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-rw-r--r--core/multimedia/opieplayer/audiodevice.cpp8
-rw-r--r--core/multimedia/opieplayer/loopcontrol.cpp14
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;