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[alsa-devel] support for jackSense feature in modern sound cards
by Balaji Rao 10 Feb '08

10 Feb '08

Hello people, How well is the jack sense feature in modern soundcards supported in the kernel ? I opened up a few driver sources like hda-intel and found the word 'jack sense' in some files, but i could not find it being used anywhere. Are there standard interfaces in kmix or equivalent software that use the jack sense feature ? If not, wouldn't it be nice to have a standard interface to expose this feature so that kmix and others could make use of in a device independent way ? If yes, I would be more than willing to contribute to its development. Please clarify. -- regards, balaji rao

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[alsa-devel] alsa-info.sh debugging script
by Travis Place 09 Feb '08

09 Feb '08

Hi everyone, I am new to this list, so be nice please :P I am the author of the alsa-info.sh debugging script, and heard you may want to use it within ALSA.. I have been speaking to Tobin, and he mentioned that a few things may need to be added (like alsactl output). I just want to get some ideas of things i can add, and if they should be included automatically, or require a --with-alsactl (for example). I have added the alsactl output to the script now, its included by default.. Any other ideas/changes that need to be made ? Travis Place (wishie) -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] snd-dummy and snd_pcm_new()
by William Juul 08 Feb '08

08 Feb '08

I am using snd-dummy on a NGW100 card with buildroot provided by Atmel. The alsa version is 1.0.14a and the kernel is 2.6.23. I have compiled three versions of the snd-dummy driver: the original, one with capture_count=0 and one with playback_count=0. When I use snd_pcm_new() with playback_count=0, there will be no PCM devices other than the "null" (see below). Is this expected? Can I still capture from this driver (and a real driver with the same symptoms)? Best regards William Juul Listings (annotated): (with original snd-dummy and modified snd-dummy to use capture_count=0) # modprobe snd-dummy # # arecord -l **** List of CAPTURE Hardware Devices **** card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 # # arecord -L default:CARD=Dummy Dummy, Dummy PCM Default Audio Device null Discard all samples (playback) or generate zero samples (capture) # reboot .. (with modified snd-dummy to use playback_count=0) # modprobe snd-dummy # arecord -l **** List of CAPTURE Hardware Devices **** card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 # arecord -L null Discard all samples (playback) or generate zero samples (capture) #

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[alsa-devel] PATCH: remove duplicate controls in alc268 test mixer
by Jonathan Woithe 08 Feb '08

08 Feb '08

Hi guys I've just noticed that there are a handful of duplicate controls in the ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes them. Signed-off-by: Jonathan Woithe <jwoithe(a)physics.adelaide.edu.au> --- patch_realtek.c-orig 2008-02-05 19:53:24.000000000 +1030 +++ patch_realtek.c 2008-02-08 09:23:02.000000000 +1030 @@ -9834,11 +9834,6 @@ #ifdef CONFIG_SND_DEBUG static struct snd_kcontrol_new alc268_test_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - /* Volume widgets */ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),

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[alsa-devel] [PATCH] Fix race condition in generic bound volume/swtich controls
by Serge A. Suchkov 08 Feb '08

08 Feb '08

Hi all, Attached patch fix race condition in hd_codec generic bound volume/swtich controls oops on this bug can be easy reproduced by two mixer apps on SMP system with PREEMPT kernel dmesg: ALSA /home/ss/ALSA/alsa-driver-1.0.16/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:596: hda_intel: azx_get_response timeout, switching to polling mode: las t cmd=0x014f0900 BUG: unable to handle kernel paging request at virtual address 00070006 printing eip: f8f43e95 *pde = 00000000 Oops: 0000 [#1] PREEMPT SMP Modules linked in: i915 drm snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_pcm_oss snd_mixer_oss bnep rfcomm hidp l2cap bluetooth w lan_wep acpi_cpufreq coretemp hwmon mmc_block pcspkr psmouse wlan_scan_sta ath_rate_sample snd_hda_intel ath_pci serio_raw wlan tg3 sdhci snd_pcm firewire_o hci mmc_core i2c_i801 snd_timer firewire_core snd_page_alloc ath_hal(P) snd_hwdep snd iTCO_wdt crc_itu_t iTCO_vendor_support shpchp video output acer_acpi b acklight led_class wmi_acer Pid: 3969, comm: gkrellm Tainted: P (2.6.24-jm #4) EIP: 0060:[<f8f43e95>] EFLAGS: 00010292 CPU: 0 EIP is at snd_hda_mixer_bind_ctls_info+0x20/0x43 [snd_hda_intel] EAX: 00000000 EBX: f7478e00 ECX: f763e000 EDX: f764f788 ESI: 00070002 EDI: edce5e00 EBP: edc3fe64 ESP: edc3fe54 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process gkrellm (pid: 3969, ti=edc3e000 task=f1e4e000 task.ti=edc3e000) Stack: f764f77c f7478e00 edce5e00 f6dd6000 edc3fe84 f8e590e8 edc7a239 f6d14034 f764f34c f6c0f7e0 edc3ff30 f6d14034 edc3fea8 f8e591b7 edc3ff30 edc3ff2c 00000000 f70aa668 f6d14034 f8e59165 bfbfadb0 edc3ff40 f8e587aa edc3ff2c Call Trace: [<c0104fbb>] show_trace_log_lvl+0x1a/0x2f [<c010506d>] show_stack_log_lvl+0x9d/0xa5 [<c0105119>] show_registers+0xa4/0x1bd [<c0105354>] die+0x122/0x206 [<c03daccc>] do_page_fault+0x535/0x623 [<c03d940a>] error_code+0x72/0x78 [<f8e590e8>] snd_mixer_oss_get_volume1_vol+0x74/0xf1 [snd_mixer_oss] [<f8e591b7>] snd_mixer_oss_get_volume1+0x52/0xa5 [snd_mixer_oss] [<f8e587aa>] snd_mixer_oss_ioctl1+0x673/0x71e [snd_mixer_oss] [<f8e588af>] snd_mixer_oss_ioctl+0xb/0xd [snd_mixer_oss] [<c017af0a>] do_ioctl+0x22/0x67 [<c017b186>] vfs_ioctl+0x237/0x24a [<c017b1ca>] sys_ioctl+0x31/0x4b [<c010402e>] syscall_call+0x7/0xb ======================= Code: 3f 49 c7 89 f8 59 5b 5e 5f 5d c3 55 89 e5 57 89 d7 56 53 89 c3 83 ec 04 8b 70 5c 8b 40 60 05 7c 01 00 00 89 45 f0 e8 c0 3f 49 c7 <8b> 46 04 89 fa 89 4 3 5c 89 d8 8b 0e ff 11 89 73 5c 89 c7 8b 45 EIP: [<f8f43e95>] snd_hda_mixer_bind_ctls_info+0x20/0x43 [snd_hda_intel] SS:ESP 0068:edc3fe54 ---[ end trace 0a20bc209e9397cc ]--- similar issue report present in ALSA bugtracking system https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3652 -- Regards, Serge A. Suchkov -- Democracy is a form of government that substitutes election by the incompetent many for appointment by the corrupt few. -- G. B. Shaw

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Re: [alsa-devel] 1.0.16rc2 -build issues
by Alan Horstmann 08 Feb '08

08 Feb '08

Hi Takashi, I tried out yesterdays tarball hg20080207 to confirm DSP2000 Midi stuff before pointing another DSP2000 user to it. Build proceeded OK but I did notice the following warnings, which I might have missed previously, or they could be new. Do you care about redefinitions? pcm_oss.c:639:1: warning: "AFMT_S32_LE" redefined In file included from pcm_oss.c:43: /lib/modules/2.4.21-99-athlon/build/include/linux/soundcard.h:562:1: warning: this is the location of the previous definition pcm_oss.c:640:1: warning: "AFMT_S32_BE" redefined /lib/modules/2.4.21-99-athlon/build/include/linux/soundcard.h:563:1: warning: this is the location of the previous definition pcm_oss.c:641:1: warning: "AFMT_S24_LE" redefined /lib/modules/2.4.21-99-athlon/build/include/linux/soundcard.h:558:1: warning: this is the location of the previous definition pcm_oss.c:642:1: warning: "AFMT_S24_BE" redefined /lib/modules/2.4.21-99-athlon/build/include/linux/soundcard.h:559:1: warning: this is the location of the previous definition Regards Alan

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[alsa-devel] new driver - some initial problems
by William Juul 08 Feb '08

08 Feb '08

I am new to alsa development, but I am trying to write a alsa capture driver for a DAC. The driver is based on at73c213 since it will be running on AVR32 and using the SSC bus. The driver is running on a NGW100 card and I am using buildroot provided by Atmel. The alsa version is 1.0.14a and the kernel is 2.6.23. The DAC is not using any configuration bus so I have switched to using platform_* functions instead of dev/spi (grabbed from snd-dummy). I have also been taking some capture pieces from the snd-dummy driver. I am now stuck with two problems: 1) When using arecord -L, I only get the "null" device 2) When I try to use arecord to do a recording I get "arecord: set_params:1039: unable to install sw params:". I get the same message when trying to use the snd-dummy driver, even though this driver also has PCM device. Any hints on how to proceed? I can post the driver source if that is interesting. Best regards William Juul Listings (annotated): # modprobe -v snd-dummy Loading module snd-page-alloc Loading module soundcore Loading module snd Loading module snd-timer Loading module snd-pcm Loading module snd-dummy # # # arecord -l **** List of CAPTURE Hardware Devices **** card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 # # arecord -L default:CARD=Dummy Dummy, Dummy PCM Default Audio Device null Discard all samples (playback) or generate zero samples (capture) # # # # arecord > dummy.wav Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono pcm->setup: 1 stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 355 period_size : 355 period_time : 44375 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 355 xfer_align : 355 start_threshold : 1 stop_threshold : 355 silence_threshold: 0 silence_size : 0 boundary : 1488977920 arecord: set_params:1039: unable to install sw params: start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 0 xfer_align: 0 silence_threshold: 0 silence_size: 0 boundary: 1488977920 # # # arecord -r 22786 -c 1 -f U8 -s 1 -A 100 -v > dummy.wav # (This works OK) # # reboot ... # # # modprobe -v snd-ads127x Loading module soundcore Loading module snd Loading module snd-page-alloc Loading module snd-timer Loading module snd-pcm Loading module snd-ads127x snd_ads127x_probe OK # # # arecord -l **** List of CAPTURE Hardware Devices **** card 0: ads127x_0 [AVR32 NGW100 external DAC], device 0: ads127x [ads127x] Subdevices: 1/1 Subdevice #0: subdevice #0 # arecord -L null Discard all samples (playback) or generate zero samples (capture) # # # # # arecord -r 22786 -c 1 -f S24_LE -s 1 -A 100 > ads127x.wav ALSA /home/williaj/src/ads127x/src/ads127x.c:141: snd_ads127x_pcm_open Recording WAVE 'stdin' : Signed 24 bit Little Endian, Rate 22786 Hz, Mono pcm->setup: 1 stream : CAPTURE accessALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 : RW_INTERLEAVED format : S24_LEALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 subformat : STD channels : 1 raALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 te : 22786 exact rate : 22786 (22786/1) msbits : 32 buffer_size : 4044 period_size : 1011 period_time : 44369 tick_time ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 : 4000 tstamp_mode : NONE period_step ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 : 1 sleep_min : 0 ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 xfer_align : 1011 start_threshold : 1 ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 stop_threshold : 4044 silence_threshold: ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 0 silence_size : 0 boundary : 21202206ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 72 overrun!!! (at least 1.074 ms long) overrun!!! (at least 0.151 ms long) overrun!!! (at least 0.139 ms long) overrun!!! (at least 0.141 ms long) overrALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 un!!! (at least 0.142 ms long) overrun!!! (at lALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 east 0.139 ms long) overrun!!! (at least 0.137 ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 ms long) overrun!!! (at least 0.139 ms long) oALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 verrun!!! (at least 0.138 ms long) overrun!!! (ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 at least 0.149 ms long) overrun!!! (at least 0.ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 145 ms long) overrun!!! (at least 0.154 ms long) overrun!!! (at least 0.146 ms long) overrun!!! (at least 0.146 ms long) overrun!!! (at least 0.724 ms loALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 ng) overrun!!! (at least 0.152 ms long) ALSA sound/core/pcm_lib.c:155: BUG: stream = 1, pos = 0x3b5f034, buffer size = 0xfcc, period size = 0x3f3 overrun!!! (at least 0.125 ms long)

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[alsa-devel] New Guy Here...
by Minko Minkov 08 Feb '08

08 Feb '08

Greetings, I recently bought an Edirol UA1-EX. While looking at the Matrix I saw that you need help testing the device. I can devote some time in testing in development. Although I am not new to linux I'm new to the alsa development process. I would very much appreciate if you point me to some starter's documentation or anything that will help. In the meantime if there is anything I can help with I would gladly do so. Thanks, Minko

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[alsa-devel] em8300 and ALSA problems
by Jan Willies 08 Feb '08

08 Feb '08

Hi, I really want pulseaudio badly and since it doesn't work very good with OSS output, I decided to switch to ALSA. However, I'm running into some problems with ALSA. My dxr3 (em8300) card seems to be detected by ALSA (/proc/asound/cards lists it too): [root@server ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: I82801BAICH2 [Intel 82801BA-ICH2], device 0: Intel ICH [Intel 82801BA-ICH2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: EM8300 [Sigma Designs' EM8300], device 0: EM8300/0 [EM8300 DAC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: EM8300 [Sigma Designs' EM8300], device 1: EM8300/1 [EM8300 IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 But for some reason it won't play anything: [root@server ~]# alsaplayer --verbose -o alsa -d hw:1,0 Strong.mp3 AlsaPlayer 0.99.80 (C) 1999-2004 Andy Lo A Foe <andy(a)alsaplayer.org> and others. Output plugin: ALSA output v1.9.0beta12 error on set_format SND_PCM_FORMAT_S16 Unavailable hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_BE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: [32000 48000] PERIOD_TIME: (21333 32000] PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: [2 256] BUFFER_TIME: (42666 8192000] BUFFER_SIZE: [2048 262144] BUFFER_BYTES: [8192 1048576] TICK_TIME: ALL error on set_format SND_PCM_FORMAT_S16 Unavailable hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_BE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: [32000 48000] PERIOD_TIME: (21333 32000] PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: [2 256] BUFFER_TIME: (42666 8192000] BUFFER_SIZE: [2048 262144] BUFFER_BYTES: [8192 1048576] TICK_TIME: ALL failed to configure output device...trying OSS error opening /dev/dsp Failed to initialize plugin! Failed to register plugin: /usr/lib/alsaplayer/output/liboss_out.so failed to load output plugin (alsa). exitting... I don't really have a clue what's going on. People on #alsa didn't either. Here's a link to tsalsa output: http://nopaste.com/p/a0NYa75gW The module was loaded with options em8300 dicom_fix=1 dicom_control=1 dicom_other_pal=1 audio_driver=alsa. I'm running em8300-0.16.4-rc1 - Jan

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[alsa-devel] [PATCH] SND_SEQ_BLOCK and SND_SEQ_PORT_CAP_NONE defines
by Aldrin Martoq 08 Feb '08

08 Feb '08

Hey hackers, While developing pyalsa sequencer API, I found that sometimes 0 is used instead of a #defined value or constant. In my opinion, given that the alsa API is opaque (hides internal structure and provides methods for accessing them), we should define those missing "constants" for being consistent. So far I found: SND_SEQ_BLOCK 0 (blocking mode) SND_SEQ_PORT_CAP_NONE (the port defines no access capabilites) BTW, Thanks Jaroslav for putting my code; I will send 3rd patch for covering the missing items of the python API sequencer when I'll finish it. Signed-off-by: Aldrin Martoq <amartoq(a)dcc.uchile.cl> --- diff --git a/include/seq.h b/include/seq.h --- a/include/seq.h +++ b/include/seq.h @@ -56,7 +56,8 @@ typedef struct _snd_seq snd_seq_t; /** * sequencer opening mode */ -#define SND_SEQ_NONBLOCK 0x0001 /**< non-blocking mode (flag to open mode) */ +#define SND_SEQ_BLOCK 0x0000 /**< blocking opening mode (flag for #snd_seq_open, #snd_seq_nonblock) */ +#define SND_SEQ_NONBLOCK 0x0001 /**< non-blocking opening mode (flag for #snd_seq_open, #snd_seq_nonblock) */ /** sequencer handle type */ typedef enum _snd_seq_type { @@ -204,6 +205,7 @@ typedef struct _snd_seq_port_info snd_se #define SND_SEQ_PORT_SYSTEM_ANNOUNCE 1 /**< system announce port */ /** port capabilities (32 bits) */ +#define SND_SEQ_PORT_CAP_NONE (0<<0) /**< port has no access capabilities */ #define SND_SEQ_PORT_CAP_READ (1<<0) /**< readable from this port */ #define SND_SEQ_PORT_CAP_WRITE (1<<1) /**< writable to this port */ diff --git a/src/seq/seq.c b/src/seq/seq.c --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -958,7 +958,7 @@ static int snd_seq_open_noupdate(snd_seq * \note Internally, these are translated to \c O_WRONLY, \c O_RDONLY and * \c O_RDWR respectively and used as the second argument to the C library * open() call. - * \param mode Optional modifier. Can be either 0, or + * \param mode Optional modifier. Can be either #SND_SEQ_BLOCK, or * #SND_SEQ_NONBLOCK, which will make read/write operations * non-blocking. This can also be set later using #snd_seq_nonblock(). * \return 0 on success otherwise a negative error code @@ -2161,8 +2161,9 @@ void snd_seq_port_info_set_timestamp_que * Each port has the capability bit-masks to specify the access capability * of the port from other clients. * The capability bit flags are defined as follows: + * - #SND_SEQ_PORT_CAP_NONE No access capabilities * - #SND_SEQ_PORT_CAP_READ Readable from this port - * - #SND_SEQ_PORT_CAP_WRITE Writable to this port. + * - #SND_SEQ_PORT_CAP_WRITE Writable to this port * - #SND_SEQ_PORT_CAP_SYNC_READ For synchronization (not implemented) * - #SND_SEQ_PORT_CAP_SYNC_WRITE For synchronization (not implemented) * - #SND_SEQ_PORT_CAP_DUPLEX Read/write duplex access is supported -- Aldrin Martoq

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