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* Basically, the audio buffer size was simply too small and caused

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audio playback with frequent glitches. I am testing with HDA and have
  perfect audio now. The only remaining problem is the drop-sample
  resampling in the system mixer, which gives a somewhat metallic sound.
* Removed hardcoded values from preferred format and used them if
  the consumer supplies wildcard values. Since the system mixer supplies
  wildcards for all those values, this change does not have any effect
  for now. The code is more correct, though.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@29422 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
Stephan Aßmus 2009-03-07 11:39:48 +00:00
parent 0bd93b7def
commit 46f2d7feec
1 changed files with 34 additions and 13 deletions
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47
src/apps/mediaplayer/media_node_framework/audio/AudioProducer.cpp
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@ -108,19 +108,25 @@ AudioProducer::AudioProducer(const char* name, AudioSupplier* supplier,
fPreferredFormat.u.raw_audio.format
= media_raw_audio_format::B_AUDIO_FLOAT;
// = media_raw_audio_format::B_AUDIO_SHORT;
fPreferredFormat.u.raw_audio.channel_count = 2;
fPreferredFormat.u.raw_audio.frame_rate = 44100.0;
fPreferredFormat.u.raw_audio.byte_order
= (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN;
#if 0
fPreferredFormat.u.raw_audio.channel_count = 2;
fPreferredFormat.u.raw_audio.frame_rate = 44100.0;
// NOTE: the (buffer_size * 1000000) needs to be dividable by
// fPreferredFormat.u.raw_audio.frame_rate!
fPreferredFormat.u.raw_audio.buffer_size = 441 * 2
fPreferredFormat.u.raw_audio.buffer_size = 441 * 4
* (fPreferredFormat.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK);
if (!fLowLatency)
fPreferredFormat.u.raw_audio.buffer_size *= 6;
fPreferredFormat.u.raw_audio.buffer_size *= 3;
#else
fPreferredFormat.u.raw_audio.channel_count = 0;
fPreferredFormat.u.raw_audio.frame_rate = 0.0;
fPreferredFormat.u.raw_audio.buffer_size = 0;
#endif
// we're not connected yet
fOutput.destination = media_destination::null;
@ -340,21 +346,36 @@ AudioProducer::PrepareToConnect(const media_source& what,
return B_MEDIA_BAD_FORMAT;
}
// !!! validate all other fields except for buffer_size here, because the
// consumer might have supplied different values from AcceptFormat()?
if (format->u.raw_audio.channel_count
== media_raw_audio_format::wildcard.channel_count) {
format->u.raw_audio.channel_count = 2;
printf(" -> adjusting channel count, it was wildcard\n");
}
if (format->u.raw_audio.frame_rate
== media_raw_audio_format::wildcard.frame_rate) {
format->u.raw_audio.frame_rate = 44100.0;
printf(" -> adjusting frame rate, it was wildcard\n");
}
// check the buffer size, which may still be wildcarded
if (format->u.raw_audio.buffer_size
== media_raw_audio_format::wildcard.buffer_size) {
// pick something comfortable to suggest
// NOTE: the (buffer_size * 1000000) needs to be dividable by
// fPreferredFormat.u.raw_audio.frame_rate!
// TODO: this needs to depend on the other parameters
// (but it doesn't matter sincer the AudioProducer is not
// currently used like that)
TRACE(" -> adjusting buffer size, it was wildcard\n");
format->u.raw_audio.buffer_size = 441 * 2 * sizeof(float);
}
printf(" -> adjusting buffer size, it was wildcard\n");
// NOTE: the (buffer_size * 1000000) needs to be dividable by
// format->u.raw_audio.frame_rate! (We assume frame rate is a multiple of
// 25, which it usually is.)
format->u.raw_audio.buffer_size
= uint32(format->u.raw_audio.frame_rate / 25.0)
* (format->u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK);
if (!fLowLatency)
format->u.raw_audio.buffer_size *= 3;
}
// Now reserve the connection, and return information about it
fOutput.destination = where;