mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Improved audio drift correction support.
This commit is contained in:
parent
144f22314f
commit
43b48224b5
@ -1,6 +1,7 @@
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User johns
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Date:
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Improved audio drift correction support.
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Experimental audio drift correction support.
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Add SVDRP HOTK command support.
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Increased audio buffer time for PES packets.
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48
audio.c
48
audio.c
@ -140,8 +140,6 @@ static volatile char AudioPaused; ///< audio paused
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static unsigned AudioSampleRate; ///< audio sample rate in hz
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static unsigned AudioChannels; ///< number of audio channels
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static const int AudioBytesProSample = 2; ///< number of bytes per sample
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static uint32_t AudioTicks; ///< audio ticks
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static uint32_t AudioTickPTS; ///< audio pts of tick
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static int64_t AudioPTS; ///< audio pts clock
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static int AudioBufferTime = 336; ///< audio buffer time in ms
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@ -294,7 +292,7 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
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}
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if (!AudioRunning) {
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Debug(3, "audio/alsa: start %zd ms %d\n",
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Debug(3, "audio/alsa: start %4zd ms %d v-buf\n",
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(RingBufferUsedBytes(AlsaRingBuffer) * 1000)
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/ (AudioSampleRate * AudioChannels * AudioBytesProSample),
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VideoGetBuffers());
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@ -333,7 +331,8 @@ static int AlsaPlayRingbuffer(void)
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if (n == -EAGAIN) {
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continue;
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}
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Error(_("audio/alsa: underrun error?\n"));
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Error(_("audio/alsa: avail underrun error? '%s'\n"),
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snd_strerror(n));
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err = snd_pcm_recover(AlsaPCMHandle, n, 0);
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if (err >= 0) {
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continue;
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@ -401,7 +400,8 @@ static int AlsaPlayRingbuffer(void)
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goto again;
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}
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*/
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Error(_("audio/alsa: underrun error?\n"));
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Error(_("audio/alsa: writei underrun error? '%s'\n"),
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snd_strerror(err));
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err = snd_pcm_recover(AlsaPCMHandle, err, 0);
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if (err >= 0) {
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goto again;
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@ -433,7 +433,7 @@ static void AlsaFlushBuffers(void)
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RingBufferReadAdvance(AlsaRingBuffer,
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RingBufferUsedBytes(AlsaRingBuffer));
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state = snd_pcm_state(AlsaPCMHandle);
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Debug(3, "audio/alsa: state %d - %s\n", state,
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Debug(3, "audio/alsa: flush state %d - %s\n", state,
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snd_pcm_state_name(state));
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if (state != SND_PCM_STATE_OPEN) {
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if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
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@ -446,7 +446,6 @@ static void AlsaFlushBuffers(void)
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}
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}
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AudioRunning = 0;
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AudioTicks = 0;
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AudioPTS = INT64_C(0x8000000000000000);
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}
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@ -651,14 +650,15 @@ static void AlsaThread(void)
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break;
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}
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// wait for space in kernel buffers
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if ((err = snd_pcm_wait(AlsaPCMHandle, 100)) < 0) {
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Error(_("audio/alsa: wait underrun error?\n"));
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if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) {
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Error(_("audio/alsa: wait underrun error? '%s'\n"),
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snd_strerror(err));
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err = snd_pcm_recover(AlsaPCMHandle, err, 0);
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if (err >= 0) {
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continue;
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}
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Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
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usleep(100 * 1000);
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usleep(24 * 1000);
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continue;
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}
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if (AlsaFlushBuffer || AudioPaused) {
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@ -888,6 +888,9 @@ static uint64_t AlsaGetDelay(void)
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if (!AlsaPCMHandle || !AudioSampleRate) {
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return 0UL;
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}
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if (!AudioRunning) { // audio not running
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return 0UL;
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}
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// FIXME: thread safe? __assert_fail_base in snd_pcm_delay
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// delay in frames in alsa + kernel buffers
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@ -1303,7 +1306,7 @@ static int OssAddToRingbuffer(const void *samples, int count)
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}
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if (!AudioRunning) {
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Debug(3, "audio/oss: start %zd ms %d\n",
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Debug(3, "audio/oss: start %4zd ms %d v-buf\n",
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(RingBufferUsedBytes(OssRingBuffer) * 1000)
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/ (AudioSampleRate * AudioChannels * AudioBytesProSample),
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VideoGetBuffers());
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@ -1387,7 +1390,6 @@ static void OssFlushBuffers(void)
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}
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}
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AudioRunning = 0;
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AudioTicks = 0;
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AudioPTS = INT64_C(0x8000000000000000);
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}
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@ -1672,8 +1674,7 @@ static uint64_t OssGetDelay(void)
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if (OssPcmFildes == -1) { // setup failure
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return 0UL;
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}
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if (!AudioRunning) {
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if (!AudioRunning) { // audio not running
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return 0UL;
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}
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// delay in bytes in kernel buffers
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@ -2185,25 +2186,6 @@ int64_t AudioGetClock(void)
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int64_t delay;
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if ((delay = AudioGetDelay())) {
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#if 0
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int64_t pts;
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uint32_t ticks;
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pts = AudioPTS - delay;
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ticks = GetMsTicks();
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if (AudioTicks) {
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static int64_t drift;
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drift += ((pts - AudioTickPTS) - (ticks - AudioTicks) * 90);
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drift /= 2;
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if (abs(drift) > 90) {
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printf("audio-drift: %d\n", (int)drift);
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}
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}
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AudioTicks = ticks;
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AudioTickPTS = pts;
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return pts;
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#endif
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return AudioPTS - delay;
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}
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}
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112
codec.c
112
codec.c
@ -609,13 +609,12 @@ struct _audio_decoder_
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ReSampleContext *ReSample; ///< audio resampling context
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int64_t StartPTS; ///< start PTS
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struct timespec StartTime; ///< start time
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int64_t LastDelay; ///< last delay
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struct timespec LastTime; ///< last time
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int64_t LastPTS; ///< last PTS
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int Drift; ///< drift correction value
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#define AVERAGE 10 ///< number of average values
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int Average[AVERAGE]; ///< average for drift calculation
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int Drift; ///< accumulated audio drift
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int DriftCorr; ///< audio drift correction value
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struct AVResampleContext *AvResample; ///< second audio resample context
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#define MAX_CHANNELS 8 ///< max number of channels supported
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@ -728,7 +727,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
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audio_decoder->Channels = 0;
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audio_decoder->HwSampleRate = 0;
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audio_decoder->HwChannels = 0;
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audio_decoder->StartPTS = AV_NOPTS_VALUE;
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audio_decoder->LastDelay = 0;
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}
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/**
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@ -851,56 +850,75 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
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static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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{
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struct timespec nowtime;
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int64_t delay;
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int64_t tim_diff;
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int64_t pts_diff;
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int64_t drift;
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int corr;
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AudioSetClock(pts);
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// start drift detection
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if (audio_decoder->StartPTS == (int64_t) AV_NOPTS_VALUE && AudioGetDelay()) {
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audio_decoder->StartPTS = AudioGetClock();
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audio_decoder->LastPTS = audio_decoder->StartPTS;
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clock_gettime(CLOCK_REALTIME, &audio_decoder->StartTime);
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delay = AudioGetDelay();
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if (!delay) {
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return;
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}
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pts = AudioGetClock();
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clock_gettime(CLOCK_REALTIME, &nowtime);
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pts_diff = pts - audio_decoder->StartPTS;
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tim_diff = (nowtime.tv_sec - audio_decoder->StartTime.tv_sec)
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* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
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audio_decoder->StartTime.tv_nsec);
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drift = pts_diff * 1000 * 1000 / 90 - tim_diff;
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if (abs(drift) > 100 * 1000 * 1000) {
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// drift too big, pts changed?
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audio_decoder->StartPTS = pts;
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audio_decoder->LastPTS = audio_decoder->StartPTS;
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audio_decoder->StartTime = nowtime;
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if (!audio_decoder->LastDelay) {
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audio_decoder->LastTime = nowtime;
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audio_decoder->LastPTS = pts;
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audio_decoder->LastDelay = delay;
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audio_decoder->Drift = 0;
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Debug(3, "codec/audio: inital delay %zd ms\n", delay / 90);
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return;
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}
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// collect over some time
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if (pts - audio_decoder->LastPTS < 10 * 1000 * 90) {
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pts_diff = pts - audio_decoder->LastPTS;
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if (pts_diff < 10 * 1000 * 90) {
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return;
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}
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tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
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* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
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audio_decoder->LastTime.tv_nsec);
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drift =
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(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
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audio_decoder->LastDelay;
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audio_decoder->LastTime = nowtime;
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audio_decoder->LastPTS = pts;
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audio_decoder->LastDelay = delay;
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audio_decoder->Drift +=
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(int)((10 * audio_decoder->SampleRate * drift) / tim_diff);
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if (audio_decoder->AvResample) {
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av_resample_compensate(audio_decoder->AvResample, audio_decoder->Drift,
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10 * audio_decoder->SampleRate);
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if (1) {
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Debug(3, "codec/audio: interval P:%5zdms T:%5zdms D:%4zdms %f %d\n",
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pts_diff / 90, tim_diff / (1000 * 1000), delay / 90, drift / 90.0,
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audio_decoder->DriftCorr);
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}
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Info("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %g\n",
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audio_decoder->Drift, drift / (1000 * 1000), drift,
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(double)drift / tim_diff);
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printf("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %d\n",
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audio_decoder->Drift, drift / (1000 * 1000), drift,
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(int)((10 * audio_decoder->SampleRate * drift) / tim_diff));
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if (abs(drift) > 5 * 90) {
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// drift too big, pts changed?
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Debug(3, "codec/audio: drift(%5d) %3" PRId64 "ms reset\n",
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audio_decoder->DriftCorr, drift);
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audio_decoder->LastDelay = 0;
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return;
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}
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drift += audio_decoder->Drift;
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audio_decoder->Drift = drift;
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corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
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audio_decoder->DriftCorr -= corr;
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if (audio_decoder->DriftCorr < -20000) { // limit correction
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audio_decoder->DriftCorr = -20000;
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} else if (audio_decoder->DriftCorr > 20000) {
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audio_decoder->DriftCorr = 20000;
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}
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if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
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av_resample_compensate(audio_decoder->AvResample,
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audio_decoder->DriftCorr / 10, 10 * audio_decoder->HwSampleRate);
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}
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printf("codec/audio: drift(%5d) %8" PRId64 "us %4d\n",
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audio_decoder->DriftCorr, drift * 1000 / 90, corr);
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}
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/**
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@ -977,6 +995,12 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
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if (!audio_decoder->AvResample) {
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Error(_("codec/audio: AvResample setup error\n"));
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} else {
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// reset drift to some default value
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audio_decoder->DriftCorr /= 2;
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av_resample_compensate(audio_decoder->AvResample,
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audio_decoder->DriftCorr / 10,
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10 * audio_decoder->HwSampleRate);
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}
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}
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}
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@ -1053,18 +1077,6 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
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}
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n *= 2;
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#if 0
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// FIXME: must split channels, filter, join channels
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n = av_resample(audio_decoder->AvResample, buf, data, &consumed, count,
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sizeof(buf), 1);
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if (n < 0) {
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Error(_("codec/audio: can't av_resample\n"));
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return;
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}
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if (consumed != count) {
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Error(_("codec/audio: av_resample didn't consume all samples\n"));
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}
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#endif
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n *= audio_decoder->HwChannels;
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CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
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AudioEnqueue(buf, n);
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