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
@@ -14,357 +14,365 @@
 ** See http://www.trolltech.com/gpl/ for GPL licensing information.
 **
 ** Contact info@trolltech.com if any conditions of this licensing are
 ** not clear to you.
 **
 **********************************************************************/
 // L.J.Potter added better error code Fri 02-15-2002 14:37:47
 
 #include <stdlib.h>
 #include <stdio.h>
 #include <qpe/qpeapplication.h>
 #include <qpe/config.h>
 #include "audiodevice.h"
 
 #if ( defined Q_WS_QWS || defined(_WS_QWS_) ) && !defined(QT_NO_COP)
 #include "qpe/qcopenvelope_qws.h"
 #endif
 
 #ifdef Q_WS_WIN
 #include <windows.h>
 #include <mmsystem.h>
 #include <mmreg.h>
 #endif
 
 #if defined(Q_WS_X11) || defined(Q_WS_QWS)
 #include <fcntl.h>
 #include <sys/ioctl.h>
 #include <sys/soundcard.h>
 #include <sys/stat.h>
 #include <sys/time.h>
 #include <sys/types.h>
 #include <unistd.h>
 #endif
 
 #if defined(Q_OS_WIN32)
 static const int expectedBytesPerMilliSecond = 2 * 2 * 44000 / 1000;
 static const int timerResolutionMilliSeconds = 30;
 static const int sound_fragment_bytes = timerResolutionMilliSeconds * expectedBytesPerMilliSecond;
 #else
 # if defined(QT_QWS_IPAQ)
 static const int sound_fragment_shift = 14;
 # else
 static const int sound_fragment_shift = 16;
 # endif
 static const int sound_fragment_bytes = (1<<sound_fragment_shift);
 #endif
 
 
 class AudioDevicePrivate {
 public:
     int handle;
     unsigned int frequency;
     unsigned int channels;
     unsigned int bytesPerSample;
     unsigned int bufferSize;
 #ifndef Q_OS_WIN32
     bool can_GETOSPACE;
     char* unwrittenBuffer;
     unsigned int unwritten;
 #endif
 
     static int dspFd;
     static bool muted;
     static unsigned int leftVolume;
     static unsigned int rightVolume;
 };
 
 
 #ifdef Q_WS_QWS
 // This is for keeping the device open in-between playing files when
 // the device makes clicks and it starts to drive you insane! :)
 // Best to have the device not open when not using it though
 //#define KEEP_DEVICE_OPEN
 #endif
 
 
 int AudioDevicePrivate::dspFd = 0;
 bool AudioDevicePrivate::muted = FALSE;
 unsigned int AudioDevicePrivate::leftVolume = 0;
 unsigned int AudioDevicePrivate::rightVolume = 0;
 
 
 void AudioDevice::getVolume( unsigned int& leftVolume, unsigned int& rightVolume, bool &muted ) {
     muted = AudioDevicePrivate::muted;
     unsigned int volume;
 #ifdef Q_OS_WIN32
     HWAVEOUT handle;
     WAVEFORMATEX formatData;
     formatData.cbSize = sizeof(WAVEFORMATEX);
     formatData.wFormatTag = WAVE_FORMAT_PCM;
     formatData.nAvgBytesPerSec = 4 * 44000;
     formatData.nBlockAlign = 4;
     formatData.nChannels = 2;
     formatData.nSamplesPerSec = 44000;
     formatData.wBitsPerSample = 16;
     waveOutOpen(&handle, WAVE_MAPPER, &formatData, 0L, 0L, CALLBACK_NULL);
     if ( waveOutGetVolume( handle, (LPDWORD)&volume ) )
 //  qDebug( "get volume of audio device failed" );
     waveOutClose( handle );
     leftVolume = volume & 0xFFFF;
     rightVolume = volume >> 16;
 #else
     int mixerHandle = open( "/dev/mixer", O_RDWR );
     if ( mixerHandle >= 0 ) {
         if(ioctl( mixerHandle, MIXER_READ(0), &volume )==-1)
         perror("ioctl(\"MIXER_READ\")");
         close( mixerHandle );
     } else
         perror("open(\"/dev/mixer\")");
     leftVolume  = ((volume & 0x00FF) << 16) / 101;
     rightVolume = ((volume & 0xFF00) <<  8) / 101;
 #endif
 }
 
 
 void AudioDevice::setVolume( unsigned int leftVolume, unsigned int rightVolume, bool muted ) {
     AudioDevicePrivate::muted = muted;
     if ( muted ) {
   AudioDevicePrivate::leftVolume = leftVolume;
   AudioDevicePrivate::rightVolume = rightVolume;
   leftVolume = 0;
   rightVolume = 0;
     } else {
         leftVolume  = ( (int) leftVolume < 0 ) ? 0 : ((  leftVolume > 0xFFFF ) ? 0xFFFF :  leftVolume );
   rightVolume = ( (int)rightVolume < 0 ) ? 0 : (( rightVolume > 0xFFFF ) ? 0xFFFF : rightVolume );
     }
 #ifdef Q_OS_WIN32
     HWAVEOUT handle;
     WAVEFORMATEX formatData;
     formatData.cbSize = sizeof(WAVEFORMATEX);
     formatData.wFormatTag = WAVE_FORMAT_PCM;
     formatData.nAvgBytesPerSec = 4 * 44000;
     formatData.nBlockAlign = 4;
     formatData.nChannels = 2;
     formatData.nSamplesPerSec = 44000;
     formatData.wBitsPerSample = 16;
     waveOutOpen(&handle, WAVE_MAPPER, &formatData, 0L, 0L, CALLBACK_NULL);
     unsigned int volume = (rightVolume << 16) | leftVolume;
     if ( waveOutSetVolume( handle, volume ) )
 //  qDebug( "set volume of audio device failed" );
     waveOutClose( handle );
 #else
     // Volume can be from 0 to 100 which is 101 distinct values
     unsigned int rV = (rightVolume * 101) >> 16;
 
 # 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;
 }
 
 
 unsigned int AudioDevice::bytesPerSample() const
 {
     return d->bytesPerSample;
 }
 
 
 unsigned int AudioDevice::bufferSize() const
 {
     return d->bufferSize;
 }
 
 unsigned int AudioDevice::canWrite() const
 {
 #ifdef Q_OS_WIN32
     return bufferSize(); // Any better?
 #else
     audio_buf_info info;
     if ( d->can_GETOSPACE && ioctl(d->handle,SNDCTL_DSP_GETOSPACE,&info) ) {
   d->can_GETOSPACE = FALSE;
   fcntl( d->handle, F_SETFL, O_NONBLOCK );
     }
     if ( d->can_GETOSPACE ) {
   int t = info.fragments * sound_fragment_bytes;
   return QMIN(t,(int)bufferSize());
     } else {
   if ( d->unwritten ) {
       int t = ::write( d->handle, d->unwrittenBuffer, d->unwritten );
       if ( t<0 ) t = 0;
       if ( (unsigned)t!=d->unwritten ) {
     memcpy(d->unwrittenBuffer,d->unwrittenBuffer+t,d->unwritten-t);
     d->unwritten -= t;
       } else {
     d->unwritten = 0;
       }
   }
   if ( d->unwritten )
       return 0;
   else
       return d->bufferSize;
     }
 #endif
 }
 
 
 int AudioDevice::bytesWritten() {
 #ifdef Q_OS_WIN32
     MMTIME pmmt = { TIME_BYTES, 0 };
     if ( ( waveOutGetPosition( (HWAVEOUT)d->handle, &pmmt, sizeof(MMTIME) ) != MMSYSERR_NOERROR ) || ( pmmt.wType != TIME_BYTES ) ) {
   qDebug( "failed to get audio device position" );
   return -1;
     }
     return pmmt.u.cb;
 #else
     int buffered = 0;
     if ( ioctl( d->handle, SNDCTL_DSP_GETODELAY, &buffered ) ) {
   qDebug( "failed to get audio device position" );
   return -1;
     }
     return buffered;
 #endif
 }
 
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
@@ -48,392 +48,392 @@ extern MediaPlayerState *mediaPlayerState;
 static char     *audioBuffer = NULL;
 static AudioDevice  *audioDevice = NULL;
 static bool     disabledSuspendScreenSaver = FALSE;
 static bool     previousSuspendMode = FALSE;
 
 
 pthread_t audio_tid;
 pthread_attr_t  audio_attr;
 bool threadOkToGo = FALSE;
 
 
 class Mutex {
 public:
     Mutex() {
   pthread_mutexattr_t attr;
   pthread_mutexattr_init( &attr );
   pthread_mutex_init( &mutex, &attr );
   pthread_mutexattr_destroy( &attr );
     }
 
     ~Mutex() {
   pthread_mutex_destroy( &mutex );
     }
 
     void lock() {
   pthread_mutex_lock( &mutex );
     }
 
     void unlock() {
   pthread_mutex_unlock( &mutex );
     }
 private:
     pthread_mutex_t mutex;
 };
 
 
 void *startAudioThread( void *ptr ) {
     LoopControl *mpegView = (LoopControl *)ptr;
     while ( TRUE ) {
   if ( threadOkToGo && mpegView->moreAudio )
       mpegView->startAudio();
   else
       usleep( 10000 ); // Semi-buzy-wait till we are playing again
     }
     return 0;
 }
 
 
 Mutex *audioMutex;
 
 
 LoopControl::LoopControl( QObject *parent, const char *name )
     : QObject( parent, name ) {
     isMuted = FALSE;
     connect( qApp, SIGNAL( volumeChanged(bool) ), this, SLOT( setMute(bool) ) );
 //qDebug("starting loopcontrol");
     audioMutex = new Mutex;
 
     pthread_attr_init(&audio_attr);
 #define USE_REALTIME_AUDIO_THREAD
 #ifdef USE_REALTIME_AUDIO_THREAD
     // Attempt to set it to real-time round robin
     if ( pthread_attr_setschedpolicy( &audio_attr, SCHED_RR ) == 0 ) {
   sched_param params;
   params.sched_priority = 50;
   pthread_attr_setschedparam(&audio_attr,&params);
     } else {
   qDebug( "Error setting up a realtime thread, reverting to using a normal thread." );
   pthread_attr_destroy(&audio_attr);
   pthread_attr_init(&audio_attr);
     }
 #endif
 //qDebug("create audio thread");
     pthread_create(&audio_tid, &audio_attr, (void * (*)(void *))startAudioThread, this);
 }
 
 
 LoopControl::~LoopControl() {
     stop();
 }
 
 
 static long prev_frame = 0;
 static int currentSample = 0;
 
 
 void LoopControl::timerEvent( QTimerEvent *te ) {
 
     if ( te->timerId() == videoId )
   startVideo();
 
     if ( te->timerId() == sliderId ) {
   if ( hasAudioChannel && !hasVideoChannel && moreAudio ) {
       mediaPlayerState->updatePosition( audioSampleCounter );
   } else if ( hasVideoChannel && moreVideo ) {
       mediaPlayerState->updatePosition( current_frame );
   }
     }
 
     if ( !moreVideo && !moreAudio ) { 
         mediaPlayerState->setPlaying( FALSE );
         mediaPlayerState->setNext();
     }
 }
 
 
 void LoopControl::setPosition( long pos ) {
     audioMutex->lock();
 
     if ( hasVideoChannel && hasAudioChannel ) {
   playtime.restart();
   playtime = playtime.addMSecs( long((double)-pos * 1000.0 / framerate) );
   current_frame = pos + 1;
   mediaPlayerState->curDecoder()->videoSetFrame( current_frame, stream );
   prev_frame = current_frame - 1;
   currentSample = (int)( (double)current_frame * freq / framerate );
   mediaPlayerState->curDecoder()->audioSetSample( currentSample, stream );
   audioSampleCounter = currentSample - 1;
     } else if ( hasVideoChannel ) {
   playtime.restart();
   playtime = playtime.addMSecs( long((double)-pos * 1000.0 / framerate) );
   current_frame = pos + 1;
   mediaPlayerState->curDecoder()->videoSetFrame( current_frame, stream );
   prev_frame = current_frame - 1;
     } else if ( hasAudioChannel ) {
   playtime.restart();
   playtime = playtime.addMSecs( long((double)-pos * 1000.0 / freq) );
   currentSample = pos + 1;
   mediaPlayerState->curDecoder()->audioSetSample( currentSample, stream );
   audioSampleCounter = currentSample - 1;
     }
 
     audioMutex->unlock();
 }
 
 
 void LoopControl::startVideo() {
 
     if ( moreVideo ) {
 
         if ( mediaPlayerState->curDecoder() ) {
 
       if ( hasAudioChannel && !isMuted ) {
 
     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;
   threadOkToGo = TRUE;
     }
 
     sliderId = startTimer( 300 ); // update slider every 1/3 second
 
     audioMutex->unlock();
 }
 
 
 void LoopControl::setPaused( bool pause ) {
 
     if ( !mediaPlayerState->curDecoder() || !mediaPlayerState->curDecoder()->isOpen() )
   return;
 
     if ( pause ) {
   killTimers();
     } else {
   // Force an update of the position
   mediaPlayerState->setPosition( mediaPlayerState->position() + 1 );
   mediaPlayerState->setPosition( mediaPlayerState->position() - 1 );
   // Just like we never stopped
   startTimers();
     }
 }
 
 
 void LoopControl::stop( bool willPlayAgainShortly ) {
 
 #if defined(Q_WS_QWS) && !defined(QT_NO_COP)
     if ( !willPlayAgainShortly && disabledSuspendScreenSaver ) {
   disabledSuspendScreenSaver = FALSE; 
   // Re-enable the suspend mode
   QCopEnvelope("QPE/System", "setScreenSaverMode(int)" ) << QPEApplication::Enable;
     }
 #endif
 
     if ( mediaPlayerState->curDecoder() && mediaPlayerState->curDecoder()->isOpen() ) {
 
   killTimers();
 
   audioMutex->lock();
 
   mediaPlayerState->curDecoder()->close();
 
   if ( audioDevice ) {
       delete audioDevice;
       delete audioBuffer;
       audioDevice = 0;
       audioBuffer = 0;
   }
 
   audioMutex->unlock();
 
     }
 }
 
 
 bool LoopControl::init( const QString& filename ) {
     stop();
 
     audioMutex->lock();
 
     fileName = filename;
     stream = 0; // only play stream 0 for now
     current_frame = total_video_frames = total_audio_samples = 0;
     
     qDebug( "Using the %s decoder", mediaPlayerState->curDecoder()->pluginName() );
    
     // ### Hack to use libmpeg3plugin to get the number of audio samples if we are using the libmad plugin
     if ( mediaPlayerState->curDecoder()->pluginName() == QString("LibMadPlugin") ) {
         if ( mediaPlayerState->libMpeg3Decoder() && mediaPlayerState->libMpeg3Decoder()->open( filename )) {
             total_audio_samples = mediaPlayerState->libMpeg3Decoder()->audioSamples( 0 );
             mediaPlayerState->libMpeg3Decoder()->close();
         }
     }
     
     if ( !mediaPlayerState->curDecoder()|| !mediaPlayerState->curDecoder()->open( filename ) ) {
   audioMutex->unlock();
   return FALSE;
     }
 
     hasAudioChannel = mediaPlayerState->curDecoder()->audioStreams() > 0;
     hasVideoChannel = mediaPlayerState->curDecoder()->videoStreams() > 0;
 
     if ( hasAudioChannel ) {
   int astream = 0;
 
     if ( mediaPlayerState->curDecoder()->pluginName() == QString("LibMpeg3Plugin") )
         channels = 2; //dont akx me why, but it needs this hack
     else
         channels = mediaPlayerState->curDecoder()->audioChannels( astream );
 
     qDebug( "LC- channels = %d", channels );
   
   if ( !total_audio_samples )
       total_audio_samples = mediaPlayerState->curDecoder()->audioSamples( astream );
 
   total_audio_samples += 1000;
 
   mediaPlayerState->setLength( total_audio_samples );
   
   freq = mediaPlayerState->curDecoder()->audioFrequency( astream );
   qDebug( "LC- frequency = %d", freq );
 
   audioSampleCounter = 0;
    int bits_per_sample;
    if ( mediaPlayerState->curDecoder()->pluginName() == QString("LibWavPlugin") ) {
        bits_per_sample =(int) mediaPlayerState->curDecoder()->getTime();
        qDebug("using stupid hack");
    } else {
            bits_per_sample=0;
    }
 
   audioDevice = new AudioDevice( freq, channels, bits_per_sample);
   audioBuffer = new char[ audioDevice->bufferSize() ];
   channels = audioDevice->channels();
 
   //### must check which frequency is actually used.
   static const int size = 1;
   short int buf[size];
   long samplesRead = 0;
   mediaPlayerState->curDecoder()->audioReadSamples( buf, channels, size, samplesRead, stream );
     }
 
     if ( hasVideoChannel ) {
   total_video_frames = mediaPlayerState->curDecoder()->videoFrames( stream );
 
   mediaPlayerState->setLength( total_video_frames );
 
   framerate = mediaPlayerState->curDecoder()->videoFrameRate( stream );
   DecodeLoopDebug(( "Frame rate %g total %ld", framerate, total_video_frames ));
 
   if ( framerate <= 1.0 ) {
       DecodeLoopDebug(( "Crazy frame rate, resetting to sensible" ));
       framerate = 25;
   }
 
   if ( total_video_frames == 1 ) {
       DecodeLoopDebug(( "Cannot seek to frame" ));
   }
 
     }
 
     current_frame = 0;