mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Improved audio skip, after channel switch.
This commit is contained in:
Johns
parent
57af986367
commit
0f62a521f4
57
audio.c
57
audio.c
@ -178,6 +178,7 @@ static int AudioStereoDescent; ///< volume descent for stereo
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static int AudioVolume; ///< current volume (0 .. 1000)
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extern int VideoAudioDelay; ///< import audio/video delay
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extern int VideoGetBuffers(void); ///< Get number of input buffers.
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/// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
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static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
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@ -623,6 +624,7 @@ typedef struct _audio_ring_ring_
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{
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char FlushBuffers; ///< flag: flush buffers
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char UseAc3; ///< flag: use ac3 pass-through
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int16_t PacketSize; ///< packet size
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unsigned HwSampleRate; ///< hardware sample rate in Hz
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unsigned HwChannels; ///< hardware number of channels
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unsigned InSampleRate; ///< input sample rate in Hz
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@ -680,6 +682,7 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
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// FIXME: don't flush buffers here
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AudioRing[AudioRingWrite].FlushBuffers = 1;
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AudioRing[AudioRingWrite].UseAc3 = use_ac3;
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AudioRing[AudioRingWrite].PacketSize = 0;
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AudioRing[AudioRingWrite].InSampleRate = sample_rate;
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AudioRing[AudioRingWrite].InChannels = channels;
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AudioRing[AudioRingWrite].HwSampleRate = sample_rate;
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@ -3589,6 +3592,11 @@ void AudioEnqueue(const void *samples, int count)
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Debug(3, "audio: enqueue not ready\n");
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return; // no setup yet
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}
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// save packet size
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if (!AudioRing[AudioRingWrite].PacketSize) {
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AudioRing[AudioRingWrite].PacketSize = count;
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Debug(3, "audio: a/v packet size %d bytes\n", count);
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}
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if (AudioRing[AudioRingWrite].UseAc3) {
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buffer = (void *)samples;
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} else {
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@ -3633,6 +3641,15 @@ void AudioEnqueue(const void *samples, int count)
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n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
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skip = AudioSkip;
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// FIXME: round to packet size
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Debug(3, "audio: start? %4zdms skip %dms\n", (n * 1000)
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/ (AudioRing[AudioRingWrite].HwSampleRate *
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AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
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(skip * 1000)
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/ (AudioRing[AudioRingWrite].HwSampleRate *
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AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
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if (skip) {
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if (n < (unsigned)skip) {
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skip = n;
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@ -3641,12 +3658,8 @@ void AudioEnqueue(const void *samples, int count)
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RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
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n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
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}
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Debug(3, "audio: start? %4zdms\n", (n * 1000)
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/ (AudioRing[AudioRingWrite].HwSampleRate *
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AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
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// forced start or enough video + audio buffered
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// for some exotic channels * 4 too small
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if (AudioStartThreshold * 4 < n || (AudioVideoIsReady
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&& AudioStartThreshold < n)) {
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// restart play-back
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@ -3703,6 +3716,7 @@ void AudioVideoReady(int64_t pts)
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size_t used;
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if (pts == (int64_t) INT64_C(0x8000000000000000)) {
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Debug(3, "audio: a/v start, no valid video\n");
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return;
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}
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// no valid audio known
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@ -3725,18 +3739,24 @@ void AudioVideoReady(int64_t pts)
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Debug(3, "audio: a/v buf:%4zdms %s|%s = %dms video ready\n",
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(used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
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AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
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Timestamp2String(pts), Timestamp2String(audio_pts),
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Timestamp2String(audio_pts), Timestamp2String(pts),
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(int)(pts - audio_pts) / 90);
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if (!AudioRunning) {
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int skip;
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// keep ~5 frames
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skip = pts - 5 * 20 * 90 - audio_pts - VideoAudioDelay;
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if (skip > 0) {
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// buffer ~15 video frames
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// FIXME: HDTV can use smaller video buffer
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skip =
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pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts +
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VideoAudioDelay;
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printf("%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
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VideoAudioDelay / 90, skip / 90);
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if (1 && skip > 0) {
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skip = (((int64_t) skip * AudioRing[AudioRingWrite].HwSampleRate)
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/ (1000 * 90))
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* AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample;
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// FIXME: round to packet size
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if ((unsigned)skip > used) {
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AudioSkip = skip - used;
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skip = used;
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@ -3748,13 +3768,15 @@ void AudioVideoReady(int64_t pts)
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RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
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used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
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}
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// FIXME: skip<0 we need bigger audio buffer
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// enough video + audio buffered
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if (AudioStartThreshold < used) {
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AudioRunning = 1;
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pthread_cond_signal(&AudioStartCond);
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}
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}
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}
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AudioVideoIsReady = 1;
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#if 0
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@ -3811,6 +3833,7 @@ void AudioFlushBuffers(void)
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{
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#ifdef USE_AUDIORING
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int old;
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int i;
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old = AudioRingWrite;
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AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
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@ -3829,10 +3852,18 @@ void AudioFlushBuffers(void)
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atomic_inc(&AudioRingFilled);
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// FIXME: wait for flush complete?
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for (i = 0; i < 24 * 2; ++i) {
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if (!AudioRunning) { // wakeup thread to flush buffers
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AudioRunning = 1;
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pthread_cond_signal(&AudioStartCond);
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}
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if (!atomic_read(&AudioRingFilled)) {
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break;
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}
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usleep(1 * 1000); // avoid hot polling
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}
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Debug(3, "audio: audio flush %dms\n", i);
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#else
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AudioUsedModule->FlushBuffers();
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#endif
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@ -3947,7 +3978,10 @@ int64_t AudioGetClock(void)
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// delay zero, if no valid time stamp
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if ((delay = AudioGetDelay())) {
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return AudioRing[AudioRingRead].PTS - delay;
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if (AudioRing[AudioRingRead].UseAc3) {
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return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
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}
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return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
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}
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}
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return INT64_C(0x8000000000000000);
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@ -3986,7 +4020,6 @@ void AudioSetVolume(int volume)
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}
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#endif
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AudioAmplifier = volume;
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printf("volume %d\n", volume);
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if (!AudioSoftVolume) {
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AudioUsedModule->SetVolume(volume);
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}
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10
video.c
10
video.c
@ -349,7 +349,7 @@ static VideoZoomModes Video4to3ZoomMode;
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static char Video60HzMode; ///< handle 60hz displays
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static char VideoSoftStartSync; ///< soft start sync audio/video
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static const int VideoSoftStartFrames = 120; ///< soft start frames
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static const int VideoSoftStartFrames = 100; ///< soft start frames
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static char VideoShowBlackPicture; ///< flag show black picture
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static xcb_atom_t WmDeleteWindowAtom; ///< WM delete message atom
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@ -7943,21 +7943,21 @@ static void VdpauSyncDecoder(VdpauDecoder * decoder)
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// both clocks are known
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if (abs(video_clock - audio_clock + VideoAudioDelay) > 5000 * 90) {
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err = VdpauMessage(3, "video: audio/video difference too big\n");
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err = VdpauMessage(2, "video: audio/video difference too big\n");
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} else if (video_clock > audio_clock + VideoAudioDelay + 100 * 90) {
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// FIXME: this quicker sync step, did not work with new code!
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err = VdpauMessage(3, "video: slow down video, duping frame\n");
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err = VdpauMessage(2, "video: slow down video, duping frame\n");
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++decoder->FramesDuped;
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decoder->SyncCounter = 1;
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goto out;
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} else if (video_clock > audio_clock + VideoAudioDelay + 45 * 90) {
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err = VdpauMessage(3, "video: slow down video, duping frame\n");
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err = VdpauMessage(2, "video: slow down video, duping frame\n");
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++decoder->FramesDuped;
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decoder->SyncCounter = 1;
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goto out;
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} else if (audio_clock + VideoAudioDelay > video_clock + 15 * 90
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&& filled > 1 + 2 * decoder->Interlaced) {
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err = VdpauMessage(3, "video: speed up video, droping frame\n");
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err = VdpauMessage(2, "video: speed up video, droping frame\n");
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++decoder->FramesDropped;
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VdpauAdvanceDecoderFrame(decoder);
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decoder->SyncCounter = 1;
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