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
@@ -158,151 +158,159 @@ void AudioDevice::setVolume( unsigned int leftVolume, unsigned int rightVolume,
 
 # if 0
     unsigned int lV = (leftVolume  * 101) >> 16;
     unsigned int volume = ((rV << 8) & 0xFF00) | (lV & 0x00FF);
     int mixerHandle = 0;
     if ( ( mixerHandle = open( "/dev/mixer", O_RDWR ) ) >= 0 ) {
         if(ioctl( mixerHandle, MIXER_WRITE(0), &volume ) ==-1)
             perror("ioctl(\"MIXER_WRITE\")");
         close( mixerHandle );
     } else
         perror("open(\"/dev/mixer\")");
 
 # else
     // This is the way this has to be done now I guess, doesn't allow for
     // independant right and left channel setting, or setting for different outputs
     Config cfg("Sound");
     cfg.setGroup("System");
     cfg.writeEntry("Volume",(int)rV);
 # endif
 
 #endif
 //    qDebug( "setting volume to: 0x%x", volume ); 
 #if ( defined Q_WS_QWS || defined(_WS_QWS_) ) && !defined(QT_NO_COP)
     // Send notification that the volume has changed
     QCopEnvelope( "QPE/System", "volumeChange(bool)" ) << muted; 
 #endif
 }
 
 
 
 
 AudioDevice::AudioDevice( unsigned int f, unsigned int chs, unsigned int bps ) {
   qDebug("creating new audio device");
   d = new AudioDevicePrivate;
     d->frequency = f;
     d->channels = chs;
     d->bytesPerSample = bps;
     qDebug("%d",bps);
     int format=0;
     if( bps == 8)  format  = AFMT_U8;
     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 
   AudioDevicePrivate::dspFd = d->handle;
     } else {
         d->handle = AudioDevicePrivate::dspFd;
     }
 #endif
 
     if(ioctl( d->handle, SNDCTL_DSP_GETCAPS, &capabilities )==-1)
         perror("ioctl(\"SNDCTL_DSP_GETCAPS\")");
     if(ioctl( d->handle, SNDCTL_DSP_SETFRAGMENT, &fragments )==-1)
         perror("ioctl(\"SNDCTL_DSP_SETFRAGMENT\")");
     if(ioctl( d->handle, SNDCTL_DSP_SETFMT, & format )==-1)
         perror("ioctl(\"SNDCTL_DSP_SETFMT\")");
     qDebug("freq %d", d->frequency);
     if(ioctl( d->handle, SNDCTL_DSP_SPEED, &d->frequency )==-1)
         perror("ioctl(\"SNDCTL_DSP_SPEED\")");
     qDebug("channels %d",d->channels);
     if ( ioctl( d->handle, SNDCTL_DSP_CHANNELS, &d->channels ) == -1 ) {
         d->channels = ( d->channels == 1 ) ? 2 : d->channels;
         if(ioctl( d->handle, SNDCTL_DSP_CHANNELS, &d->channels )==-1)
             perror("ioctl(\"SNDCTL_DSP_CHANNELS\")");
     }
 
     d->bufferSize = sound_fragment_bytes;
     d->unwrittenBuffer = new char[d->bufferSize];
     d->unwritten = 0;
     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;
 }
 
 
 void AudioDevice::write( char *buffer, unsigned int length )
 {
 #ifdef Q_OS_WIN32
     // returns immediately and (to be implemented) emits completedIO() when finished writing
     WAVEHDR *lpWaveHdr = (WAVEHDR *)malloc( sizeof(WAVEHDR) );
     // maybe the buffer should be copied so that this fool proof, but its a performance hit
     lpWaveHdr->lpData = buffer;
     lpWaveHdr->dwBufferLength = length;
     lpWaveHdr->dwFlags = 0L;
     lpWaveHdr->dwLoops = 0L;
     waveOutPrepareHeader( (HWAVEOUT)d->handle, lpWaveHdr, sizeof(WAVEHDR) );
     // waveOutWrite returns immediately. the data is sent in the background. 
     if ( waveOutWrite( (HWAVEOUT)d->handle, lpWaveHdr, sizeof(WAVEHDR) ) )
   qDebug( "failed to write block to audio device" );
     // emit completedIO();
 #else
     int t = ::write( d->handle, buffer, length );
     if ( t<0 ) t = 0;
     if ( t != (int)length) {
   qDebug("Ahhh!! memcpys 1");
   memcpy(d->unwrittenBuffer,buffer+t,length-t);
   d->unwritten = length-t;
     }
 #endif
 }
 
 
 unsigned int AudioDevice::channels() const
 {
     return d->channels;
 }
 
 
 unsigned int AudioDevice::frequency() const
 {
     return d->frequency;
 }
 
 
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
@@ -192,104 +192,104 @@ void LoopControl::startVideo() {
     current_frame = long( playtime.elapsed() * framerate / 1000 );
 
     if ( prev_frame != -1 && current_frame <= prev_frame )
         return;
 
       } else {
     // Don't skip
     current_frame++;
       }
 
       if ( prev_frame == -1 || current_frame > prev_frame ) {
     if ( current_frame > prev_frame + 1 ) {
         mediaPlayerState->curDecoder()->videoSetFrame( current_frame, stream );
     }
     moreVideo = videoUI->playVideo();
     prev_frame = current_frame;
       }
 
   } else {
 
       moreVideo = FALSE;
       killTimer( videoId );
 
   }
 
     }
 }
 
 
 void LoopControl::startAudio() {
     
 //qDebug("start audio");
     audioMutex->lock();
     if ( moreAudio ) {
 
   if ( !isMuted && mediaPlayerState->curDecoder() ) {
 
       currentSample = audioSampleCounter + 1;
 
       if ( currentSample != audioSampleCounter + 1 )
     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);
 
   } else {
 
       moreAudio = FALSE;
 
   }
 
     }
 
     audioMutex->unlock();
 }
 
 
 void LoopControl::killTimers() {
 
     audioMutex->lock();
 
     if ( hasVideoChannel ) 
   killTimer( videoId );
     killTimer( sliderId );
     threadOkToGo = FALSE;
 
     audioMutex->unlock();
 }
 
 
 void LoopControl::startTimers() {
 
     audioMutex->lock();
 
     moreVideo = FALSE;
     moreAudio = FALSE;
 
     if ( hasVideoChannel ) {
   moreVideo = TRUE;
   int mSecsBetweenFrames = (int)(100 / framerate); // 10% of the real value
   videoId = startTimer( mSecsBetweenFrames );
     }
 
     if ( hasAudioChannel ) {
   moreAudio = TRUE;