qint64 QAudioOutputPrivate::elapsedUSecs() const
{
if(!handle)
return 0;
if (deviceState == QAudio::StoppedState)
return 0;
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
snd_pcm_status_t* status;
snd_pcm_status_alloca(&status);
snd_timestamp_t t1,t2;
if( snd_pcm_status(handle, status) >= 0) {
snd_pcm_status_get_tstamp(status,&t1);
snd_pcm_status_get_trigger_tstamp(status,&t2);
t1.tv_sec-=t2.tv_sec;
signed long l = (signed long)t1.tv_usec - (signed long)t2.tv_usec;
if(l < 0) {
t1.tv_sec--;
l = -l;
l %= 1000000;
}
return ((t1.tv_sec * 1000000)+l);
} else
return 0;
#else
return clockStamp.elapsed()*1000;
#endif
return 0;
}
void *CallbackThread( void *userData )
{
PaAlsaStream *stream = (PaAlsaStream*)userData;
pthread_cleanup_push( &Stop, stream ); // Execute Stop on exit
if( stream->pcm_playback )
snd_pcm_start( stream->pcm_playback );
else if( stream->pcm_capture )
snd_pcm_start( stream->pcm_capture );
while(1)
{
int frames_avail;
int frames_got;
PaStreamCallbackTimeInfo timeInfo = {0,0,0}; /* IMPLEMENT ME */
int callbackResult;
int framesProcessed;
pthread_testcancel();
{
/* calculate time info */
snd_timestamp_t capture_timestamp;
snd_timestamp_t playback_timestamp;
snd_pcm_status_t *capture_status;
snd_pcm_status_t *playback_status;
snd_pcm_status_alloca( &capture_status );
snd_pcm_status_alloca( &playback_status );
if( stream->pcm_capture )
{
snd_pcm_status( stream->pcm_capture, capture_status );
snd_pcm_status_get_tstamp( capture_status, &capture_timestamp );
}
if( stream->pcm_playback )
{
snd_pcm_status( stream->pcm_playback, playback_status );
snd_pcm_status_get_tstamp( playback_status, &playback_timestamp );
}
/* Hmm, we potentially have both a playback and a capture timestamp.
* Hopefully they are the same... */
if( stream->pcm_capture && stream->pcm_playback )
{
float capture_time = capture_timestamp.tv_sec +
((float)capture_timestamp.tv_usec/1000000);
float playback_time= playback_timestamp.tv_sec +
((float)playback_timestamp.tv_usec/1000000);
if( fabsf(capture_time-playback_time) > 0.01 )
PA_DEBUG(("Capture time and playback time differ by %f\n", fabsf(capture_time-playback_time)));
timeInfo.currentTime = capture_time;
}
else if( stream->pcm_playback )
{
timeInfo.currentTime = playback_timestamp.tv_sec +
((float)playback_timestamp.tv_usec/1000000);
}
else
{
timeInfo.currentTime = capture_timestamp.tv_sec +
((float)capture_timestamp.tv_usec/1000000);
}
if( stream->pcm_capture )
{
snd_pcm_sframes_t capture_delay = snd_pcm_status_get_delay( capture_status );
timeInfo.inputBufferAdcTime = timeInfo.currentTime -
(float)capture_delay / stream->streamRepresentation.streamInfo.sampleRate;
}
if( stream->pcm_playback )
{
snd_pcm_sframes_t playback_delay = snd_pcm_status_get_delay( playback_status );
timeInfo.outputBufferDacTime = timeInfo.currentTime +
(float)playback_delay / stream->streamRepresentation.streamInfo.sampleRate;
}
}
/*
IMPLEMENT ME:
- handle buffer slips
*/
/*
depending on whether the host buffers are interleaved, non-interleaved
or a mixture, you will want to call PaUtil_ProcessInterleavedBuffers(),
PaUtil_ProcessNonInterleavedBuffers() or PaUtil_ProcessBuffers() here.
*/
framesProcessed = frames_avail = wait( stream );
while( frames_avail > 0 )
{
//PA_DEBUG(( "%d frames available\n", frames_avail ));
/* Now we know the soundcard is ready to produce/receive at least
* one period. We just need to get the buffers for the client
* to read/write. */
PaUtil_BeginBufferProcessing( &stream->bufferProcessor, &timeInfo,
0 /* @todo pass underflow/overflow flags when necessary */ );
frames_got = setup_buffers( stream, frames_avail );
PaUtil_BeginCpuLoadMeasurement( &stream->cpuLoadMeasurer );
callbackResult = paContinue;
/* this calls the callback */
framesProcessed = PaUtil_EndBufferProcessing( &stream->bufferProcessor,
&callbackResult );
PaUtil_EndCpuLoadMeasurement( &stream->cpuLoadMeasurer, framesProcessed );
/* inform ALSA how many frames we wrote */
if( stream->pcm_capture )
snd_pcm_mmap_commit( stream->pcm_capture, stream->capture_offset, frames_got );
if( stream->pcm_playback )
snd_pcm_mmap_commit( stream->pcm_playback, stream->playback_offset, frames_got );
if( callbackResult != paContinue )
break;
frames_avail -= frames_got;
}
/*
If you need to byte swap outputBuffer, you can do it here using
routines in pa_byteswappers.h
*/
if( callbackResult != paContinue )
{
stream->callback_finished = 1;
stream->callbackAbort = (callbackResult == paAbort);
pthread_exit( NULL );
}
}
/* This code is unreachable, but important to include regardless because it
* is possibly a macro with a closing brace to match the opening brace in
* pthread_cleanup_push() above. The documentation states that they must
* always occur in pairs. */
pthread_cleanup_pop( 1 );
}
