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

[alsa-devel] CMI8788/AV200/Oxygen Volume Controls in alsamixer (oxygen_mixer.c:dac_volume_get())
by John L. Utz III 27 Jan '09

27 Jan '09

Hi Clemens and all; ( To those following along at home, kindly open alsa-driver-1.0.18a/alsa-kernel/pci/oxygen/oxygen_mixer.c and look at dac_volume_get() and dac_volume_put() ) It appears to me that these functions are attempting to work on all of the channels at once instead of just choosing Master, Front, Rear, Surround, or what have you: static int dac_volume_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; unsigned int i; mutex_lock(&chip->mutex); for (i = 0; i < chip->model.dac_channels; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; } This is then surfaced to userland thusly in the controls array: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Stereo Upmixing", .info = upmix_info, .get = upmix_get, .put = upmix_put, }, { blah }, { blah }, { blah } }; this leads to a rather awkward result in alsamixer where there are 5 volume controls called "Master": < Master > Master Master Master Master IMHO, it would be more desirable for them to say: < Master > Front C/LFE Surround Rear Surr Our app would certainly be much happier :-) So, i *think* that i can fix this thusly: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 1 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 1 }, and alter dac_volume_*() to not use a for() loop and just work off the private_value. anybody have any opinions? i'll probably have tried this already before anybody responds but it seemed reasonable to surface the idea. tnx! johnu

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[alsa-devel] ICE1712 softdevices fixes
by Diego E. 'Flameeyes' Pettenò 26 Jan '09

26 Jan '09

The following patches fixes the default device as they appear for ICE1712-based cards. Without this cards, 'aplay -L' will show a series of devices like surround40 and surround71 that are not meaningful for this card, and a front device that would not work. Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound devices on ICE1712. Make the front:idx device work for ICE1712.

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[alsa-devel] [PATCH] hda: 92hd71xxx disable unmute support for codecs that don't have input amps.
by Matthew Ranostay 26 Jan '09

26 Jan '09

Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write_cache() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c28..ffef091 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- 1.5.6.3

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[alsa-devel] [PATCH] hda: 92hd71xxx disable unmute support for codecs that don't have input amps.
by Matthew Ranostay 26 Jan '09

26 Jan '09

Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c28..ffef091 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- 1.5.6.3

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[alsa-devel] snd_pcm_set_params() bug?
by Christian Morales Vega 26 Jan '09

26 Jan '09

This code: #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } fails with error "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Disable resampling and you get "ALSA lib pcm.c:7142:(snd_pcm_set_params) Rate doesn't match (requested 24000Hz, get 0Hz)" (normal, I suppose). With other samplerates works fine. I'm using alsa-driver-kmp-default-1.0.16.20080602 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems. This was discovered with bsnes ( http://byuu.cinnamonpirate.com/bsnes/ - http://board.zsnes.com/phpBB2/viewtopic.php?t=11584 ) that can use openal, libao or alsa directly. The problem just happens when using 24KHz with alsa. So I suppose the problem is in the snd_pcm_set_params() itself, since openal and libao have not the problem but they set parameters "manually", not with snd_pcm_set_params().

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[alsa-devel] [PATCH] Replace hardcoded hexa formating
by LCID Fire 26 Jan '09

26 Jan '09

The attached patch replaces the hardcoded hexa formats in usbaudio.c commit 079e4efba0603e7a32bf1d352d98c2b265b91cd6 Author: LCID Fire <lcid-fire(a)gmx.net> Date: Fri Jan 23 23:39:06 2009 +0100 Rather use printf format instead of hardcoding it. A next step would be to predefine certain formats. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index bbd70d5..77ac5c9 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1292,14 +1292,14 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n", + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n", + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ } @@ -1468,7 +1468,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); if (!fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", + snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", format, rate, channels); return -EINVAL; } @@ -2160,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: 0x%x\n", fp->format); + snd_iprintf(buffer, " Format: %#x\n", fp->format); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, @@ -2180,7 +2180,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, "\n"); } // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); - // snd_iprintf(buffer, " EP Attribute = 0x%x\n", fp->attributes); + // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } } @@ -2619,7 +2619,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: - snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n", + snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); fp->format = SNDRV_PCM_FORMAT_MPEG; break; @@ -2817,7 +2817,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); err = add_audio_endpoint(chip, stream, fp); if (err < 0) { kfree(fp->rate_table);

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[alsa-devel] opti92x datasheets
by Krzysztof Helt 26 Jan '09

26 Jan '09

Hi, I have found that Regensburg University mirrored Opti ftp site with drivers. The directories contain datasheets as well. Takashi can ask them if these datasheets can be copied into the alsa datasheet archive. The links are here: http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C930/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C929/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C925/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C924/ There are also some additional application notes. Regards, Krzysztof ---------------------------------------------------------------------- Kochanka - kobieta drugiej kategorii? sprawdz >>>http://link.interia.pl/f2026

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[alsa-devel] Question on usbaudio.c: snd_usb_find_csint_desc
by LCID Fire 26 Jan '09

26 Jan '09

I was looking at 'snd_usb_find_csint_desc' and I really wonder why there is a while loop since it seems to me that the variables don't really change. Am I missing something?

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[alsa-devel] Enumerating PCM devices
by Bankim Bhavsar 25 Jan '09

25 Jan '09

Hi Alsa developers/users, I'm using snd_device_name_hint("pcm"..) and then snd_device_name_get_hint(.."NAME") to get the NAME parameter of pcm devices.

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Re: [alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (4th rev)
by Krzysztof Helt 24 Jan '09

24 Jan '09

On Sat, 24 Jan 2009 13:35:28 +0100 Krzysztof Helt <krzysztof.h1(a)poczta.fm> wrote: > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > + .buffer_bytes_max = 0x3000, > + .period_bytes_min = 0x40, > + .period_bytes_max = 0x1800, > + .periods_min = 2, > + .periods_max = 3, I have investigated the max_periods number and at least some code called from the interrupt assumes no more then 3 periods in the buffer. I do not understand the code enough to change this. Regards, Krzysztof ---------------------------------------------------------------------- Zobacz gry internetowe dla dzieci! sprawdz >>>http://link.interia.pl/f202c

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