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[alsa-devel] snd_mixer_selem_channel_id -- should i be using this? if not, what should i use?
by John L. Utz III 11 Apr '07

11 Apr '07

Hello Everybody; mixer.h indicates that snd_mixer_selem_channel_id is a legacy api. so, what is the non legacy api for setting volumes on individual channels, ie Master L&R, PCM L&R and Surround L&R ? tnx! johnu

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[alsa-devel] Interrupts and kmalloc in alsa modules
by Ciaccia 11 Apr '07

11 Apr '07

Hi all, I am working with an ARM embedded system and I porting an alsa application to that system. Cirrus (the manufacturer of the board) wrote a generic alsa driver for "their" kernel (a kernel based on the 2.6.8 with some patches) and I ported the module to the vanilla kernel 2.6.19. The module works fine (play and capture both work), but the function that allocates the memory prints a strange message in the kernel log when the buffers are allocated (the module calls kmalloc, which prints "sleeping function called from invalid context" to the log). The buffers are allocated when the streams are started, not when the module is loaded, I think this is standard... The buffers are allocated with the following line: s->buffers = (audio_buf_t *) kmalloc(sizeof(audio_buf_t) * s->nbfrags, GFP_KERNEL); When the line here above is executed, slab.c prints the error the kernel log. The error comes from the "might_sleep_if" line: static inline void cache_alloc_debugcheck_before(struct kmem_cache *cachep, gfp_t flags) { might_sleep_if(flags & __GFP_WAIT); #if DEBUG kmem_flagcheck(cachep, flags); #endif } Playing with the module, I discovered that if I enable the interrupts before calling kmalloc, the error disappears. I wonder if this is correct, or if something is wrong... Could someone of you explain to me why I have to enable the interrupts before calling kmalloc? The strange thing is that the module never explicitly turns them off (I attached the file to the mail) Any help is greatly appreciated ;-) Bye Andrea ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

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[alsa-devel] sound on asus p5ld2-se
by Andrius 10 Apr '07

10 Apr '07

Hi, It says "If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML" in ALSA-configuration.txt so, here it is: i'm using asus p5ld2-se mobo with intel ich7 chipset, and just have installed new debian stable (4.0). And encountered a problem: onboard audio sound from one of the speakers was just static noise, and if i adjusted volume or anything in alsamixer, the sound kept dissappearing. I tried module=3stack option in modprobe, added line options snd-hda-intel index=0 model=3stack in /etc/modprobe.d/sound and everything is ok now. i attached lspci -nv output to this letter p.s. on ubuntu live cd and other distros i've tried there were same problem

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[alsa-devel] Driver for user-space sound source/sink(s)
by Alex Makarov 10 Apr '07

10 Apr '07

Hi We're considering implementing the driver for user-space sound input/output. For example: "Line in" is provided by some application by streaming data to a file in /dev/..., "Line out" can be read by any application from /dev/... It would be also nice to have arbitrary number of streams, for example to have different applications to "capture" audio from different sources. I wonder if we reinvent the wheel, or maybe there's an ongoing project for the similar driver. If not, could anyone advise how this can be achieved. We had no experience with ALSA before, but pretty strong in C/C++ development. Thanks! Alex

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[alsa-devel] Volume control channels mixed
by Simon van der Linden 10 Apr '07

10 Apr '07

Hello, I use Alsa in the latest Ubuntu bêta, with my MacBook. lspci gives the audio card as: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Sound is working pretty good. However, volume control channels (in alsamixer) are mixed: with laptop speakers: Master controls treble volume Front controls bass volume with headphones: Master has no effect Front controls the volume I'm unfamiliar with Alsa, but I guess it shouldn't be too hard to correct that. Browsing the web didn't give me more informations. Perhaps some of you could help me? -- Simon van der Linden jid: simon(a)vanderlinden.eu.org

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Re: [alsa-devel] [Audacity-devel] distortion with alsa I/O - no distortion with oss emulation
by Simon Lewis 10 Apr '07

10 Apr '07

Dear Alsa developers Audacity seems to work perfectly with AOSS and Jack but not directly with ALSA on some linux PCs. Trying to use ALSA directly leads to severe distortion due to lack of sync with Alsa. (I hope that is the correct terminology.) I received this reply from the Audacity developers, what tools are available from ALSA to investigate this problem? How can I determine what the real "Alsa configuration" is that portaudio has used setting up communication with Alsa. Best Regards, Simon > Markus Meyer schrieb: > > > As for Simon's problems with "synchronisation", I'm not so sure that it > is a Portaudio issue, or a software issue at all. Software issues tend > to be of the "on/off" type, and Portaudio does work well with ALSA in > the general case. Apart from being a hardware issue (defective sound > card, main board or something) this could also well be a race condition > in the ALSA driver (kernel module) which is only triggered by certain > buffer sizes, order of driver requests etc. Because Audacity is > multi-threaded it certainly stresses the driver/kernel subsystem more > than just, say, "arecord", which just gets blocks from the driver and > writes it out to disk. > > > Markus > > >> Simon Lewis schrieb: >> >> The main issue is that on some linux machines, mine included, there is >> severe distortion when Audacity is used with ALSA drivers. >> >> You will be pleased to hear that I have found the root cause of the >> problem. It is nothing to do with the options with which Audacity is >> compiled. The root cause is that portaudio loses synchronisation with >> ALSA's data stream. >> >> I was able to verify this yesterday, as I could start Audacity with >> Jack, and through Jack set up a number of simulation conditions and >> monitor the results. >> >> With my standard Jack settings of 96kHz / 256 Frames per period / 2 >> periods per buffer (latency 5.33 msecs.), ALSA initialises with sample >> format 32bit little-endian, 2 periods for capture and 2 periods for >> playback. >> >> Under these conditions Audacity cannot synchronise with the data stream >> and severe distortion is the result. >> >> However, increasing the frames per period to 1024 (latency 21.3 msec) >> and Audacity (1.3.2 with resampling disabled) runs perfectly and I can >> record for a long time. (CPU load is 9.6%) >> >> Reducing the frames per period to 512 and the same version of Audacity >> starts OK but after a few seconds, sometimes longer losses sync with the >> data stream. >> >> The standard 1.3.2 version of Audacity with resampling enabled does not >> sync on start with 512 frames per period but runs OK with Jack set to >> 1024 frames per period. >> >> The 1.3.3 version of Audacity complied with libsamplerate without >> portmixer runs OK with 512 frames per period under jack. >> >> NONE of the Audacity versions will run directly with the ALSA drivers. >> The Version with resampling disabled often starts OK, but after 30 sec >> to a few minutes of recording sync is lost and naturally distortion is >> the result. >> >> The conclusion is that portaudio is very poor at syncing with the >> alsadata stream, something which Jack can do with very little buffering. >> >> >> Ciao, Simon

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[alsa-devel] Correct use of ak4114.c?
by dustin＠seznam.cz 10 Apr '07

10 Apr '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O card, I used corresponding code from revo.c (and similarly from juli.c), calling function snd_ak4114_create. However, this function does not initialize array ak4114->kctls[idx], leading to consequent kernel oops after the periodically called snd_ak4114_check_rate_and_errors, specifically in function snd_ctl_notify called by snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4114->kctls[0]->id) Temporarily I resolved the problem by commenting out the snd_ctl_notify calls. However, it seems author of ak4114.c intended to create the device with snd_ak4114_build instead which defines all the controls. I did not hit the problem until fixing the 4-wire communication in revo.c - that explaing why nobody had this problem with revo cards. I do not understand how the code can work in juli.c - perhaps I have overlooked something there. My question: Is the correct way to add null pointer checks before snd_ctl_notify calls only, or to use snd_ak4114_build insted of snd_ak4114_create? If using snd_ak4114_build, where do I get the parameters ply_substream and cap_substream? int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *playback_substream, struct snd_pcm_substream *capture_substream); Thanks for any suggestions. Pavel.

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[alsa-devel] seq: timing unstable on slow computer when busy
by Gerald Grabner 09 Apr '07

09 Apr '07

Hi there, I'm trying to use a simple (self written) sequencer application on an old laptop with a 350 MHz CPU. It works, but the timing of output events becomes unstable when my program is busy with things like scheduling events for the subsequent loop or an unmuted track. I'm wondering about the best approach to tackle this problem. Depending on how the priority of the kernel's event scheduling is related to the priority of an application, I could imagine some of the following measures: * Running my process with lower or higher priority * Using (priorized) threads for the event scheduling * Setting Linux kernel config options * Setting ALSA related options (driver or lib) * ... While playing midi, I'm scheduling an echo event a few ticks before the end of every sequence, upon which a function to schedule the events of the next sequence is called. What is the typical length of this scheduling span: Just one bar? Or longer? Many thanks in advance, Gerald BTW: My operation system is Gentoo's Linux 2.6.19 with alsa-driver 1.0.14_rc1.

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[alsa-devel] Popping noise on HP dv6000z (CX20549) running x86 kernel
by enigma_0Z 08 Apr '07

08 Apr '07

Here's my situation: If I play a game (any game, in particular, however, neverball/putt or sauerbraten), the sound has these little soft "pop" noises happening so quick per second I cannot count them. This is when I'm running on an x86 kernel, however, when I run an x86_64 kernel, these are not present. I've got a 64-bit processor, specifically an AMD Sempron 3500+ Does Anyone know why? I've got a conexant CX20549 in my laptop. Thanks, enigma

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[alsa-devel] dont knw what this error means
by Ashlesha Shintre 06 Apr '07

06 Apr '07

Hi, I am getting the following error when I try to playback a raw sample file: "ALSA lib pcm.c:7216snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Does anyone know what this means? Thanks Ashlesha.

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