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

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests 0/5
by mark gross 07 Dec '09

07 Dec '09

This is a 5 patch series against linux-next-20091204 that changes pm_qos to use a handle based access model for the kernel api. The string based model was shown to be slow if pm_qos parameters are updated on hot code paths. Most of the design changes where motivated by a patch aili(a)codeaurora.org sent me, and I spent some time cleaning up the API. Also, I couldn't resist myself in renaming the API's a bit given the fact that the signatures changed enough that I had to touch all the pm_qos users anyway. I changed *requirement* to *request* in keeping with the way PM_QOS really only does best effort. I've felt "requirement" is too strong a word for the way it works. If folks would rather me do the function re-naming in a separate patch set we can do that too. diffstat for the series against linux-next-20091204 : drivers/acpi/processor_idle.c | 2 drivers/cpuidle/governors/ladder.c | 2 drivers/cpuidle/governors/menu.c | 2 drivers/net/e1000e/netdev.c | 22 ++- drivers/net/igbvf/netdev.c | 6 - drivers/net/wireless/ipw2x00/ipw2100.c | 11 + include/linux/netdevice.h | 4 include/linux/pm_qos_params.h | 14 +- include/sound/pcm.h | 3 kernel/pm_qos_params.c | 188 ++++++++++++++------------------- net/mac80211/mlme.c | 2 sound/core/pcm.c | 3 sound/core/pcm_native.c | 14 +- 13 files changed, 132 insertions(+), 141 deletions(-) --mgross

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[alsa-devel] About TASCAM US-122 mkII support
by Tobias Hansen 07 Dec '09

07 Dec '09

Alberto Martinez de la Ossa <delaossa(a)mail.desy.de> wrote: > Hello, > I have just got the TASCAM US-122 mkI > <http://www.tascam.com/products/us-122mkII.html>I audio/midi interface, > which seemed to me a great product and a good deal. > The problem is that I use a linux system (ubuntu-studio karmic) for > music production and the interface can not be recognized when plugged. > Among other reasons I chose this device because I thought it would work > in linux: > There were alsa drivers for its predecessors, the us-122 > <http://alsa.opensrc.org/Tascam_US-122> and the us-122l > <http://www.pps.jussieu.fr/%7Esmimram/tascam/> models. > The latter, thanks to Karsten Wiese who wrote a patch for the us-122l > version, which was latter included by default in newer alsa-firmware > releases. > > Unfortunately, the new model (US-122 mkII), the one I have, is not > supported yet. > I have even tried to get it replaced by the previous us-122l model, but > it is not available anymore in the music store -> http://www.justmusic.de > On the other hand, the "linux specialist" of the music store told me > that a new linux driver for the us-122 mkII is been prepared, > although he doesn't know when it will be ready. > Being you the experts I have found in the web in regards with ALSA and > TASCAM usb interfaces, I have decided to directly ask you about this. > I would really appreciate any information about the topic. Hi, I know that the US-144 MKII has a different Product ID (8020) than the US-144 (800F), so the US-122 MKII probably also has another one than the US-122L. If you post your output of lsusb with the device plugged in, I could make a patch that adds these both IDs to the us122l driver for testing if that suffices. I don't know if you are familiar with compiling kernel modules etc. If you need more detailed instructions, you can join the discussion at http://forum.ubuntuusers.de/topic/tascam-us-144 in either German or English (I know desy.de doesn't necessarily mean you speak German) or send a mail. By the way, I suspect the "linux specialist" at that shop is good in making things up, because Tascam has never provided any Linux drivers and he can't really know what the Linux people do on their own. Best regards, Tobias Hansen

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Re: [alsa-devel] [PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Mark Brown 07 Dec '09

07 Dec '09

On Sat, Nov 28, 2009 at 07:46:22PM -0700, Tabi Timur-B04825 wrote: > Have you tested it on a Freescale MPC8610 HPCD? Are you sure it still > compiles and builds on any PowerPC system? I can do a compile test on PowerPC (and that's one of the platforms that gets tested with linux-next too) but I don't have hardware to do runtime tests with. Timur, would you be OK with compile tests for now - I'd not expect to see 2.6.33 go out this year so if there are any runtime issues it should be possible to fix them once you're back in the office?

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[alsa-devel] plans to support mixer widget on hda?
by Alexey Fisher 07 Dec '09

07 Dec '09

hallo, are there any plans to support mixer widget? I found it's really nice. Currently i use it with hda-verb for karaoke, to record skype call or just to make some live mix for online radio. Alexey

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[alsa-devel] About TASCAM US-122 mkII support
by Alberto Martinez de la Ossa 06 Dec '09

06 Dec '09

Hello, I have just got the TASCAM US-122 mkI <http://www.tascam.com/products/us-122mkII.html>I audio/midi interface, which seemed to me a great product and a good deal. The problem is that I use a linux system (ubuntu-studio karmic) for music production and the interface can not be recognized when plugged. Among other reasons I chose this device because I thought it would work in linux: There were alsa drivers for its predecessors, the us-122 <http://alsa.opensrc.org/Tascam_US-122> and the us-122l <http://www.pps.jussieu.fr/%7Esmimram/tascam/> models. The latter, thanks to Karsten Wiese who wrote a patch for the us-122l version, which was latter included by default in newer alsa-firmware releases. Unfortunately, the new model (US-122 mkII), the one I have, is not supported yet. I have even tried to get it replaced by the previous us-122l model, but it is not available anymore in the music store -> http://www.justmusic.de On the other hand, the "linux specialist" of the music store told me that a new linux driver for the us-122 mkII is been prepared, although he doesn't know when it will be ready. Being you the experts I have found in the web in regards with ALSA and TASCAM usb interfaces, I have decided to directly ask you about this. I would really appreciate any information about the topic. Thanks in advance for your attention. Best regards,

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Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, in my previous mail I put some snd_pcm_ioplug_set_param_minmax code in the wrong callback function. The corrected code should be below: #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int format_list[] = { SND_PCM_FORMAT_S16_LE, }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

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Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, Raul Xiong wrote: > Hi Stefan > > This is because your pointer callback function is not implemented properly. > "pointer" callback function should return the right position of the > playback/capture buffer, apps over alsa framework will call > snd_pcm_avail_update to get the position. > Also, you should call snd_pcm_ioplug_set_param_minmax in your > play_hw_constraint function to define the buffer size relevant parameters. Well, I did what you suggested but the start() callback is still not called: $ aplay -D play test_8khz_16LE_mono.wav Playing WAVE 'test_8khz_16LE_mono.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono snd_pcm_play_hw_params:84 24610 snd_pcm_play_prepare:71 24610 hw_ptr -> 0 snd_pcm_play_pointer:54 24610 ptr: 0 snd_pcm_play_transfer:63 24610 aborting with -EINVAL ... aplay: pcm_write:1442: write error: Invalid argument Also, when transferring PCM samples from the ALSA buffer to somewhere else, it seems logical to me that the initial value of the hw_ptr has to be 0, right ? As you can see above, the situation is still exactly the same and the start callback is never called at all. The code for the plugin is below. cheers, stefan #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; snd_pcm_hw_params_t *hw_params; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; unsigned int format_list[] = { SND_PCM_FORMAT_S16 }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int formats[] = { SND_PCM_FORMAT_S16_LE, }; int err; err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) { SNDERR("cannot set HW access constraint list: %s", snd_strerror(err)); return err; } err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(formats), formats); if (err < 0) { SNDERR("cannot set HW format constraint list: %s", snd_strerror(err)); return err; } return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

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[alsa-devel] ControlNames: Input Source/ Capture Source/ Jack Mode?
by Alexey Fisher 06 Dec '09

06 Dec '09

Hallo, i have question about controlnames, or if i correctly understand it: Capture Source - you can capture only from one of two (or more) sources. On switching mic a or mic b will be enabled or disabled. /--(o)--mic b cap switch--- \--(x)--mic a Input Source - same as capture source but more verbose /---(Mute/Vol)--Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)--Mic b more complicated setup: +---(Mute/Vol)--Mic a input switch a-(a(x)/b(o))--+ +---(Mute/Vol)--Mic b input switch b-(a(x)/b(o))--+ jack mode - you can choice to put VREF on the line or not. /---(Mute/Vol)-(mode)-Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)-(mode)-Mic b i hope i understand it correctly. Now is my situation: +---------+--(Mute/Rec)-(mode)-Mic a input switch a-(a(x)/b(o)/mix(o))--+ | +---------+--(Mute/Rec)-(mode)-Mic b input switch b-(a(x)/b(o)/mix(o))--+ | +--mixer--+<---(Mute/Vol for Mic a/b) are there any way to make alsamixer less verbose but still productive?

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[alsa-devel] ALSA on Alienware M17x
by Adam Gray 06 Dec '09

06 Dec '09

Hi guys, I'm having a lot of trouble getting sound out of my Alienware M17x. I've attached the output from alsa-info. I'm running kernel version 2.6.28 with ALSA driver version 1.0.18. There's no sound at all from the speakers or the headphones. In alsamixer, all channels appear un-muted and turned up full. If you need any further information please let me know. Regards, Adam Gray

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[alsa-devel] when i plugin my headphone the sound come out from speaker and headphone
by Michael Domanski 05 Dec '09

05 Dec '09

Hi, When I plugin my headphone the sound come out from speaker and headphone. The speaker and headphone working in the same time. I want my speaker to mute when headphone is plugged. http://www.alsa-project.org/db/?f=841de9447634e7e35ae0171f968c70de253259af mixer@mdomansky ~ $ sudo lspci |grep -i "multi\|audio\|040[0-3]" 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) mixer@mdomansky ~ $ sudo rmmod snd-hda-intel mixer@mdomansky ~ $ sudo modprobe snd-hda-intel model=hp WARNING: Deprecated config file /etc/modprobe.conf, all config files belong into /etc/modprobe.d/. WARNING: All config files need .conf: /etc/modprobe.d/alsa, it will be ignored in a future release. mixer@mdomansky ~ $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x92400000 irq 22 mixer@mdomansky ~ $ speaker-test -c2 -D hw:0,0 -t wav -l1 ^^^^^ sounds ok ^^^^^^ -- Sincerely, Michael +7 904 512 59 36 MasterWeb.PRO

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