCM middle layer inquires the current > hardware position on the buffer. The position must be returned in frames, > ranging from 0 to buffer_size - 1. i am using a ringbuffer to communicate with my audio device. but when the position is reset to 0, alsa requests to stop the stream (SNDRV_PCM_TRIGGER_STOP). the brief log is (4 periods of 64 frames): stream_pointer at 64 stream_pointer at 128 stream_pointer at 192 stream_pointer at 256 stream_pointer at 0 after returning 0, a stop is requested. am i missing something? thanks, tim -- tim(a)klingt.org http://tim.klingt.org Contrary to general belief, an artist is never ahead of his time but most people are far behind theirs. Edgar Varèse

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[alsa-devel] Xonar U1 volume controls
by Elie Morisse 28 Feb '09

28 Feb '09

Hi! I'm answering to cladisch call for the "lsusb -v" ouput for this device on Asus forums: http://vip.asus.com/forum/view.aspx?id=20090105132008643&board_id=21&model=… us The analog ouput and microphone input work flawlessly with alsa-driver-1.0.19 but the volume controls don't, although they are listed in evtest pushing or rotating the knob(X not running) does nothing : $ sudo evtest /dev/input/event2 Input driver version is 1.0.0 Input device ID: bus 0x3 vendor 0xb05 product 0x1743 version 0x100 Input device name: "C-Media Electronics Inc. USB Advanced Audio Device" Supported events: Event type 0 (Reset) Event code 0 (Reset) Event code 1 (Key) Event code 4 (?) Event type 1 (Key) Event code 113 (Mute) Event code 114 (VolumeDown) Event code 115 (VolumeUp) Event type 4 (?) Event code 4 (?) Testing ... (interrupt to exit) Besides it also has a LED that can "pulse" and can be blue, purple or red, and there doesn't seem to be any way to control this either. On Vista the Asus driver defaults it to pulsing blue when ouputting sound while on Linux it blinks alternatively purple and blue, how ugly and less relaxing!.. Hope the lsusb output will be useful and thanks a lot for all your hard work, Elie

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[alsa-devel] E-MU 0404 Support
by Kristian Lumme 27 Feb '09

27 Feb '09

Hi! I reported a bug (or what I think is a bug) with the E-MU 0404 USB driver (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4410) (this probably concerns other E-MU cards as well). Now, the supported soundcards list states that E-MU (or Creative) have supplied ALSA developers with datasheets to help the development of the driver. Well... in my bug report I was referred to this discussion on the ALSA devel list: http://thread.gmane.org/gmane.linux.alsa.devel/56288/focus=56355 If I understand this correctly, people that are offering to help develop the driver can't get in touch with the person who has the datasheets (and there hasn't been any update from him in a long time... or so it seems--sorry if I'm mistaken). Surely, there's some way to make these specifications available to interested developers so the support for the E-MU cards can be improved?

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