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[alsa-devel] Question about "control names"
by Juergen Beisert 23 May '08

23 May '08

Hi, I'm playing with the control names of my "snd_kcontrol_new" controls. When I'm using "PCM Playback Switch" and "PCM Playback Volume" alsamixer shows me a slider with a switch at its bottom. That's what I expected. But when I'm using a second set ("line in" to "line out" bypass), with "Line Bypass Capture Switch" and "Line Bypass Capture Volume" control names, its confusing (BTW: "Line Bypass Capture Volume" is of type SNDRV_CTL_ELEM_TYPE_ENUMERATED). The "Line Bypass Capture Volume" control is shown on the [Playback] page with the name "Line Byp" and the default value. I can walk through the enumerated values. But the "Line Bypass Capture Switch" is shown on the [Capture] page in alsamixer as: L R CAPTUR <Line Byp> What does it mean? Problem of alsamixer or the name of this set? Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

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[alsa-devel] softvol plug-in dependancies?
by Simon Dunn 23 May '08

23 May '08

Hi Guys The softvol plug-in, are there any pre-requisites or dependancies to using this? I've tried various entries in the /etc/asound.conf configuration file all with the same result. I can see the control in alsamixer, once I've run the speaker-test through it, but cannot adjust any settings. Trying to run 'alsactl store 0' results in this error : . alsactl: control.c:2437: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof (obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. Aborted . Curiously using amixer the device is listed but its properties are displayed like this . Simple mixer control 'PCM Vol',0 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front C enter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Capture channels: Front Left - Front Right - Rear Left - Rear Right - Front Ce nter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Front Center: 0 [0%] Woofer: 0 [0%] Side Left: 0 [0%] Side Right: 0 [0%] Rear Center: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] Any clues as to what's going on here would be much appreciated before I have to start dissecting the library. TIA for any hints Simon -- Simon Dunn, Software Engineer MPC Data Limited e-mail: sdunn(a)mpc-data.co.uk web: www.mpc-data.co.uk tel: +44 (0) 1225 710600 fax: +44 (0) 1225 710601 ddi: +44 (0) 1225 710631 MPC Data Limited is a company registered in England and Wales with company number 05507446 Registered Address: County Gate, County Way, Trowbridge, Wiltshire, BA14 7FJ VAT no: 850625238 The information in this email and in the attached documents is confidential and may be legally privileged. Any unauthorized review, copying, disclosure or distribution is prohibited and may be unlawful. It is intended solely for the addressee. Access to this email by anyone else is unauthorized. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. When addressed to our clients any opinions or advice contained in this email is subject to the terms and conditions expressed in the governing contract.

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[alsa-devel] HG -> GIT migration
by Jaroslav Kysela 23 May '08

23 May '08

Hi all, I finished migration from HG to GIT. All repositories are converted and we have one "new" repository - alsa-kernel.git which is full Linux 2.6 GIT tree. The old HG alsa-kernel repository was renamed to alsa-kmirror - just to track our changes in continuous timeline for releases and packaging. HG access to all repositories is closed now. More information can be obtained here: http://www.alsa-project.org/main/index.php/GIT_Server Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] [Alsa-user] Pc Speaker
by Rene Herman 23 May '08

23 May '08

On 13-05-08 12:44, Armin ranjbar wrote: >> Device Drivers -> >> Input device support -> >> Miscelaneous devices >> >> With current mainline, it's: >> >> General setup -> >> Configure standard kernel features (for small systems) >> Enable PC-Speaker support > > Thank you very much for your reply , > > but i have to say that i have this options activated , still no pc > speaker . Rechecked. Yes, you need this option, you need the ALSA one disabled (not modular, disabled) and then the PC speaker driver option appears in its input menu where you want to enable it. Paths as given in last reply. Yes, that would seem to be amazingly clumsily done. Perhaps there was a reason (adding alsa-devel). Rene.

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[alsa-devel] [PATCH] Patch for alsa-info.sh to check if pastebin.ca is down, and handle with grace.
by Travis Place 22 May '08

22 May '08

Changelog: This patch will allow alsa-info.sh to check if the pastebin site (www.pastebin.ca) is down, before trying to upload to it. It will fail gracefully, with a message letting the user know what happened. The script will also keep the ALSA information it collected (same way it does with --no-upload). Patch is attached. Signed-off-by: Travis Place <wishie at wishie dot net> -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] Moving sound/* to drivers/ ?
by Takashi Iwai 22 May '08

22 May '08

While we are at the issue ALSA HG <-> GIT transition... One thing that annoys me sometimes is that the sound driver codes are on a different subdirectory than other normal drivers, namely under /sound. I don't remember exactly why it came there, but this often makes people forget to change the necessary code for the sound subsystem, or slip from the statistics (I'm not sure whether Linus counted sound/* with recent his posts). If I understand correctly, with git, we can move the files in relatively little costs. So, what about moving sound/* back to drivers/sound/* or drivers/media/sound/*? Of course, the primary question is whether it's really worth. The obvious drawback is that patches won't be applicable after the move. So, if we do it, doing at the last seems practical. But, this can be a bit problem with keeping such a change on linux-next until the next merge... And, another question is to where. drivers/sound or drivers/media/sound, or whatever. Any comments appreciated. thanks, Takashi

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Re: [alsa-devel] [pulseaudio-discuss] Asynchronous notification in ALSA - I/O Plugin - Pulseaudio
by Lennart Poettering 22 May '08

22 May '08

On Thu, 22.05.08 14:11, Colin Guthrie (gmane(a)colin.guthr.ie) wrote: > > That freeze is an unfixable race condition that Flash has when > > shutting down audio. > > > > See the explanation down on http://pulseaudio.org/ticket/267 > > I thought the whole point was that by adding async code to ioplug, the > libflashsupport stuff wasn't needed and flash could just use it's native > alsa support + alsa's pulse plugin.... or am I missing the point here? The destruction dead lock issue is present both in libflashsupport and the native ALSA backend. It's just that it is almost never triggered if a hw device is used. Adobe released the sources of libflashsupport where you can actually check this. 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] Add async support to ioplug
by Takashi Iwai 22 May '08

22 May '08

Hi, the patch below adds a framework to support async on ioplug. Now an external ioplug plugin can implement appropriate async methods if possible. The async callback is just a copy of snd_pcm_async(). When sig >= 0, activate the async, and with a negative value, deactivate async. The test patch to alsa-plugins will follow. Takashi --- diff -r 8b22022a861d include/local.h --- a/include/local.h Fri May 09 16:02:02 2008 +0200 +++ b/include/local.h Tue May 20 09:02:35 2008 +0200 @@ -149,7 +149,13 @@ void *private_data; struct list_head glist; struct list_head hlist; + int no_signal; }; + +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal); +void _snd_async_call_handlers(int fd); typedef enum _snd_set_mode { SND_CHANGE, diff -r 8b22022a861d include/pcm_ioplug.h --- a/include/pcm_ioplug.h Fri May 09 16:02:02 2008 +0200 +++ b/include/pcm_ioplug.h Tue May 20 09:02:35 2008 +0200 @@ -66,7 +66,7 @@ */ #define SND_PCM_IOPLUG_VERSION_MAJOR 1 /**< Protocol major version */ #define SND_PCM_IOPLUG_VERSION_MINOR 0 /**< Protocol minor version */ -#define SND_PCM_IOPLUG_VERSION_TINY 1 /**< Protocol tiny version */ +#define SND_PCM_IOPLUG_VERSION_TINY 2 /**< Protocol tiny version */ /** * IO-plugin protocol version */ @@ -114,6 +114,10 @@ unsigned int rate; /**< rate; filled after hw_params is called */ snd_pcm_uframes_t period_size; /**< period size; filled after hw_params is called */ snd_pcm_uframes_t buffer_size; /**< buffer size; filled after hw_params is called */ + /** + * fields added by version 1.0.2 + */ + unsigned int no_signal_async; /**< async support without signal */ }; /** Callback table of ioplug */ @@ -189,6 +193,10 @@ * get the delay for the running PCM; optional */ int (*delay)(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp); + /** + * set up async handler; optional; added in version 1.0.2 + */ + int (*async)(snd_pcm_ioplug_t *io, int sig, pid_t pid); }; @@ -212,6 +220,9 @@ /* change PCM status */ int snd_pcm_ioplug_set_state(snd_pcm_ioplug_t *ioplug, snd_pcm_state_t state); +/* call async handlers */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug); + /** \} */ #endif /* __ALSA_PCM_IOPLUG_H */ diff -r 8b22022a861d src/async.c --- a/src/async.c Fri May 09 16:02:02 2008 +0200 +++ b/src/async.c Tue May 20 09:02:35 2008 +0200 @@ -48,19 +48,48 @@ #endif static LIST_HEAD(snd_async_handlers); +static int sig_count; -static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +#ifndef DOC_HIDDEN +void _snd_async_call_handlers(int fd) { - int fd; struct list_head *i; - //assert(siginfo->si_code == SI_SIGIO); - fd = siginfo->si_fd; list_for_each(i, &snd_async_handlers) { snd_async_handler_t *h = list_entry(i, snd_async_handler_t, glist); if (h->fd == fd && h->callback) h->callback(h); } } +#endif + +static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +{ + int fd; + //assert(siginfo->si_code == SI_SIGIO); + fd = siginfo->si_fd; + _snd_async_call_handlers(fd); +} + +#ifndef DOC_HIDDEN +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal) +{ + snd_async_handler_t *h; + assert(handler); + h = malloc(sizeof(*h)); + if (!h) + return -ENOMEM; + h->fd = fd; + h->callback = callback; + h->private_data = private_data; + list_add_tail(&h->glist, &snd_async_handlers); + INIT_LIST_HEAD(&h->hlist); + h->no_signal = no_signal; + *handler = h; + return 0; +} +#endif /** * \brief Registers an async handler. @@ -94,20 +123,13 @@ int snd_async_add_handler(snd_async_handler_t **handler, int fd, snd_async_callback_t callback, void *private_data) { - snd_async_handler_t *h; - int was_empty; - assert(handler); - h = malloc(sizeof(*h)); - if (!h) - return -ENOMEM; - h->fd = fd; - h->callback = callback; - h->private_data = private_data; - was_empty = list_empty(&snd_async_handlers); - list_add_tail(&h->glist, &snd_async_handlers); - INIT_LIST_HEAD(&h->hlist); - *handler = h; - if (was_empty) { + int err; + + err = _snd_async_add_handler(handler, fd, callback, private_data, 0); + if (err < 0) + return err; + + if (!sig_count) { int err; struct sigaction act; memset(&act, 0, sizeof(act)); @@ -116,10 +138,15 @@ sigemptyset(&act.sa_mask); err = sigaction(snd_async_signo, &act, NULL); if (err < 0) { + err = -errno; SYSERR("sigaction"); - return -errno; + list_del(&(*handler)->glist); + free(*handler); + *handler = NULL; + return err; } } + sig_count++; return 0; } @@ -133,15 +160,18 @@ int err = 0; assert(handler); list_del(&handler->glist); - if (list_empty(&snd_async_handlers)) { - struct sigaction act; - memset(&act, 0, sizeof(act)); - act.sa_flags = 0; - act.sa_handler = SIG_DFL; - err = sigaction(snd_async_signo, &act, NULL); - if (err < 0) { - SYSERR("sigaction"); - return -errno; + if (!handler->no_signal) { + sig_count--; + if (!sig_count) { + struct sigaction act; + memset(&act, 0, sizeof(act)); + act.sa_flags = 0; + act.sa_handler = SIG_DFL; + err = sigaction(snd_async_signo, &act, NULL); + if (err < 0) { + SYSERR("sigaction"); + /* return -errno; */ + } } } if (handler->type == SND_ASYNC_HANDLER_GENERIC) diff -r 8b22022a861d src/pcm/pcm.c --- a/src/pcm/pcm.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm.c Tue May 20 09:02:36 2008 +0200 @@ -1984,8 +1984,10 @@ int err; int was_empty; snd_async_handler_t *h; - err = snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), - callback, private_data); + + err = _snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), + callback, private_data, + pcm->no_signal_async); if (err < 0) return err; h->type = SND_ASYNC_HANDLER_PCM; diff -r 8b22022a861d src/pcm/pcm_ioplug.c --- a/src/pcm/pcm_ioplug.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_ioplug.c Tue May 20 09:02:36 2008 +0200 @@ -698,11 +698,15 @@ return 0; } -static int snd_pcm_ioplug_async(snd_pcm_t *pcm ATTRIBUTE_UNUSED, - int sig ATTRIBUTE_UNUSED, - pid_t pid ATTRIBUTE_UNUSED) +static int snd_pcm_ioplug_async(snd_pcm_t *pcm, int sig, pid_t pid) { - return -ENOSYS; + ioplug_priv_t *io = pcm->private_data; + + if (io->data->version >= 0x010002 && + io->data->callback->async) + return io->data->callback->async(io->data, sig, pid); + else + return -ENOSYS; } static int snd_pcm_ioplug_munmap(snd_pcm_t *pcm ATTRIBUTE_UNUSED) @@ -1037,6 +1041,9 @@ ioplug->pcm->poll_events = ioplug->poll_events; ioplug->pcm->monotonic = (ioplug->flags & SND_PCM_IOPLUG_FLAG_MONOTONIC) != 0; ioplug->pcm->mmap_rw = ioplug->mmap_rw; + if (ioplug->version >= 0x010002) + ioplug->pcm->no_signal_async = ioplug->no_signal_async; + return 0; } @@ -1070,3 +1077,16 @@ ioplug->state = state; return 0; } + +/** + * \brief Call the associated async handlers + * \param ioplug the ioplug handle + * + * Call the async handlers associated with the given ioplug. + * This function is usually called from the plugin's async handling + * routine appropriately. + */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug) +{ + _snd_async_call_handlers(ioplug->poll_fd); +} diff -r 8b22022a861d src/pcm/pcm_local.h --- a/src/pcm/pcm_local.h Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_local.h Tue May 20 09:02:36 2008 +0200 @@ -220,6 +220,7 @@ * use the mmaped buffer of the slave */ unsigned int donot_close: 1; /* don't close this PCM */ + unsigned int no_signal_async :1; /* async without signal */ snd_pcm_channel_info_t *mmap_channels; snd_pcm_channel_area_t *running_areas; snd_pcm_channel_area_t *stopped_areas;

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Re: [alsa-devel] [Bug #10679] 2.6.26-rc1 regression: e5e1d3cb20034a3cbcfff1f0bae12201aa2ce17e breaks artsd
by Stas Sergeev 22 May '08

22 May '08

Hello. Dave Jones wrote: > More seriously, it doesn't seem that there's any way we can build > this driver modular without it screwing up already working systems Actually, I have just upgraded f8->f9, and guess what? Lost the sound... And not because of snd-pcsp, but rather because snd-hda-intel started to take the first slot and snd-intel8x0 takes the second. On f8 snd-hda-intel have never been loaded for me at all. I don't need it, it finds a HDA chip on an ATI video board, which it never did before. So... it doesn't look like this problem is specific to snd-pcsp. Anyone can get into that trouble it seems.

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[alsa-devel] No audio from laptop speakers, and very faint audio through headphones
by Adam Wiseman 22 May '08

22 May '08

Note: I originally posted my problem on the alsa-user list, but due to the fact that I've got an intel "integrated-everything" chipset, which alsa apparently seems to be having all sorts of trouble with, I was referred to the devel list. So here's my problem: On my Panasonic Toughbook, I get no sound from the laptop speakers, and just barely audible sound from the headphones. I've googled, searched the ubuntu forums, and chatted with a very helpful individual (wishie) on IRC, to no avail. Here is my pastebin of my alsa-info.sh output: http://pastebin.ca/1021657 Here is my pastebin of my amixer output: http://pastebin.ca/1021660 I would be ridiculously grateful if anyone can help me get the sound working on this thing. Thanks, Adam

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