mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 17:16:51 +00:00
Experimental audio drift correction support.
This commit is contained in:
Johns
336
codec.c
336
codec.c
@@ -30,8 +30,10 @@
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/// many bugs and incompatiblity in it. Don't use this shit.
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///
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/// compile with passthrough support (experimental)
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/// compile with passthrough support (stable, ac3 only)
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#define USE_PASSTHROUGH
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/// compile audio drift correction support (experimental)
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#define noUSE_AUDIO_DRIFT_CORRECTION
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#include <stdio.h>
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#include <unistd.h>
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@@ -606,6 +608,22 @@ struct _audio_decoder_
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int HwChannels; ///< hw channels
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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 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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struct AVResampleContext *AvResample; ///< second audio resample context
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#define MAX_CHANNELS 8 ///< max number of channels supported
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int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers
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int BufferSize; ///< size of sample buffer
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int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples
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int RemainSize; ///< size of remain buffer
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int RemainCount; ///< number of remaining samples
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};
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#ifdef USE_PASSTHROUGH
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@@ -710,6 +728,7 @@ 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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}
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/**
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@@ -720,6 +739,21 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
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void CodecAudioClose(AudioDecoder * audio_decoder)
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{
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// FIXME: output any buffered data
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if (audio_decoder->AvResample) {
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int ch;
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av_resample_close(audio_decoder->AvResample);
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audio_decoder->AvResample = NULL;
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audio_decoder->RemainCount = 0;
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audio_decoder->BufferSize = 0;
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audio_decoder->RemainSize = 0;
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for (ch = 0; ch < MAX_CHANNELS; ++ch) {
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free(audio_decoder->Buffer[ch]);
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audio_decoder->Buffer[ch] = NULL;
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free(audio_decoder->Remain[ch]);
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audio_decoder->Remain[ch] = NULL;
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}
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}
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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audio_decoder->ReSample = NULL;
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@@ -809,6 +843,238 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
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}
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}
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/**
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** Set/update audio pts clock.
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**
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** @param audio_decoder audio decoder data
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*/
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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 tim_diff;
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int64_t pts_diff;
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int64_t drift;
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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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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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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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return;
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}
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audio_decoder->LastPTS = pts;
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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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}
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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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}
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/**
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** Handle audio format changes.
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**
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** @param audio_decoder audio decoder data
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*/
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static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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{
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const AVCodecContext *audio_ctx;
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int err;
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int isAC3;
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audio_ctx = audio_decoder->AudioCtx;
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audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
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// FIXME: use swr_convert from swresample (only in ffmpeg!)
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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audio_decoder->ReSample = NULL;
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}
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if (audio_decoder->AvResample) {
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av_resample_close(audio_decoder->AvResample);
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audio_decoder->AvResample = NULL;
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audio_decoder->RemainCount = 0;
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}
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audio_decoder->SampleRate = audio_ctx->sample_rate;
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audio_decoder->HwSampleRate = audio_ctx->sample_rate;
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audio_decoder->Channels = audio_ctx->channels;
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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audio_decoder->HwChannels = 2;
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isAC3 = 1;
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} else {
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audio_decoder->HwChannels = audio_ctx->channels;
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isAC3 = 0;
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}
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// channels not support?
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if ((err =
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AudioSetup(&audio_decoder->HwSampleRate,
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&audio_decoder->HwChannels, isAC3))) {
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Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
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audio_ctx->sample_rate, audio_ctx->channels,
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audio_decoder->HwSampleRate, audio_decoder->HwChannels);
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if (err == 1) {
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audio_decoder->ReSample =
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av_audio_resample_init(audio_decoder->HwChannels,
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audio_ctx->channels, audio_decoder->HwSampleRate,
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audio_ctx->sample_rate, audio_ctx->sample_fmt,
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audio_ctx->sample_fmt, 16, 10, 0, 0.8);
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// libav-0.8_pre didn't support 6 -> 2 channels
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if (!audio_decoder->ReSample) {
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Error(_("codec/audio: resample setup error\n"));
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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return;
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}
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} else {
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Debug(3, "codec/audio: audio setup error\n");
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// FIXME: handle errors
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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return;
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}
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}
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// prepare audio drift resample
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if (!isAC3) {
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audio_decoder->AvResample =
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av_resample_init(audio_decoder->HwSampleRate,
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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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}
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}
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}
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/**
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** Codec enqueue audio samples.
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**
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** @param audio_decoder audio decoder data
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** @param data samples data
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** @param count number of samples
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**
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*/
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void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
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{
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#ifdef USE_AUDIO_DRIFT_CORRECTION
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if (audio_decoder->AvResample) {
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int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
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int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
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int consumed;
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int i;
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int n;
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int ch;
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int bytes_n;
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bytes_n = count / audio_decoder->HwChannels;
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// resize sample buffer, if needed
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if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) {
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audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n;
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for (ch = 0; ch < MAX_CHANNELS; ++ch) {
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audio_decoder->Buffer[ch] =
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realloc(audio_decoder->Buffer[ch],
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audio_decoder->BufferSize);
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}
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}
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// copy remaining bytes into sample buffer
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for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
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memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch],
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audio_decoder->RemainCount);
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}
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// deinterleave samples into sample buffer
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for (i = 0; i < bytes_n / 2; i++) {
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for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
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audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i]
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= data[i * audio_decoder->HwChannels + ch];
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}
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}
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bytes_n += audio_decoder->RemainSize;
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n = 0; // keep gcc lucky
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// resample the sample buffer into tmp buffer
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for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
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n = av_resample(audio_decoder->AvResample, buftmp[ch],
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audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
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sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
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//printf("remain%d: %d = %d/%d\n", ch, n, consumed, bytes_n /2);
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// fixme remaining channels
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if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
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audio_decoder->RemainSize = bytes_n - consumed * 2;
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}
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audio_decoder->Remain[ch] =
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realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize);
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memcpy(audio_decoder->Remain[ch],
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audio_decoder->Buffer[ch] + consumed,
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audio_decoder->RemainSize);
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audio_decoder->RemainCount = audio_decoder->RemainSize;
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}
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// interleave samples from sample buffer
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for (i = 0; i < n; i++) {
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for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
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buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i];
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}
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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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return;
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}
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#endif
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CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
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AudioEnqueue(data, count);
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}
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/**
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** Decode an audio packet.
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**
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@@ -847,64 +1113,17 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
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#else
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#endif
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// Update audio clock
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// update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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AudioSetClock(avpkt->pts);
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CodecAudioSetClock(audio_decoder, avpkt->pts);
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}
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// FIXME: must first play remainings bytes, than change and play new.
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if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
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|| audio_decoder->SampleRate != audio_ctx->sample_rate
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|| audio_decoder->Channels != audio_ctx->channels) {
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int err;
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int isAC3;
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audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
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// FIXME: use swr_convert from swresample (only in ffmpeg!)
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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audio_decoder->ReSample = NULL;
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}
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audio_decoder->SampleRate = audio_ctx->sample_rate;
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audio_decoder->HwSampleRate = audio_ctx->sample_rate;
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audio_decoder->Channels = audio_ctx->channels;
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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audio_decoder->HwChannels = 2;
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isAC3 = 1;
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} else {
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audio_decoder->HwChannels = audio_ctx->channels;
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isAC3 = 0;
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}
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// channels not support?
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if ((err =
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AudioSetup(&audio_decoder->HwSampleRate,
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&audio_decoder->HwChannels, isAC3))) {
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Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
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audio_ctx->sample_rate, audio_ctx->channels,
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audio_decoder->HwSampleRate, audio_decoder->HwChannels);
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if (err == 1) {
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audio_decoder->ReSample =
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av_audio_resample_init(audio_decoder->HwChannels,
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audio_ctx->channels, audio_decoder->HwSampleRate,
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audio_ctx->sample_rate, audio_ctx->sample_fmt,
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audio_ctx->sample_fmt, 16, 10, 0, 0.8);
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// libav-0.8_pre didn't support 6 -> 2 channels
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if (!audio_decoder->ReSample) {
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Error(_("codec/audio: resample setup error\n"));
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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}
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} else {
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Debug(3, "codec/audio: audio setup error\n");
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// FIXME: handle errors
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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return;
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}
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}
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CodecAudioUpdateFormat(audio_decoder);
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}
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if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
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@@ -931,9 +1150,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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audio_decoder->HwChannels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
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CodecReorderAudioFrame(outbuf, outlen,
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audio_decoder->HwChannels);
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AudioEnqueue(outbuf, outlen);
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CodecAudioEnqueue(audio_decoder, outbuf, outlen);
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}
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} else {
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#ifdef USE_PASSTHROUGH
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@@ -955,6 +1172,10 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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buf[3] = htole16(avpkt->size * 8);
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swab(avpkt->data, buf + 4, avpkt->size);
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memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
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// don't play with the ac-3 samples
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AudioEnqueue(buf, buf_sz);
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// FIXME: handle drift.
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return;
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}
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#if 0
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//
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@@ -1008,8 +1229,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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// True HD?
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#endif
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#endif
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CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
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AudioEnqueue(buf, buf_sz);
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CodecAudioEnqueue(audio_decoder, buf, buf_sz);
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}
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}
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}
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