mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Experimental audio drift correction support.
This commit is contained in:
Johns
parent
51eb720265
commit
144f22314f
@ -1,6 +1,7 @@
|
||||
User johns
|
||||
Date:
|
||||
|
||||
Experimental audio drift correction support.
|
||||
Add SVDRP HOTK command support.
|
||||
Increased audio buffer time for PES packets.
|
||||
Support configuration and set of video background.
|
||||
|
||||
4
Makefile
4
Makefile
@ -19,9 +19,11 @@ GIT_REV = $(shell git describe --always 2>/dev/null)
|
||||
### Configuration (edit this for your needs)
|
||||
|
||||
CONFIG := #-DDEBUG
|
||||
CONFIG += -DUSE_AUDIO_DRIFT_CORRECTION # build new audio drift code
|
||||
CONFIG += -DAV_INFO -DAV_INFO_TIME=3000 # debug a/v sync
|
||||
#CONFIG += -DHAVE_PTHREAD_NAME # supports new pthread_setname_np
|
||||
#CONFIG += -DUSE_TS_AUDIO # build new ts audio parser
|
||||
CONFIG += -DUSE_TS_AUDIO # build new ts audio parser
|
||||
#CONFIG += -DUSE_TS_VIDEO # build new ts video parser
|
||||
CONFIG += $(shell pkg-config --exists vdpau && echo "-DUSE_VDPAU")
|
||||
CONFIG += $(shell pkg-config --exists libva && echo "-DUSE_VAAPI")
|
||||
CONFIG += $(shell pkg-config --exists alsa && echo "-DUSE_ALSA")
|
||||
|
||||
2
Todo
2
Todo
@ -85,6 +85,8 @@ audio:
|
||||
Mute should do a real mute and not only set volume to zero.
|
||||
Starting suspended and muted, didn't register the mute.
|
||||
Relaxed audio sync checks at end of packet and already in sync
|
||||
samplerate problem resume/suspend.
|
||||
only wait for video buffers, if video is running.
|
||||
|
||||
audio/alsa:
|
||||
better downmix of >2 channels on 2 channel hardware
|
||||
|
||||
336
codec.c
336
codec.c
@ -30,8 +30,10 @@
|
||||
/// many bugs and incompatiblity in it. Don't use this shit.
|
||||
///
|
||||
|
||||
/// compile with passthrough support (experimental)
|
||||
/// compile with passthrough support (stable, ac3 only)
|
||||
#define USE_PASSTHROUGH
|
||||
/// compile audio drift correction support (experimental)
|
||||
#define noUSE_AUDIO_DRIFT_CORRECTION
|
||||
|
||||
#include <stdio.h>
|
||||
#include <unistd.h>
|
||||
@ -606,6 +608,22 @@ struct _audio_decoder_
|
||||
int HwChannels; ///< hw channels
|
||||
|
||||
ReSampleContext *ReSample; ///< audio resampling context
|
||||
|
||||
int64_t StartPTS; ///< start PTS
|
||||
struct timespec StartTime; ///< start time
|
||||
int64_t LastPTS; ///< last PTS
|
||||
|
||||
int Drift; ///< drift correction value
|
||||
#define AVERAGE 10 ///< number of average values
|
||||
int Average[AVERAGE]; ///< average for drift calculation
|
||||
|
||||
struct AVResampleContext *AvResample; ///< second audio resample context
|
||||
#define MAX_CHANNELS 8 ///< max number of channels supported
|
||||
int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers
|
||||
int BufferSize; ///< size of sample buffer
|
||||
int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples
|
||||
int RemainSize; ///< size of remain buffer
|
||||
int RemainCount; ///< number of remaining samples
|
||||
};
|
||||
|
||||
#ifdef USE_PASSTHROUGH
|
||||
@ -710,6 +728,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
|
||||
audio_decoder->Channels = 0;
|
||||
audio_decoder->HwSampleRate = 0;
|
||||
audio_decoder->HwChannels = 0;
|
||||
audio_decoder->StartPTS = AV_NOPTS_VALUE;
|
||||
}
|
||||
|
||||
/**
|
||||
@ -720,6 +739,21 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
|
||||
void CodecAudioClose(AudioDecoder * audio_decoder)
|
||||
{
|
||||
// FIXME: output any buffered data
|
||||
if (audio_decoder->AvResample) {
|
||||
int ch;
|
||||
|
||||
av_resample_close(audio_decoder->AvResample);
|
||||
audio_decoder->AvResample = NULL;
|
||||
audio_decoder->RemainCount = 0;
|
||||
audio_decoder->BufferSize = 0;
|
||||
audio_decoder->RemainSize = 0;
|
||||
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
|
||||
free(audio_decoder->Buffer[ch]);
|
||||
audio_decoder->Buffer[ch] = NULL;
|
||||
free(audio_decoder->Remain[ch]);
|
||||
audio_decoder->Remain[ch] = NULL;
|
||||
}
|
||||
}
|
||||
if (audio_decoder->ReSample) {
|
||||
audio_resample_close(audio_decoder->ReSample);
|
||||
audio_decoder->ReSample = NULL;
|
||||
@ -809,6 +843,238 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
** Set/update audio pts clock.
|
||||
**
|
||||
** @param audio_decoder audio decoder data
|
||||
*/
|
||||
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
|
||||
{
|
||||
struct timespec nowtime;
|
||||
int64_t tim_diff;
|
||||
int64_t pts_diff;
|
||||
int64_t drift;
|
||||
|
||||
AudioSetClock(pts);
|
||||
|
||||
// start drift detection
|
||||
if (audio_decoder->StartPTS == (int64_t) AV_NOPTS_VALUE && AudioGetDelay()) {
|
||||
audio_decoder->StartPTS = AudioGetClock();
|
||||
audio_decoder->LastPTS = audio_decoder->StartPTS;
|
||||
clock_gettime(CLOCK_REALTIME, &audio_decoder->StartTime);
|
||||
return;
|
||||
}
|
||||
|
||||
pts = AudioGetClock();
|
||||
clock_gettime(CLOCK_REALTIME, &nowtime);
|
||||
pts_diff = pts - audio_decoder->StartPTS;
|
||||
|
||||
tim_diff = (nowtime.tv_sec - audio_decoder->StartTime.tv_sec)
|
||||
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
|
||||
audio_decoder->StartTime.tv_nsec);
|
||||
drift = pts_diff * 1000 * 1000 / 90 - tim_diff;
|
||||
|
||||
if (abs(drift) > 100 * 1000 * 1000) {
|
||||
// drift too big, pts changed?
|
||||
audio_decoder->StartPTS = pts;
|
||||
audio_decoder->LastPTS = audio_decoder->StartPTS;
|
||||
audio_decoder->StartTime = nowtime;
|
||||
return;
|
||||
}
|
||||
// collect over some time
|
||||
if (pts - audio_decoder->LastPTS < 10 * 1000 * 90) {
|
||||
return;
|
||||
}
|
||||
audio_decoder->LastPTS = pts;
|
||||
|
||||
audio_decoder->Drift +=
|
||||
(int)((10 * audio_decoder->SampleRate * drift) / tim_diff);
|
||||
|
||||
if (audio_decoder->AvResample) {
|
||||
av_resample_compensate(audio_decoder->AvResample, audio_decoder->Drift,
|
||||
10 * audio_decoder->SampleRate);
|
||||
}
|
||||
|
||||
Info("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %g\n",
|
||||
audio_decoder->Drift, drift / (1000 * 1000), drift,
|
||||
(double)drift / tim_diff);
|
||||
printf("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %d\n",
|
||||
audio_decoder->Drift, drift / (1000 * 1000), drift,
|
||||
(int)((10 * audio_decoder->SampleRate * drift) / tim_diff));
|
||||
}
|
||||
|
||||
/**
|
||||
** Handle audio format changes.
|
||||
**
|
||||
** @param audio_decoder audio decoder data
|
||||
*/
|
||||
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
|
||||
{
|
||||
const AVCodecContext *audio_ctx;
|
||||
int err;
|
||||
int isAC3;
|
||||
|
||||
audio_ctx = audio_decoder->AudioCtx;
|
||||
|
||||
audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
|
||||
|
||||
// FIXME: use swr_convert from swresample (only in ffmpeg!)
|
||||
if (audio_decoder->ReSample) {
|
||||
audio_resample_close(audio_decoder->ReSample);
|
||||
audio_decoder->ReSample = NULL;
|
||||
}
|
||||
if (audio_decoder->AvResample) {
|
||||
av_resample_close(audio_decoder->AvResample);
|
||||
audio_decoder->AvResample = NULL;
|
||||
audio_decoder->RemainCount = 0;
|
||||
}
|
||||
|
||||
audio_decoder->SampleRate = audio_ctx->sample_rate;
|
||||
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
|
||||
audio_decoder->Channels = audio_ctx->channels;
|
||||
// SPDIF/HDMI passthrough
|
||||
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
|
||||
audio_decoder->HwChannels = 2;
|
||||
isAC3 = 1;
|
||||
} else {
|
||||
audio_decoder->HwChannels = audio_ctx->channels;
|
||||
isAC3 = 0;
|
||||
}
|
||||
|
||||
// channels not support?
|
||||
if ((err =
|
||||
AudioSetup(&audio_decoder->HwSampleRate,
|
||||
&audio_decoder->HwChannels, isAC3))) {
|
||||
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
|
||||
audio_ctx->sample_rate, audio_ctx->channels,
|
||||
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
|
||||
|
||||
if (err == 1) {
|
||||
audio_decoder->ReSample =
|
||||
av_audio_resample_init(audio_decoder->HwChannels,
|
||||
audio_ctx->channels, audio_decoder->HwSampleRate,
|
||||
audio_ctx->sample_rate, audio_ctx->sample_fmt,
|
||||
audio_ctx->sample_fmt, 16, 10, 0, 0.8);
|
||||
// libav-0.8_pre didn't support 6 -> 2 channels
|
||||
if (!audio_decoder->ReSample) {
|
||||
Error(_("codec/audio: resample setup error\n"));
|
||||
audio_decoder->HwChannels = 0;
|
||||
audio_decoder->HwSampleRate = 0;
|
||||
return;
|
||||
}
|
||||
} else {
|
||||
Debug(3, "codec/audio: audio setup error\n");
|
||||
// FIXME: handle errors
|
||||
audio_decoder->HwChannels = 0;
|
||||
audio_decoder->HwSampleRate = 0;
|
||||
return;
|
||||
}
|
||||
}
|
||||
// prepare audio drift resample
|
||||
if (!isAC3) {
|
||||
audio_decoder->AvResample =
|
||||
av_resample_init(audio_decoder->HwSampleRate,
|
||||
audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
|
||||
if (!audio_decoder->AvResample) {
|
||||
Error(_("codec/audio: AvResample setup error\n"));
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
** Codec enqueue audio samples.
|
||||
**
|
||||
** @param audio_decoder audio decoder data
|
||||
** @param data samples data
|
||||
** @param count number of samples
|
||||
**
|
||||
*/
|
||||
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
|
||||
{
|
||||
#ifdef USE_AUDIO_DRIFT_CORRECTION
|
||||
if (audio_decoder->AvResample) {
|
||||
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
|
||||
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
|
||||
int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
|
||||
int consumed;
|
||||
int i;
|
||||
int n;
|
||||
int ch;
|
||||
int bytes_n;
|
||||
|
||||
bytes_n = count / audio_decoder->HwChannels;
|
||||
// resize sample buffer, if needed
|
||||
if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) {
|
||||
audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n;
|
||||
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
|
||||
audio_decoder->Buffer[ch] =
|
||||
realloc(audio_decoder->Buffer[ch],
|
||||
audio_decoder->BufferSize);
|
||||
}
|
||||
}
|
||||
// copy remaining bytes into sample buffer
|
||||
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
|
||||
memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch],
|
||||
audio_decoder->RemainCount);
|
||||
}
|
||||
// deinterleave samples into sample buffer
|
||||
for (i = 0; i < bytes_n / 2; i++) {
|
||||
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
|
||||
audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i]
|
||||
= data[i * audio_decoder->HwChannels + ch];
|
||||
}
|
||||
}
|
||||
|
||||
bytes_n += audio_decoder->RemainSize;
|
||||
n = 0; // keep gcc lucky
|
||||
// resample the sample buffer into tmp buffer
|
||||
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
|
||||
n = av_resample(audio_decoder->AvResample, buftmp[ch],
|
||||
audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
|
||||
sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
|
||||
//printf("remain%d: %d = %d/%d\n", ch, n, consumed, bytes_n /2);
|
||||
// fixme remaining channels
|
||||
if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
|
||||
audio_decoder->RemainSize = bytes_n - consumed * 2;
|
||||
}
|
||||
audio_decoder->Remain[ch] =
|
||||
realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize);
|
||||
memcpy(audio_decoder->Remain[ch],
|
||||
audio_decoder->Buffer[ch] + consumed,
|
||||
audio_decoder->RemainSize);
|
||||
audio_decoder->RemainCount = audio_decoder->RemainSize;
|
||||
}
|
||||
|
||||
// interleave samples from sample buffer
|
||||
for (i = 0; i < n; i++) {
|
||||
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
|
||||
buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i];
|
||||
}
|
||||
}
|
||||
n *= 2;
|
||||
|
||||
#if 0
|
||||
// FIXME: must split channels, filter, join channels
|
||||
n = av_resample(audio_decoder->AvResample, buf, data, &consumed, count,
|
||||
sizeof(buf), 1);
|
||||
if (n < 0) {
|
||||
Error(_("codec/audio: can't av_resample\n"));
|
||||
return;
|
||||
}
|
||||
if (consumed != count) {
|
||||
Error(_("codec/audio: av_resample didn't consume all samples\n"));
|
||||
}
|
||||
#endif
|
||||
n *= audio_decoder->HwChannels;
|
||||
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
|
||||
AudioEnqueue(buf, n);
|
||||
return;
|
||||
}
|
||||
#endif
|
||||
CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
|
||||
AudioEnqueue(data, count);
|
||||
}
|
||||
|
||||
/**
|
||||
** Decode an audio packet.
|
||||
**
|
||||
@ -847,64 +1113,17 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
|
||||
avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
|
||||
#else
|
||||
#endif
|
||||
// Update audio clock
|
||||
|
||||
// update audio clock
|
||||
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
|
||||
AudioSetClock(avpkt->pts);
|
||||
CodecAudioSetClock(audio_decoder, avpkt->pts);
|
||||
|
||||
}
|
||||
// FIXME: must first play remainings bytes, than change and play new.
|
||||
if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
|
||||
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|
||||
|| audio_decoder->Channels != audio_ctx->channels) {
|
||||
int err;
|
||||
int isAC3;
|
||||
|
||||
audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
|
||||
// FIXME: use swr_convert from swresample (only in ffmpeg!)
|
||||
if (audio_decoder->ReSample) {
|
||||
audio_resample_close(audio_decoder->ReSample);
|
||||
audio_decoder->ReSample = NULL;
|
||||
}
|
||||
|
||||
audio_decoder->SampleRate = audio_ctx->sample_rate;
|
||||
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
|
||||
audio_decoder->Channels = audio_ctx->channels;
|
||||
// SPDIF/HDMI passthrough
|
||||
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
|
||||
audio_decoder->HwChannels = 2;
|
||||
isAC3 = 1;
|
||||
} else {
|
||||
audio_decoder->HwChannels = audio_ctx->channels;
|
||||
isAC3 = 0;
|
||||
}
|
||||
|
||||
// channels not support?
|
||||
if ((err =
|
||||
AudioSetup(&audio_decoder->HwSampleRate,
|
||||
&audio_decoder->HwChannels, isAC3))) {
|
||||
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
|
||||
audio_ctx->sample_rate, audio_ctx->channels,
|
||||
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
|
||||
|
||||
if (err == 1) {
|
||||
audio_decoder->ReSample =
|
||||
av_audio_resample_init(audio_decoder->HwChannels,
|
||||
audio_ctx->channels, audio_decoder->HwSampleRate,
|
||||
audio_ctx->sample_rate, audio_ctx->sample_fmt,
|
||||
audio_ctx->sample_fmt, 16, 10, 0, 0.8);
|
||||
// libav-0.8_pre didn't support 6 -> 2 channels
|
||||
if (!audio_decoder->ReSample) {
|
||||
Error(_("codec/audio: resample setup error\n"));
|
||||
audio_decoder->HwChannels = 0;
|
||||
audio_decoder->HwSampleRate = 0;
|
||||
}
|
||||
} else {
|
||||
Debug(3, "codec/audio: audio setup error\n");
|
||||
// FIXME: handle errors
|
||||
audio_decoder->HwChannels = 0;
|
||||
audio_decoder->HwSampleRate = 0;
|
||||
return;
|
||||
}
|
||||
}
|
||||
CodecAudioUpdateFormat(audio_decoder);
|
||||
}
|
||||
|
||||
if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
|
||||
@ -931,9 +1150,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
|
||||
audio_decoder->HwChannels *
|
||||
av_get_bytes_per_sample(audio_ctx->sample_fmt);
|
||||
Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
|
||||
CodecReorderAudioFrame(outbuf, outlen,
|
||||
audio_decoder->HwChannels);
|
||||
AudioEnqueue(outbuf, outlen);
|
||||
CodecAudioEnqueue(audio_decoder, outbuf, outlen);
|
||||
}
|
||||
} else {
|
||||
#ifdef USE_PASSTHROUGH
|
||||
@ -955,6 +1172,10 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
|
||||
buf[3] = htole16(avpkt->size * 8);
|
||||
swab(avpkt->data, buf + 4, avpkt->size);
|
||||
memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
|
||||
// don't play with the ac-3 samples
|
||||
AudioEnqueue(buf, buf_sz);
|
||||
// FIXME: handle drift.
|
||||
return;
|
||||
}
|
||||
#if 0
|
||||
//
|
||||
@ -1008,8 +1229,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
|
||||
// True HD?
|
||||
#endif
|
||||
#endif
|
||||
CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
|
||||
AudioEnqueue(buf, buf_sz);
|
||||
CodecAudioEnqueue(audio_decoder, buf, buf_sz);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user