[Alsa-devel] Re: [alsa-dev]Question regarding period/buffer and error handling

2004-05-28 Thread Måns Rullgård
Cournapeau David [EMAIL PROTECTED] writes:

 Hi there,

 For my research, I need to use audio with matlab under linux, and
 sound support of matlab is kind of... well, crappy (basically, it
 is opening the /dev/audio file and write to it; on my computer, it
 doesn't seem to work). That's why I am thinking about using a
 wrapper to alsa.

If you are playing fairly short pieces of sound you might be able to
use this function I wrote some time ago:

function playsnd(y,fs,bits)

wf = tempname;
ws = warning;
warning off
wavwrite(y, fs, bits, wf);
warning ws
wf = [wf '.wav'];
[s,o] = unix(sprintf('aplay %s', wf));
delete(wf);


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[Alsa-devel] Re: Question regarding period/buffer and error handling

2004-05-28 Thread Måns Rullgård
Cournapeau David [EMAIL PROTECTED] writes:

 Måns Rullgård wrote:


If you are playing fairly short pieces of sound you might be able to
use this function I wrote some time ago:

function playsnd(y,fs,bits)

wf = tempname;
ws = warning;
warning off
wavwrite(y, fs, bits, wf);
warning ws
wf = [wf '.wav'];
[s,o] = unix(sprintf('aplay %s', wf));
delete(wf);

 Well, I precisely want to avoid writing to a temporary file :)

If the files are small they will probably never hit the physical
disk.  Of course, your sounds might be longer than that.

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[Alsa-devel] alsa-lib spamming terminal

2004-05-28 Thread Måns Rullgård
Is there a way to make alsa-lib stop spamming the terminal with error
messages, for instance when a non-blocking open fails?

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[Alsa-devel] Re: alsa-lib spamming terminal

2004-05-28 Thread Måns Rullgård
Clemens Ladisch [EMAIL PROTECTED] writes:

 Måns Rullgård wrote:
 Is there a way to make alsa-lib stop spamming the terminal with error
 messages, for instance when a non-blocking open fails?

 snd_lib_error_set_handler()

Thanks.  I wonder why I didn't find that when reading the docs.

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[Alsa-devel] Re: [Alsa-dev]detecing if a device is already used by an other app

2004-05-26 Thread Måns Rullgård
Cournapeau David [EMAIL PROTECTED] writes:

 Hi there,

 I am currently using two computers which have a crappy intl8x0
 audio chipset, and these chipset cannot be used by two different
 apps in the same time. For example, if an audio app is using the
 soundcard, trying to launch xmms ( with alsa output plugin) gives
 an error message which says something like check that no other
 app is blocking the device. But if I launch alsaplayer (alsa
 output plugin) instead of xmms, the app doesn't complain: it
 'just' blocks on the snd_pcm_open call.
 Basically, after having looked at the source, is seems like this
 difference is coming from the flag in snd_pcm_open: if
 SND_PCM_NONBLOCK  is used, opening an already opened device
 failed, if SND_PCM_ASYNC is used, the app just blocks, until the
 other app releases the device.

This is correct.

 Is there a (simple) way to use the blocking call and detecting if
 the device is already opened ?

You can open it in non-blocking mode and set it to blocking later, if
you want that behavior when writing data.

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[Alsa-devel] Re: Audio and Video sync. Getting Timestamps

2004-05-07 Thread Måns Rullgård
James Courtier-Dutton [EMAIL PROTECTED] writes:

 Juan Carlos Granda wrote:
 That's my app does:
 1.- Open the device for capture
 2.- Set the access mode SND_PCM_ACCESS_RW_INTERLEAVED
 3.- Set format 16 bits (SND_PCM_FORMAT_S16_LE)
 4.- Set channels 2 (stereo)
 5.- Set buffer time near 1 second
 6.- Set period time near 0.1 seconds
 7.- Copy the hardware params to alsa
 8.- Set start threshold (0x7f - Explicit start)
 9.- Set transfer align 1
 10.- Copy the software params to alsa
 read loop:
 - Read 'period size' frames (snd_pcm_readi)
 How can i get the timestamp of the first frame read? If i use
 snd_pcm_status_get_trigger_tstamp i obtain the same timestamp all the times.
 If i use snd_pcm_status_get_tstamp i obtain the now time. Is there any way
 to get the timestamp of every period.
 Thanks


 I help develope xinehq.de.
 We have needed to do audio, video sync, but we don't need any audio
 timestamps using the get_trigger_tstamp etc.
 We just use the snd_pcm_delay() call.

 Take an example.
 Sample rate 1 Samples a second (1 just to make things easy to
 explain.)

 We call snd_pcm_delay() and it returns 4000.
 4000 samples at a sample rate of 1 = 400ms.
 If we now write some samples to the sound device, we know that they
 will arrive at the speakers in 400ms time.

The OP was recording.

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[Alsa-devel] assert failures in snd_pcm_mmap_commit

2004-05-04 Thread Måns Rullgård
When playing music using alsa-lib sometimes, very rarely, the player
crashes and dumps core.  GDB gives me this stack trace:

#0  0xe410 in ?? ()
#1  0x4be05684 in ?? ()
#2  0x0006 in ?? ()
#3  0x7126 in ?? ()
#4  0x4007e859 in *__GI_raise (sig=6)
at ../nptl/sysdeps/unix/sysv/linux/raise.c:66
#5  0x40080032 in *__GI_abort () at ../sysdeps/generic/abort.c:88
#6  0x4007805f in *__GI___assert_fail (assertion=0x0, file=0x0, line=0, 
function=0x48c9c136 snd_pcm_mmap_commit) at assert.c:83
#7  0x48c3ef71 in snd_pcm_mmap_commit (pcm=0x81232f8, offset=3840, frames=480)
at pcm_local.h:289
#8  0x48c5307e in snd_pcm_rate_commit_next_period (pcm=0x80b0e68, 
appl_offset=6248) at pcm_rate.c:1005
#9  0x48c53397 in snd_pcm_rate_mmap_commit (pcm=0x80b0e68, offset=6028, 
size=220) at pcm_rate.c:1194
#10 0x48c3ef18 in snd_pcm_mmap_commit (pcm=0x80b0e68, offset=6028, frames=440)
at pcm.c:5920
#11 0x48c5716a in snd_pcm_mmap_write_areas (pcm=0x80b0e68, areas=0x4be059b0, 
offset=0, size=440) at pcm_mmap.c:120
#12 0x48c3f373 in snd_pcm_write_areas (pcm=0x80b0e68, areas=0x4be059b0, 
offset=0, size=6380, func=0x48c57090 snd_pcm_mmap_write_areas)
at pcm.c:6094
#13 0x48c57361 in snd_pcm_mmap_writei (pcm=0x80b0e68, buffer=0x0, size=0)
at pcm_mmap.c:180
#14 0x48c381d2 in snd_pcm_writei (pcm=0x7126, buffer=0x80fb420, size=6500)
at pcm_local.h:370
#15 0x48bfbb81 in alsa_write (ad=0x0, data=0x0, samples=0)
at ../../tcvp/src/output/drivers/alsa/alsa.c:132
#16 0x4ae01689 in audio_play (p=0x80f56f0)
at ../../tcvp/src/output/audio/audio.c:218
#17 0x4002b853 in start_thread (arg=0x0) at pthread_create.c:264
#18 0x4012e3aa in clone () from /lib/libc.so.6

I am using Linux 2.6.4 with ALSA driver 1.0.3 and alsa-lib 1.0.3a
(which I see now is a little out-dated).  The sound card is a SiS7012
in an Asus laptop.

Could this be caused by a bug in the player?

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[Alsa-devel] Re: Creative MP3+ or Hercules Muse Pocket

2004-05-04 Thread Måns Rullgård
Frank Barknecht [EMAIL PROTECTED] writes:

 Hallo,
 Robin Cook hat gesagt: // Robin Cook wrote:

 Trying to find out if either or both of the Creative Soundblaster MP3+
 or the Hercules Muse Pocket usb sound cards are supported by the usb
 driver in ALSA?
 
 If so are is there anything special that needs to be done to get them
 setup under alsa?

 Normally these cards just all use the snd-usb-audio driver. If you run
 kernel 2.6 you can just modprobe snd-usb-audio to get sound. I never
 hear of the Hercules card, but it looks very cool with that knob,
 which might even work as an input device on Linux. This would then
 maybe require the hid/event interface of the USB system in the kernel,
 which is not part of ALSA.

Isn't the knob just a volume control?  It wouldn't need to send
anything to the computer.

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[Alsa-devel] Re: An alsa lib, that works over OSS drivers ?

2004-04-19 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Mon, 19 Apr 2004, David Balazic wrote:

 This new library would have the same name and API as the existing alsa
 library.
 
 How does this sound ? Feasible ?

 It's enough to write OSS plugin to alsa-lib.

Has it been done?

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[Alsa-devel] Re: An alsa lib, that works over OSS drivers ?

2004-04-19 Thread Måns Rullgård
Me Eby [EMAIL PROTECTED] writes:

 Can't be done that way.  That is why there is an OSS
 wrapper for alsa.  The OSS drivers do not support
 all of the capabilities that ALSA drivers do.

So now there's one person saying it can't be done, and another saying
it has been done.  There seems to be slight contradiction here.

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[Alsa-devel] Re: An alsa lib, that works over OSS drivers ?

2004-04-19 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Mon, 19 Apr 2004, M?ns Rullg?rd wrote:

 Me Eby [EMAIL PROTECTED] writes:
 
  Can't be done that way.  That is why there is an OSS
  wrapper for alsa.  The OSS drivers do not support
  all of the capabilities that ALSA drivers do.
 
 So now there's one person saying it can't be done, and another saying
 it has been done.  There seems to be slight contradiction here.

 Not really. This comment is for the OSS API-ALSA driver conversion, but
 the former question was for the ALSA API-OSS driver conversion.

They were both replying to the same message, so I assumed they were
referring to the same idea.  I must have missed something.

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[Alsa-devel] Re: Fw: [Bug 76413] arts does not follow ALSA API

2004-03-19 Thread Måns Rullgård
James Courtier-Dutton [EMAIL PROTECTED] writes:

 Joerg Mayer wrote:
 On Fri, Mar 19, 2004 at 08:30:13PM +, James Courtier-Dutton wrote:

 The guys there just don't understand what you told them. If they
 took the time to actually read what you said, they could easily fix
 their problem.

Summary: -
No bug in alsa.
 Summary 2:
 There seem to be quite a lot of projects that have problems to get native
 alsa support right: wine, winex, arts, xmms...
 wine is only just starting to support alsa properly, due mainly to
 lack of real need to alsa support for a long time. They are starting
 to support it well now.
 xmms don't support it well for obvious reasons. They charge people for
 OSS drivers, so the xmms alsa driver has to look worse!

The funny thing is that they can't even get that right, unless, of
course, you consider random skips a sign of correctness.

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Re: [Alsa-devel] snd_pcm_wait strangeness when resampling

2004-03-12 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Thu, 11 Mar 2004, M?ns Rullg?rd wrote:

 I've noticed some strange things with snd_pcm_wait.  My application
 opens the sound device in non-blocking mode and uses snd_pcm_wait
 whenever snd_pcm_writei return EAGAIN.  This works fine as long as
 ALSA isn't doing any resampling.  However, when resampling (e.g. from
 44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that
 there is room in the buffer, after which snd_pcm_writei obviously
 returns EAGAIN once more.  The same thing happens if I use OSS
 emulation and select() or poll().  Replacing the snd_pcm_wait call
 with usleep(1) makes the CPU load look normal again, while the
 application still works.
 
 I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a
 SIS7012 chip.  It behaves the same way with kernel 2.6.3 and driver
 1.0.2c.  I'm not quite sure when it started.
 
 This application used to work properly.  Has something changed so I
 need to modify my application?

 Could you give me your code to analyze the problem?

The full code is in http://tcvp.sf.net/tcvp-snapshot.tar.gz, if you
feel like digging through 20k lines of code.

The ALSA part is in here:
http://tcvp.bkbits.net:8080/tcvp/anno/src/output/drivers/alsa/alsa.c%401.9

It is called from here:
http://tcvp.bkbits.net:8080/tcvp/anno/src/output/audio/audio.c%401.13
in the audio_play function

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Re: [Alsa-devel] snd_pcm_wait strangeness when resampling

2004-03-12 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Thu, 11 Mar 2004, M?ns Rullg?rd wrote:

 I've noticed some strange things with snd_pcm_wait.  My application
 opens the sound device in non-blocking mode and uses snd_pcm_wait
 whenever snd_pcm_writei return EAGAIN.  This works fine as long as
 ALSA isn't doing any resampling.  However, when resampling (e.g. from
 44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that
 there is room in the buffer, after which snd_pcm_writei obviously
 returns EAGAIN once more.  The same thing happens if I use OSS
 emulation and select() or poll().  Replacing the snd_pcm_wait call
 with usleep(1) makes the CPU load look normal again, while the
 application still works.
 
 I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a
 SIS7012 chip.  It behaves the same way with kernel 2.6.3 and driver
 1.0.2c.  I'm not quite sure when it started.
 
 This application used to work properly.  Has something changed so I
 need to modify my application?

 Could you give me your code to analyze the problem?

I've noticed that it works properly using alsa-lib 1.0.0rc2.  That's
what I happened to have compiled, it might work with later versions
too.

Did you manage to build the thing yet?

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[Alsa-devel] Re: [2.6.3] snd_intel8x0: playing to fast

2004-03-11 Thread Måns Rullgård
Luca [EMAIL PROTECTED] writes:

 Il Wed, Mar 10, 2004 at 11:08:22PM +0100, M?ns Rullg?rd ha scritto: 
 *This message was transferred with a trial version of CommuniGate(tm) Pro*
 Luca [EMAIL PROTECTED] writes:
 
  Hi,
  I'm using linux 2.6.3 with alsalib 1.0.2c. When using alsa output plugin
  of XMMS or mplayer the song is played too fast. Using OSS emulation
  works fine.
 
  This is from bootlog:
 
  intel8x0_measure_ac97_clock: measured 49482 usecs
  intel8x0: clocking to 48000
 
 Make sure the programs are using device default or plughw:0,0 (or
 whatever device numbers you like).  If the programs can't be persuaded
 to do that they are buggy.  

 mplayer is using hw:0,0.

Bug.

 alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian)
 alsa-init: compiled for ALSA-1.0.3
 alsa-init: soundcard set to hw:0
 alsa-init: pcm opend in block-mode
 alsa-init: chunksize set to 1024
 alsa-init: current val=16, fragcount=16
 alsa-init: got buffersize=65536
 alsa9: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian

 It doens't like plugwh:0,0:

 alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian)
 alsa-init: compiled for ALSA-1.0.3
 alsa-init: soundcard set to plughw
 alsa-init: pcm opend in block-mode
 alsa-init: unable to set periodsize: Invalid argument

Bug.

 Btw, mplayer from CVS (checked out today) works nicely with -ao alsa1x:

 alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian)
 alsa-init: compiled for ALSA-1.0.3
 alsa-init: got device=0, subdevice=0
 alsa-init: 1 soundcard found, using: hw:0,0
 alsa-init: pcm opened in block-mode
 alsa-init: chunksize set to 1024
 alsa-init: fragcount=16
 alsa-init: got buffersize=65536
 alsa1x: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian
 ...
 [resample] Changing sample rate to 48000Hz
 [resample] Changing sample rate to 48000Hz
 [resample] Using linear interpolation.

 So my sound card accept sound only at 48000Hz but old mplayer didn't
 notice that alsa-lib doesn't accept its request for 44100Hz. Correct?

So it seems.

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[Alsa-devel] snd_pcm_wait strangeness when resampling

2004-03-11 Thread Måns Rullgård

I've noticed some strange things with snd_pcm_wait.  My application
opens the sound device in non-blocking mode and uses snd_pcm_wait
whenever snd_pcm_writei return EAGAIN.  This works fine as long as
ALSA isn't doing any resampling.  However, when resampling (e.g. from
44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that
there is room in the buffer, after which snd_pcm_writei obviously
returns EAGAIN once more.  The same thing happens if I use OSS
emulation and select() or poll().  Replacing the snd_pcm_wait call
with usleep(1) makes the CPU load look normal again, while the
application still works.

I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a
SIS7012 chip.  It behaves the same way with kernel 2.6.3 and driver
1.0.2c.  I'm not quite sure when it started.

This application used to work properly.  Has something changed so I
need to modify my application?

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[Alsa-devel] Re: [2.6.3] snd_intel8x0: playing to fast

2004-03-10 Thread Måns Rullgård
Luca [EMAIL PROTECTED] writes:

 Hi,
 I'm using linux 2.6.3 with alsalib 1.0.2c. When using alsa output plugin
 of XMMS or mplayer the song is played too fast. Using OSS emulation
 works fine.

 This is from bootlog:

 intel8x0_measure_ac97_clock: measured 49482 usecs
 intel8x0: clocking to 48000

Make sure the programs are using device default or plughw:0,0 (or
whatever device numbers you like).  If the programs can't be persuaded
to do that they are buggy.  Well, those two are buggy anyway.

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[Alsa-devel] Re: When I use xmms with OSS plugin sound is normal with the ALSA plugin sound skips sometimes

2004-02-11 Thread Måns Rullgård
Kristof vansant [EMAIL PROTECTED] writes:

 When I use xmms with OSS plugin sound is normal but with the ALSA plugin
 sound skips sometimes.

 Is this a xmms problem? Or makes OSS emulation the sound stable?

It's an xmms problem.  Read the code and you'll see why.  It can skip
with OSS output too.

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[Alsa-devel] Re: Some proposed ALSA sysfs changes to stock kernel from GregKH

2004-01-20 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Tue, 20 Jan 2004, M?ns Rullg?rd wrote:

 R Chan [EMAIL PROTECTED] writes:
 
  Greg KH is proposing some sysfs changes to the stock kernel - any
  impact on CVS ALSA? Is it ok to rely solely on CVS ALSA - I always
  build 2.6 from the stock tree and then immediately overwrite the
  sound directory with ALSA CVS.
 
 Those patches do not apply cleanly to the latest ALSA version.  There
 are some minor conflicts, all easily resolved.  I'm not sure whether
 anything else depends on the ALSA changes, but I wouldn't think so.
 I've been running ALSA 1.0.1 merged with Greg's patches for some
 time, and it seems to be working well.

 We will merge these patches when they're merged to mainstream.

I suppose someone will also update the drivers as needed.  Only the
intel8x0 driver has been updated.  The rest should still work as
before these patches.

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[Alsa-devel] Using timestamp features when capturing

2004-01-15 Thread Måns Rullgård

I need to find out at what system time a particular sample was
captured.  The various snd_pcm_*_tstamp functions seem to do something
like that, but I can't find any proper documentation for them.  I've
already searched the mail archives without finding anything useful.
Could someone give me a hint?

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[Alsa-devel] Re: Moving from OSS to ALSA

2004-01-11 Thread Måns Rullgård
Lorn Potter [EMAIL PROTECTED] writes:

 On Monday 12 January 2004 4:13 am, James Wright wrote:
  I am currently looking into rewriting our current OSS sound routines to
 native ALSA, as it seems OSS will invariably be phased out now that the
 ALSA driver is distrubuted with the Linux kernel, plus ALSA seems to have a
 great number of benefits for us.

 Personally, I hope OSS compat will never be phased out. Why? OSS is simple, 
 and concise. If I am writing a simple audio recording/playing app, I can get 
 the job done using OSS code in _much_ less lines of code.

I have to disagree.  For my music/video player, I've written both ALSA
and OSS output modules.  The ALSA module is 261 lines, the OSS module
258 lines, including comments and whitespace.  The ALSA module does
things that are more or less impossible with OSS, such as sample
accurate timing.

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[Alsa-devel] Re: weird thing: sound will skip sometimes with OSS in xmms but not if I use esound

2004-01-08 Thread Måns Rullgård
Kristof vansant [EMAIL PROTECTED] writes:

 Does esound do something special to keep sound stable?
 It happens with snd-via82xx + OSS emulation.

I'd rather think xmms does something that causes skips.  I've seen it
with a cmi8738.  It's one of the reasons I stopped using xmms.  It's
actually quite simple to play sound without skips.

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[Alsa-devel] Re: ALSA release 1.0.1

2004-01-08 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Thu, 8 Jan 2004, Kristof vansant wrote:

 Does this mean alsa 1.0.1 will be added to kernel 2.6.1 ?

 No, sorry. This is beyond of my hands. I will send update patches to 
 Andrew Morton and Linus ASAP, but the inclusion is in their hands.

Is there a BK tree anywhere for ALSA?  It would simplify manual
upgrades a bit.

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[Alsa-devel] Re: ALSA release 1.0.1

2004-01-08 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

 On Thu, 8 Jan 2004, M?ns Rullg?rd wrote:

 Is there a BK tree anywhere for ALSA?  It would simplify manual
 upgrades a bit.

 Yes: http://linux-sound.bkbits.net/linux-sound
 The 1.0.1 code is in.

Perfect.  It's working with Linux 2.6.1-rc3 and a sis7012.

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[Alsa-devel] Re: alsa timer slippage

2003-12-14 Thread Måns Rullgård
Jan Depner [EMAIL PROTECTED] writes:

 On Sun, 2003-12-14 at 08:07, Chris Cannam wrote:
 
 While trying to track down the source of some poor timing in 
 sequencing, I've noticed that my ALSA sequencer queue timer has a 
 tendency to fall suddenly behind.
 
 I have a little test program (available on request) that just starts a 
 queue and every second or so compares the queue timer against real 
 time as returned by gettimeofday().  It doesn't mind if the two don't 
 quite match, but it does complain if the difference between the two 
 timers changes dramatically between two sample points.  When I run 
 it, it never lasts for more than about a minute before the ALSA queue 
 timer suddenly slips by anything from 10 to 60 milliseconds.

 If I'm not mistaken the timing for your audio is coming from your sound
 card not your system clock.  The gettimeofday is from the system clock. 
 They probably won't match.

That's true.  However, you typically see a gradual drift, not a sudden
jump.

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[Alsa-devel] Re: plz fix this before alsa 1.0.0 get's released

2003-12-14 Thread Måns Rullgård
Kristof vansant [EMAIL PROTECTED] writes:

 alsa won't compile on kernel 2.6 
 removing sndversions.h in vxp440.c and vxpocket.c does the trick. 

 /home/lupus/alsa-driver-1.0.0rc2/pcmcia/vx/vxp440.c:6:31:
 linux/modversions.h: Onbekend bestand of map
 /home/lupus/alsa-driver-1.0.0rc2/pcmcia/vx/vxp440.c:7:25: sndversions.h:
 Onbekend bestand of map

That's the Dutch message for No such file or directory, right?  It's
nice to set the language to English when posting error messages.

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[Alsa-devel] Re: Tools to help install alsa.

2003-10-06 Thread Måns Rullgård
James Courtier-Dutton [EMAIL PROTECTED] writes:

 2) For alsa to work in kernel 2.6, the first item in the OSS menu has
 to be set to M, otherwise module soundcore does not seem to
 exist. (This is unconfirmed, so might just be user error)

That is not the case.  It works fine here, without anything under OSS
selected.

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[Alsa-devel] Re: Can't record properly with cmi8738 card

2003-09-09 Thread Måns Rullgård
Tobias Deiminger [EMAIL PROTECTED] writes:

 I've got this card (Anubis Typhoon):
 [CMI8738MC6 ]: CMI8738-MC6 - C-Media PCI CMI8738-MC6
  C-Media PCI CMI8738-MC6 (model 55) at 0xd400, irq 10

 and i'm using alsa 0.9.6.

 I want to record something via mic-in using DAP for example. When I
 tried it the first time, i could only record a very low signal
 (although the signal was monitored clean and loud to the
 speakers). Then I turned off the mic-to-center switch as described
 at alsa-project.org. Now the recorded signal was higher, but still
 to low to work properly with it. What's wrong?  Mic-capture is on,
 mic and mic-caputre are set to max and mic-boost is on. Is this a
 general problem with this card/driver?

Mic input works fine with my Hercules Gamesurround DVD 5.1 (or
whatever the name was) based on the same chip, so it can work.  What
type of microphone do you use?

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[Alsa-devel] Re: Can't record properly with cmi8738 card

2003-09-09 Thread Måns Rullgård
Tobias Deiminger [EMAIL PROTECTED] writes:

 Maxtone Moving-Coil Mic Imp. 600 Ohm  but i've got the same
 problem if I use my guitar instead of a mic. I tried also to use my
 v-amp as preamp for the guitar and this works a little better, but
 if I turn the preamp master too loud then the recorded signal gets
 distorted.

Most sound cards expect a cheap electret microphone.  These have much
higher output levels than coil-based ones.  I suggest you use a
pre-amp with output level suitable for the line-in input. Whatever the
usual hifi stuff uses for interconnection is appropriate.

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[Alsa-devel] Memory leaks in alsa-lib

2003-08-26 Thread Måns Rullgård
 (../../../../tcvp/src/tcvp/tcvp.c:328)
==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453)
==3078==by 0x424D7AC9: t_event (../../../../tcvp/src/tcvp/tcvp.c:581)
==3078==by 0x402737B3: thread_wrapper 
(../../../valgrind/coregrind/vg_libpthread.c:667)
==3078==by 0x40175153: do__quit (../../../valgrind/coregrind/vg_scheduler.c:2146)
==3078== 
==3078== 
==3078== 5504 bytes in 172 blocks are still reachable in loss record 17 of 20
==3078==at 0x4002AC23: calloc (../../../valgrind/coregrind/vg_replace_malloc.c:273)
==3078==by 0x4376C73A: _snd_config_make (conf.c:836)
==3078==by 0x4376DD52: snd_config_top (conf.c:1525)
==3078==by 0x43770927: snd_config_update_r (conf.c:3045)
==3078==by 0x43770D67: snd_config_update (conf.c:3095)
==3078==by 0x43789CC6: snd_pcm_open (pcm.c:1929)
==3078==by 0x4373C262: alsa_open (../../../../../tcvp/src/output/alsa/alsa.c:355)
==3078==by 0x424D6E13: new_pipe (../../../../tcvp/src/tcvp/tcvp.c:328)
==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453)
==3078==by 0x424D7AC9: t_event (../../../../tcvp/src/tcvp/tcvp.c:581)
==3078==by 0x402737B3: thread_wrapper 
(../../../valgrind/coregrind/vg_libpthread.c:667)
==3078==by 0x40175153: do__quit (../../../valgrind/coregrind/vg_scheduler.c:2146)
==3078== 
==3078== 
==3078== 9024 bytes in 282 blocks are possibly lost in loss record 19 of 20
==3078==at 0x4002AC23: calloc (../../../valgrind/coregrind/vg_replace_malloc.c:273)
==3078==by 0x4376C73A: _snd_config_make (conf.c:836)
==3078==by 0x4376C80B: _snd_config_make_add (conf.c:862)
==3078==by 0x4376CA4D: parse_value (conf.c:956)
==3078==by 0x4376D0F3: parse_def (conf.c:1184)
==3078==by 0x4376D34D: parse_defs (conf.c:1213)
==3078==by 0x4376CD90: parse_array_def (conf.c:1008)
==3078==by 0x4376CE7B: parse_array_defs (conf.c:1051)
==3078==by 0x4376D16F: parse_def (conf.c:1170)
==3078==by 0x4376D34D: parse_defs (conf.c:1213)
==3078==by 0x4376DE43: snd_config_load1 (conf.c:1544)
==3078==by 0x4376DFDF: snd_config_load (conf.c:1593)
==3078==by 0x437709C5: snd_config_update_r (conf.c:3054)
==3078==by 0x43770D67: snd_config_update (conf.c:3095)
==3078==by 0x43789CC6: snd_pcm_open (pcm.c:1929)
==3078==by 0x4373C262: alsa_open (../../../../../tcvp/src/output/alsa/alsa.c:355)
==3078==by 0x424D6E13: new_pipe (../../../../tcvp/src/tcvp/tcvp.c:328)
==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453)


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Re: [Alsa-devel] Memory leaks in alsa-lib

2003-08-26 Thread Måns Rullgård
Jaroslav Kysela [EMAIL PROTECTED] writes:

   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND  
  7148 mru   11   0 25160  27m 2740 S  0.0 12.4   0:00.08 tcvp 
 
 Now the man page for top tells me that VIRT = SWAP + RES, so unless
 a negative amount has been swapped (what would that mean?), something
 strange is going on.  The odd thing is that this happens only when I
 use the ALSA output module.  Could it be that alsa-lib triggers some
 bug somewhere in top, or in the kernel?  This seems a little unlikely,
 though.

 I will try to investigate this problem.

I seems to be a kernel problem.  See my recent post to linux-kernel on
the subject for more details.

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[Alsa-devel] Re: plughw versus hw (fwd)

2003-07-31 Thread Måns Rullgård
[EMAIL PROTECTED] writes:

 i read that plughw does some magic conversion for samplerate and stuff. 
 how does this magic work. whats the main differenc in the concept of hw 
 and plughw. 

hw interfaces directly to the physical card.  Thus you can only use
modes supported by the hardware.  plughw will detect whether the
hardware can do what you are requesting, and automatically convert the
data to something that can be played, at the expense of CPU time.

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[Alsa-devel] Re: aplay not playing MPEG

2003-07-21 Thread Måns Rullgård
Eliot Blennerhassett [EMAIL PROTECTED] writes:

 Can aplay/alsa send raw MPEG data to a card?

   aplay -Dhw:0  -v -t raw --channels=2 --format=MPEG --rate=44100 ~/audio/44khz
   Playing raw data '/home/eliot/audio/44khz' : MPEG, Rate 44100 Hz, Stereo
   aplay: set_params:805: Sample format non available

 This command works fine if I specify S16_LE, S23, FLOAT, S24, but
 not MPEG.  Any suggestions as to whe - do I have a hope of making it
 work?  (My card supports all the above including MPEG in exactly the
 same way as far as ALSA driver is concerned)

What model is your card?  Are you sure that it really supports mpeg,
and that it isn't just a windows driver faking it for marketing
reasons?

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[Alsa-devel] Re: aplay not playing MPEG

2003-07-21 Thread Måns Rullgård
Eliot Blennerhassett [EMAIL PROTECTED] writes:

 What model is your card?  Are you sure that it really supports mpeg,
 and that it isn't just a windows driver faking it for marketing
 reasons?

 Audioscience ASI6244, definitely has MPEG decode on the card
 (I am one of the cards designers)

Well, I guess you should know.  I've some Soundblaster cards marketed
as MP3, even though they can't do anything of the kind in hardware.

 I am writing the driver for this card, but can't figure out why the
 MPEG doesn't even make it to the driver.  My lack of understanding
 of alsa-lib I expect.

It's nice that someone writes drivers for their own cards, but I'm
afraid I can't help you.

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[Alsa-devel] Re: ring buffer pointer accuracy.

2003-07-14 Thread Måns Rullgård
Giuliano Pochini [EMAIL PROTECTED] writes:

 I would like some idea of how accurate the snd_pcm_delay() result is
 likely to be. If I am trying to syncronise video to audio, and I notice
 a difference between audio and video, I need to know how accurate I can
 make it.

 I suppose you need a precision of 42ms (the duration of a cinema frame).
 You can read the position at intervals = 42ms. If the value you read is
 different then the previous one, you know that value is up to date.
 You know exactly when you started to play the sound. If you do your own
 timing and you take into account the period time, you only need to read
 position 3-4 times every 10s or so to check if you are losing sync.

You could use the ALSA timer interface.  It gives a resolution equal
to the period time.  If that's too long time, you can use the system
clock to interpolate.

 Even if you have a really bad soundcard, it's unlikely the sample clock
 skews more than 1%.

1% would be about a minute at the end of normal movie.  That's not
what I like to call acceptable.

 If some cards only update delay once per period, but others update it
 all the time, I would like the application to know about it, so it can
 decide how accurately it can actually get audio and video in sync.

 You can use shorter periods if your cards allows it.

Most cards I've come across support period lengths of down to 10 ms or
less.  That's plenty enough for video playback.

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[Alsa-devel] Re: Fwd: RE: To ESS Tech Support

2003-06-13 Thread Måns Rullgård
Warren Turkal [EMAIL PROTECTED] writes:

 Another message from the ESS people. If some other people could take
 the time to request this info from ESS also, maybe they would be
 more forthcoming with the info.

Tell them you want the complete specs for the chip, and are willing to
sign an NDA.  That might help.

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[Alsa-devel] Re: Fwd: RE: To ESS Tech Support

2003-06-13 Thread Måns Rullgård
P. Christeas [EMAIL PROTECTED] writes:

  Warren Turkal [EMAIL PROTECTED] writes:
   Another message from the ESS people. If some other people could take
   the time to request this info from ESS also, maybe they would be
   more forthcoming with the info.
 
  Tell them you want the complete specs for the chip, and are willing to
  sign an NDA.  That might help.
 
 That may not be accurate. If you're writing a driver for linux, you most 
 probably want to publish it under GPL, which definitely conflicts with the 
 NDA. The NDA should be about the chip only, not the driver.

You could get permission to release source code based on the docs.  I
got that from 3Dlabs.

 The real case you should stick at (the one for any such case) is
 that nobody could actually use the linux driver without owning the
 product, for which we've paid. Tell them that by supporting (it's so
 easy for them) linux, they get themselves out of the non-desirable
 hardware list. We (linux users) have several lists with hw and
 companies we should avoid. ESS is one of them.

This is of course preferred, but it doesn't always help.

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[Alsa-devel] Re: Fwd: RE: To ESS Tech Support

2003-06-12 Thread Måns Rullgård
Allan Klinbail [EMAIL PROTECTED] writes:

  Here is a message from ESS regarding getting some information for
  the ESS1988/Allegro-1 series cards in order to get better support
  for them in linux. I hope that some others will help me badger the
  ESS people in order to (A) release info needed to make the Linux
  driver or (B) release the Windows driver source so that it can be
  used for ideas for the Linux driver.
  
 Looks like they didn't read your email...
 
 Essentially you are offering them a favour and they see the word linux
 and run away scared.. 

It may very well have been an automatic reply triggered by the work
linux in the question.  I've seen such things before.

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[Alsa-devel] [PATCH] Fix pause function

2003-06-04 Thread Måns Rullgård

This patch fixes the kernel side of snd_pcm_pause() to actually pause
instead of always returning -EBADFD.  The patch is against
alsa-driver-0.9.4, same as CVS HEAD for this file.

--- acore/pcm_native.c~ Tue Jun  3 13:48:09 2003
+++ acore/pcm_native.c  Tue Jun  3 19:27:06 2003
@@ -863,7 +863,7 @@
 
 static int snd_pcm_pause(snd_pcm_substream_t *substream, int push)
 {
-   return snd_pcm_action(snd_pcm_action_pause, substream, 0);
+   return snd_pcm_action(snd_pcm_action_pause, substream, push);
 }
 
 #ifdef CONFIG_PM


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[Alsa-devel] 4/6-channel output on CMI8738

2003-06-03 Thread Måns Rullgård

I'm trying to get 4/6-channel playback to work with a CMI8738 chip.
It's behaving a little oddly, however.

First I have to switch off Four Channel Mode, which I guess is
normal.  With that switch off, I can play 4 channels on hw:0,1 or
surround40.  The problem is that the front output is replicated on the
rear jack.  This is cured by turning on Exchange DAC.  The PCM
volume control also works in this mode, but the Master control only
affects front output.  Anyhow, 4-channel playback is pretty much OK.

Next I try 6-channel playback.  With the same settings it works,
except that the volume control doesn't affect channels 5 and 6.  With
Exchange DAC off, the front output is silent, but channels 1 and 2
are sent to the rear instead.

All this would be acceptable if it wasn't for some quirks.  Sometimes,
the right rear output goes to maximum volume, with no apparent reason,
and a few times the output has suddenly stopped completely.  This has
happened while playing with the mixer settings.  Stopping and
restarting the playback program helps.

Finally, the surround51 device behaves like a duplicate of hw:0,0.

Am I doing something wrong, or is the driver just not perfect?

Here's my /proc/asound/cards:
0 [CMI8738MC6 ]: CMI8738-MC6 - C-Media PCI CMI8738-MC6
 C-Media PCI CMI8738-MC6 (model 55) at 0x8800, irq 24

and /proc/asound/version:
Advanced Linux Sound Architecture Driver Version 0.9.2.
Compiled on Mar 31 2003 for kernel 2.4.21-pre5.

Maybe I should upgrade, but I haven't noticed any changes to the
cmipci driver that seem related to the problems.

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