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[alsa-devel] test 3 - please ignore
by Jaroslav Kysela 23 Nov '07

23 Nov '07

test 3 - please ignore ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

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[alsa-devel] Mbox II DigiDisign
by rmgls＠free.fr 22 Nov '07

22 Nov '07

hello all, Can you tell me please if the external usb audio interface mbox 2 from digidisign is supported by alsa? Here is my config, sony laptop, with an integrated alc260 chip (snd-hda). and the mbox on debian. now the different debian messages: dmesg ... usb 2-1: new full speed USB device using uhci_hcd and address 3 usb 2-1: config 1 interface 2 has no altsetting 1 usb 2-1: config 1 interface 3 has no altsetting 1 usb 2-1: config 1 interface 4 has no altsetting 1 usb 2-1: config 1 interface 5 has no altsetting 1 usb 2-1: configuration #1 chosen from 1 choice usbcore: registered new interface driver snd-usb-audio usb 2-1: USB disconnect, address 3 usb 2-1: new full speed USB device using uhci_hcd and address 4 usb 2-1: config 1 interface 2 has no altsetting 1 usb 2-1: config 1 interface 3 has no altsetting 1 usb 2-1: config 1 interface 4 has no altsetting 1 usb 2-1: config 1 interface 5 has no altsetting 1 usb 2-1: configuration #1 chosen from 1 choice cat /dev/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.15 emulation code) Kernel: Linux mydomain.music 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: HDA Intel at 0x80000000 irq 19 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed Audio devices: 0: ALC260 Analog (DUPLEX) 1: USB Audio (DUPLEX) Synth devices: NOT ENABLED IN CONFIG Midi devices: NOT ENABLED IN CONFIG Timers: 7: system timer Mixers: 0: Realtek ALC260 1: USB Mixer aplay plays nothing and reports: (alsa-lib pcm error). alsamixer -c 1 reports: no mixer elem found. a google search does not provide no interesting technical sheet, at least i did not found any. have you something please: datasheets, config? thanks in advance. Best regards rmgls(a)free.fr

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Re: [alsa-devel] Broken playback
by Clemens Ladisch 22 Nov '07

22 Nov '07

Gabriele Melia wrote: > > snd_pcm_hw_params_set_buffer_size_near(), but the allowed values are > > hardware dependent. You might want to add your own circular buffer that > > is bigger than ALSA's. > > So your suggestion is to limit my buffer_size after decode to be same of > ALSA's? No; my suggestion is to use your own buffer which is big enough for your decoder, and use this buffer to fill ALSA's buffer, whenever possible. If one call to your decoder function needs so much time that ALSA's buffer can become emptied, you need another thread to re-fill the ALSA buffer in parallel with the decoding. Regards, Clemens

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[alsa-devel] alsa-devel on vger
by Takashi Iwai 21 Nov '07

21 Nov '07

Hi, What about the plan to move alsa-devel to vger? Shouldn't we do that soon? Takashi

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[alsa-devel] [PATCH 2/2] caiaq/NativeInstruments: control support
by Daniel Mack 21 Nov '07

21 Nov '07

Hi, This patch adds support for input controllers on "Kore controller" and uses ALSA's control API to enable LED switching on several hardware products as well as some internal audio related functions of "Audio 8 DJ". Subject: SOUND: caiaq - add control API and more input features From: Daniel Mack <daniel(a)caiaq.de> SOUND: caiaq - add control API and more input features - added support for all input controllers on Native Instrument's "Kore controller". - added ALSA controls to switch LEDs on "RigKontrol 2", "RigKontrol3", "Audio Kontrol 1" and "Kore controller". - added ALSA controls to switch input mode, software lock and ground lift features on "Audio 8 DJ". Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] [PATCH 1/2] caiaq/NativeInstruments: input handling fixes
by Daniel Mack 21 Nov '07

21 Nov '07

Hi, I'm re-posting a patch from Dmitry Torokhov which cleans up some input layer related things. It hasn't made its way to any repository yet but my subsequent patch needs it as dependency. I Cc'ed Dmitry to avoid confusion. Subject: SOUND: caiaq - misc input handling fixes From: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> SOUND: caiaq - misc input handling fixes - link input device with its parent so that it placed in proper spot in sysfs hierarchy - drivers that allow changing their keymaps should use private copy of the keymap so that one instance of a device does not affect another instance - it is preferred for drivers to properly set up input_dev->phys to help userspace locate devices - drivers should use usb_to_input_id(), or perform endianess conversion, themselves, otherwise ID is not correct on big-endian boxes - whitespace and formatting cleanup Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> Acked-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] ASoC merge
by Mark Brown 21 Nov '07

21 Nov '07

The following patch series merges some of the outstanding changes in the linux-2.6-asoc tree. It updates the core and existing drivers and adds some of the new drivers currently in the tree.

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Re: [alsa-devel] Broken playback
by Clemens Ladisch 21 Nov '07

21 Nov '07

Gabriele Melia wrote: > > Try non-blocking mode (see the snd_pcm_nonblock function). In non- > > blocking mode, the writei call will only write as much data as can be > > written without waiting (or returns -EAGAIN if the buffer is already > > full at the moment). > > I'm already in this case Then the problem is not that the writei call takes too long. An underrun can happen only when your code takes too much time between two consecutive calls to writei. > How to set the buffer size of the circular buffer in which i store the > pcm that is then called by alsa? snd_pcm_hw_params_set_buffer_size_near(), but the allowed values are hardware dependent. You might want to add your own circular buffer that is bigger than ALSA's. HTH Clemens

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Re: [alsa-devel] Broken playback
by Clemens Ladisch 21 Nov '07

21 Nov '07

Gabriele Melia wrote: > i have already tried to read and decode some data in order to accumulate > 1 second of data to prevent broken audio, > but nothing change, analysing the issue i have found that broken audio > comes when the snd_pcm_writei is called with a size that is 5000. So it > takes time to dispach all this size. Try non-blocking mode (see the snd_pcm_nonblock function). In non- blocking mode, the writei call will only write as much data as can be written without waiting (or returns -EAGAIN if the buffer is already full at the moment). HTH Clemens

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[alsa-devel] Latency and timestamps
by Claudio Matsuoka 21 Nov '07

21 Nov '07

Hi, I'm adding latency control to an application and didn't find much documentation about the pcm status functions aside from a very brief description and the latency.c example. What exactly are the "trigger timestamp" and "now timestamp" returned by snd_pcm_status_get_trigger_tstamp() and snd_pcm_status_get_tstamp()? I collected both from my output pcm handle to find values (in tv_sec.tv_usec format) such as 1195518180.938427 938427.557702 1195518180.938441 938441.557702 1195518181.300395 300395.557702 1195518181.300441 300441.557702 where the "trigger" tv_sec matches the "now" tv_usec, and the "trigger" tv_usec varies in large steps. Also the example latency.c says: 00703 showinmax(in_max); 00704 if (p_tstamp.tv_sec == p_tstamp.tv_sec && 00705 p_tstamp.tv_usec == c_tstamp.tv_usec) 00706 printf("Hardware sync\n"); 00707 snd_pcm_drop(chandle); In line 704, shouldn't it be p_tstamp.tv_sec == c_tstamp.tv_sec? My final goal is to sync on-screen events with the sound output. I'd like to do something like storing output metadata associated with a certain timestamp and collecting it back based on the timestamp of the currently playing pcm data (like the OSS sequencer echoback events), is that possible, or is there a better way to do this?

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