attempted to look at stumbling issues, and pops while starting and stoping audio device. needs to be re done

2 files changed, 15 insertions, 7 deletions

diff --git a/core/multimedia/opieplayer/audiodevice.cpp b/core/multimedia/opieplayer/audiodevice.cpp
index ad44abb..e0989d0 100644
--- a/core/multimedia/opieplayer/audiodevice.cpp
+++ b/core/multimedia/opieplayer/audiodevice.cpp
@@ -198,16 +198,18 @@ AudioDevice::AudioDevice( unsigned int f, unsigned int chs, unsigned int bps ) {
     else if( bps <= 0) format = AFMT_S16_LE;
     else format = AFMT_S16_LE;
     qDebug("AD- freq %d, channels %d, b/sample %d, bitrate %d",f,chs,bps,format);
     connect( qApp, SIGNAL( volumeChanged(bool) ), this, SLOT( volumeChanged(bool) ) );
 
     int fragments = 0x10000 * 8 + sound_fragment_shift;
     int capabilities = 0;
 
+    QCopEnvelope( "QPE/System", "volumeChange(bool)" ) << TRUE; 
+    
 #ifdef KEEP_DEVICE_OPEN
     if ( AudioDevicePrivate::dspFd == 0 ) {
 #endif
     if ( ( d->handle = ::open( "/dev/dsp", O_WRONLY ) ) < 0 ) {
         perror("open(\"/dev/dsp\") sending to /dev/null instead");
         d->handle = ::open( "/dev/null", O_WRONLY );
     }
 #ifdef KEEP_DEVICE_OPEN 
@@ -239,30 +241,36 @@ AudioDevice::AudioDevice( unsigned int f, unsigned int chs, unsigned int bps ) {
     d->can_GETOSPACE = TRUE; // until we find otherwise
  
     //if ( chs != d->channels )       qDebug( "Wanted %d, got %d channels", chs, d->channels );
     //if ( f != d->frequency )        qDebug( "wanted %dHz, got %dHz", f, d->frequency );
     //if ( capabilities & DSP_CAP_BATCH )   qDebug( "Sound card has local buffer" );
     //if ( capabilities & DSP_CAP_REALTIME )qDebug( "Sound card has realtime sync" );
     //if ( capabilities & DSP_CAP_TRIGGER ) qDebug( "Sound card has precise trigger" );
     //if ( capabilities & DSP_CAP_MMAP )    qDebug( "Sound card can mmap" );
+    QCopEnvelope( "QPE/System", "volumeChange(bool)" ) << FALSE; 
+    
 }
     
 
 AudioDevice::~AudioDevice() {
     qDebug("destryo audiodevice");
+    QCopEnvelope( "QPE/System", "volumeChange(bool)" ) << TRUE; 
+    
 #ifdef Q_OS_WIN32
     waveOutClose( (HWAVEOUT)d->handle );
 #else
 # ifndef KEEP_DEVICE_OPEN 
     close( d->handle );     // Now it should be safe to shut the handle
 # endif
     delete d->unwrittenBuffer;
     delete d;
 #endif
+    QCopEnvelope( "QPE/System", "volumeChange(bool)" ) << FALSE; 
+    
 }
 
 
 void AudioDevice::volumeChanged( bool muted )
 {
     AudioDevicePrivate::muted = muted;
 }
 
diff --git a/core/multimedia/opieplayer/loopcontrol.cpp b/core/multimedia/opieplayer/loopcontrol.cpp
index 6f86b4a..3171c4b 100644
--- a/core/multimedia/opieplayer/loopcontrol.cpp
+++ b/core/multimedia/opieplayer/loopcontrol.cpp
@@ -232,24 +232,24 @@ void LoopControl::startAudio() {
     qDebug("out of sync with decoder %i %i", currentSample, audioSampleCounter);
 
       long samplesRead = 0;
       bool readOk=mediaPlayerState->curDecoder()->audioReadSamples( (short*)audioBuffer, channels, 1024, samplesRead, stream );
       long sampleWeShouldBeAt = long( playtime.elapsed() ) * freq / 1000;
       long sampleWaitTime = currentSample - sampleWeShouldBeAt;
 
 // this causes drop outs not sure why its even here
-//          if ( ( sampleWaitTime > 2000 ) && ( sampleWaitTime < 20000 ) ) {
-//        usleep( (long)((double)sampleWaitTime * 1000000.0 / freq) );
-//          }
-//        else if ( sampleWaitTime <= -5000 ) {
-//       qDebug("need to catch up by: %li (%i,%li)", -sampleWaitTime, currentSample, sampleWeShouldBeAt );
+          if ( ( sampleWaitTime > 2000 ) && ( sampleWaitTime < 20000 ) ) {
+        usleep( (long)((double)sampleWaitTime * 1000000.0 / freq) );
+          }
+        else if ( sampleWaitTime <= -5000 ) {
+       qDebug("need to catch up by: %li (%i,%li)", -sampleWaitTime, currentSample, sampleWeShouldBeAt );
 //       //mediaPlayerState->curDecoder()->audioSetSample( sampleWeShouldBeAt, stream );
-//       currentSample = sampleWeShouldBeAt;
-//         }
+       currentSample = sampleWeShouldBeAt;
+         }
 
       audioDevice->write( audioBuffer, samplesRead * 2 * channels );
 
       if(  mediaPlayerState->isStreaming == FALSE)
       audioSampleCounter = currentSample + samplesRead - 1;
 
       moreAudio = readOk && (audioSampleCounter <= total_audio_samples);