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[alsa-devel] No sound with ice_1724 when I restart the system since I upgrade to 2.6.26 (+workaround)
by Amaury De Ganseman 20 Jul '08

20 Jul '08

Hi all, I think it's better to send here ;-) Additional infos: Kernel 2.6.26 from kernel.org /proc/asound/cards: 0 [Juli ]: ICE1724 - ESI Juli@ ESI Juli@ at 0xb400, irq 18 + /var/lib/alsa/asound.state in attach (after stop alsasound, rm asound.state and then restart alsa --> so sound is working) ---------- Forwarded message ---------- From: Amaury De Ganseman <amaury.deganseman(a)gmail.com> Date: Tue, Jul 15, 2008 at 9:50 PM Subject: No sound when I restart the system since I upgrade to 2.6.26 (+workaround) To: alsa-user(a)lists.sourceforge.net Hi all, Here is the description and workaround to have sound. It's related to mixer..but I don't know why it saves wrong mixer settings. Thanks for help #I have this problem (no sound when I start the system) since I upgrade to kernel 2.6.26 (driver: ice1724) #rm asound.state helps but after stopping alsa #This is in /var/lib/alsa: localhost alsa # rm asound.state rm: remove regular file `asound.state'? y localhost alsa # /etc/init.d/alsasound restart * WARNING: you are stopping a boot service. * Stopping pulseaudio ... [ ok ] * Storing ALSA Mixer Levels ... [ ok ] * Killing processes using ALSA ... [ ok ] * Unloading ALSA modules ... [ ok ] * Loading ALSA modules ... * Could not detect custom ALSA settings. Loading all detected alsa drivers. * Loading: snd_ice1724 ... [ ok ] * Restoring Mixer Levels ... alsactl: set_control:1300: incompatible field type for control #1 alsactl: set_control:1300: incompatible field type for control #32 [ ok ] * Starting pulseaudio ... #No sound with amarok (I restart amarok) [ ok ] localhost alsa # /etc/init.d/alsasound stop * WARNING: you are stopping a boot service. * Stopping pulseaudio ... [ ok ] * Storing ALSA Mixer Levels ... [ ok ] * Killing processes using ALSA ... [ ok ] * Unloading ALSA modules ... [ ok ] localhost alsa # rm asound.state rm: remove regular file `asound.state'? y localhost alsa # /etc/init.d/alsasound start * Loading ALSA modules ... * Could not detect custom ALSA settings. Loading all detected alsa drivers. * Loading: snd_ice1724 ... [ ok ] * Restoring Mixer Levels ... * No mixer config in /var/lib/alsa/asound.state, you have to unmute your card! [ ok ] #So I raise up the level of PCM: localhost alsa # alsamixer localhost alsa # /etc/init.d/pulseaudio start * Starting pulseaudio ... [ ok ] localhost alsa # #Now I have sound....

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[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 19 Jul '08

19 Jul '08

This didn't appear to make it through to the list when I tried a few days ago to send it, so I'll try again. --- It's been about 7 months since I last brought this up, and didn't really get any success at the time, so I thought I'd check and see if anybody has any insights again. About a year ago I posted my original request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak

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[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 19 Jul '08

19 Jul '08

It's been about 7 months since I last brought this up, and didn't really get any success at the time, so I thought I'd check and see if anybody has any insights again. About a year ago I posted my original request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 Jul '08

> From: steshenko_sergei(a)list.ru > To: b3zdomny(a)hotmail.com > CC: rene.herman(a)keyaccess.nl; tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; alsa(a)sc8-sf-spam2.sourceforge.net > Subject: Re[2]: [Alsa-user] is this card supported by ALSA? > Date: Fri, 18 Jul 2008 23:47:45 +0400 <snip> > > Were you in the correct subdirectory when running 'patch' ? > > --Sergei. Yes, I was. Thanks for confirming this. best regards, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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[alsa-devel] ASOC, master clock direction
by Jon Smirl 19 Jul '08

19 Jul '08

The MPC5200 has three clock configurations 1) Clock out, normal master mode 2) Clock in, normal slave mode 3) Something Freescale calls cellphone slave but it is more like clock distribution. In cellslave mode one i2s port is in normal slave mode. The clock from this port is used to drive up to three other i2s ports in master mode. I have two choices for set_sysclk: #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 When the clock frequency changes I need to notify the i2s drivers for the dependent channels. Do we need a third choice, SND_SOC_CLOCK_SLAVE? Alternatively I can use another clock id value. -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 Jul '08

> Date: Wed, 16 Jul 2008 16:52:39 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; ALSA(a)sc8-sf-spam2.sourceforge.net > Subject: Re: [Alsa-user] is this card supported by ALSA? > > On 16-07-08 16:44, Rene Herman wrote: > > > On 16-07-08 16:40, Rene Herman wrote: > > > >> # moodprobe -r ens1371-ac97_reset_hack.diff > >> # moodprobe -r ens1371-ac97.diff > > > > I'm getting really sick of the fact that I seem to need to make at least > > one typo or other small error per post. 'modprobe' ofcourse. > > Oh, I so love myself. Try this instead: > > # patch -p1 -r < ens1371-ac97_reset_hack.diff > # patch -p1 -r < ens1371-ac97.diff > > I'll make sure I get some sleep before I post again. > > Rene. OK, this is interesting: # patch -p1 -r < ens1371-ac97_reset_hack.diff= bash: ens1371-ac97_reset_hack.diff: No such file or directory # patch -p1 -r < ens1371-ac97.diff bash: ens1371-ac97.diff: No such file or directory Go figure. Attached are two high-res pix of the card. The smallest chips were just too darned small for me to read the printing; maybe you can magnify the pix and make something of them. Thanks again for all your help. I think I'm just going to be happy with OSS for awhile and see if I can't get it to support Skype. If you decide to keep working on this and you want to test something, you're welcome to contact me. best regards as always, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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[alsa-devel] [PATCH] ASoC: Allow codecs to override register display
by Mark Brown 18 Jul '08

18 Jul '08

Some codecs have unusual features in their register maps such as very large registers representing arrays of coefficients. Support these codecs in the register cache sysfs file by allowing them to provide a function register_display() overriding the default output for register contents. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 14 ++++++++++---- 2 files changed, 11 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1890d87..8b1a3e7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -410,6 +410,7 @@ struct snd_soc_codec { void *control_data; /* codec control (i2c/3wire) data */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + int (*display_register)(struct snd_soc_codec *, char *, unsigned int); hw_write_t hw_write; hw_read_t hw_read; void *reg_cache; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83f1190..49532d9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -970,10 +970,16 @@ static ssize_t codec_reg_show(struct device *dev, step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) - count += sprintf(buf + count, "%2x: %4x\n", i, - codec->read(codec, i)); - + for (i = 0; i < codec->reg_cache_size; i += step) { + count += sprintf(buf + count, "%2x: ", i); + if (codec->display_register) + count += codec->display_register(codec, + buf + count, i); + else + count += sprintf(buf + count, "%4x", + codec->read(codec, i)); + count += sprintf(buf + count, "\n"); + } return count; } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); -- 1.5.6.3

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[alsa-devel] [PATCH] send both an uncork and a trigger in _start()
by Lennart Poettering 18 Jul '08

18 Jul '08

Heya! Here's a patch for alsa-plugins: When playing very short streams, the pulse plugin needs call pa_stream_trigger() in snd_pcm_start() to make sure the stream is actually started, in addition to uncorking the stream. Lennart --- pulse/pcm_pulse.c | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index efff509..7edd157 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -67,8 +67,8 @@ static void update_ptr(snd_pcm_pulse_t *pcm) static int pulse_start(snd_pcm_ioplug_t *io) { snd_pcm_pulse_t *pcm = io->private_data; - pa_operation *o; - int err = 0; + pa_operation *o, *u; + int err = 0, err_o = 0, err_u = 0; assert(pcm); assert(pcm->p); @@ -88,11 +88,20 @@ static int pulse_start(snd_pcm_ioplug_t *io) goto finish; } - err = pulse_wait_operation(pcm->p, o); + u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, pcm->p); + if (!u) { + pa_operation_unref(o); + err = -EIO; + goto finish; + } + + err_o = pulse_wait_operation(pcm->p, o); + err_u = pulse_wait_operation(pcm->p, u); pa_operation_unref(o); + pa_operation_unref(u); - if (err < 0) { + if (err_o < 0 || err_u < 0) { err = -EIO; goto finish; } else @@ -543,7 +552,7 @@ static int pulse_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) break; } } - + pa_threaded_mainloop_lock(pcm->p->mainloop); pcm->frame_size = (snd_pcm_format_physical_width(io->format) * io->channels) / 8; @@ -761,7 +770,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; pcm->io.private_data = pcm; - + err = snd_pcm_ioplug_create(&pcm->io, name, stream, mode); if (err < 0) goto error; -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 3/3] fix return value of snd_pcm_rewind()/snd_pcm_forward() to return how much actually has been rewound, instead of what actually could have rewound
by Lennart Poettering 18 Jul '08

18 Jul '08

Make snd_pcm_plugin_rewind()/_forward() actually return how much has been rewound/forwarded instead of how much could have been rewounded/forwarded. This makes the code actually do what the documentation of snd_pcm_rewind() suggests. Signed-off-by: Lennart Poettering <lennart(a)poettering.net> --- src/pcm/pcm_plugin.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index b377cb2..c4f5b4a 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -222,7 +222,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames = plugin->client_frames(pcm, sframes); snd_pcm_mmap_appl_backward(pcm, (snd_pcm_uframes_t) frames); snd_atomic_write_end(&plugin->watom); - return n; + return (snd_pcm_sframes_t) frames; } static snd_pcm_sframes_t snd_pcm_plugin_forwardable(snd_pcm_t *pcm) @@ -255,7 +255,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ frames = plugin->client_frames(pcm, sframes); snd_pcm_mmap_appl_forward(pcm, (snd_pcm_uframes_t) frames); snd_atomic_write_end(&plugin->watom); - return n; + return (snd_pcm_sframes_t) frames; } static snd_pcm_sframes_t snd_pcm_plugin_write_areas(snd_pcm_t *pcm, -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 2/3] fix type of internally used sframes variable, to avoid unnecessary casts
by Lennart Poettering 18 Jul '08

18 Jul '08

This minor patch fixes the type of the sframes variable in snd_pcm_plugin_forward(). With this fix we need to cast less and the code is less confusing. Signed-off-by: Lennart Poettering <lennart(a)poettering.net> --- src/pcm/pcm_plugin.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index c199d8d..b377cb2 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -234,7 +234,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ { snd_pcm_plugin_t *plugin = pcm->private_data; snd_pcm_sframes_t n = snd_pcm_mmap_avail(pcm); - snd_pcm_uframes_t sframes; + snd_pcm_sframes_t sframes; if ((snd_pcm_uframes_t)n < frames) frames = n; @@ -246,8 +246,8 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ else sframes = frames; snd_atomic_write_begin(&plugin->watom); - sframes = INTERNAL(snd_pcm_forward)(plugin->gen.slave, (snd_pcm_uframes_t) sframes); - if ((snd_pcm_sframes_t) sframes < 0) { + sframes = INTERNAL(snd_pcm_forward)(plugin->gen.slave, sframes); + if (sframes < 0) { snd_atomic_write_end(&plugin->watom); return sframes; } -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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