frodovdr
/
vdr-plugin-softhddevice
mirror of https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Add audio compatibility with >=ffmpeg 1.1.

This commit is contained in:
Johns 2013-01-24 21:42:39 +01:00
parent 04286fb2ad
commit f9998e7664
4 changed files with 342 additions and 18 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
ChangeLog
View File

@ -1,6 +1,7 @@
User johns
Date:
Add compatibility with >=ffmpeg 1.1.
Adds PIP (Picture-in-Picture) support.
Split mpeg packets in receiver thread.

20
Makefile
View File

@ -24,6 +24,8 @@ CONFIG += -DUSE_PIP # too experimental PIP support
#CONFIG += -DHAVE_PTHREAD_NAME # supports new pthread_setname_np
#CONFIG += -DNO_TS_AUDIO # disable ts audio parser
#CONFIG += -DUSE_TS_VIDEO # build new ts video parser
# use ffmpeg libswresample
CONFIG += $(shell pkg-config --exists libswresample && echo "-DUSE_SWRESAMPLE")
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")
@ -73,12 +75,12 @@ _CFLAGS = $(DEFINES) $(INCLUDES) \
`pkg-config --cflags x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
`pkg-config --cflags gl glu` \
$(if $(findstring USE_VDPAU,$(CONFIG)), \
`pkg-config --cflags vdpau`) \
$(if $(findstring USE_SWRESAMPLE,$(CONFIG)), \
$(shell pkg-config --cflags libswresample)) \
$(if $(findstring USE_VAAPI,$(CONFIG)), \
`pkg-config --cflags libva-x11 libva-glx libva`) \
`pkg-config --cflags libva-x11 libva-glx libva`) \
$(if $(findstring USE_ALSA,$(CONFIG)), \
`pkg-config --cflags alsa`)
`pkg-config --cflags alsa`)
#override _CFLAGS += -Werror
override CXXFLAGS += $(_CFLAGS)
@ -89,12 +91,14 @@ LIBS += -lrt \
`pkg-config --libs x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
`pkg-config --libs gl glu` \
$(if $(findstring USE_SWRESAMPLE,$(CONFIG)), \
$(shell pkg-config --libs libswresample)) \
$(if $(findstring USE_VDPAU,$(CONFIG)), \
`pkg-config --libs vdpau`) \
`pkg-config --libs vdpau`) \
$(if $(findstring USE_VAAPI,$(CONFIG)), \
`pkg-config --libs libva-x11 libva-glx libva`) \
`pkg-config --libs libva-x11 libva-glx libva`) \
$(if $(findstring USE_ALSA,$(CONFIG)), \
`pkg-config --libs alsa`)
`pkg-config --libs alsa`)
### The object files (add further files here):
@ -132,7 +136,7 @@ I18Npot = $(PODIR)/$(PLUGIN).pot
%.mo: %.po
msgfmt -c -o $@ $<
$(I18Npot): $(wildcard *.cpp) $(wildcard *.c)
$(I18Npot): $(wildcard *.cpp) $(wildcard *.c)
xgettext -C -cTRANSLATORS --no-wrap --no-location -k -ktr -ktrNOOP \
-k_ -k_N --package-name=VDR --package-version=$(VDRVERSION) \
--msgid-bugs-address='<see README>' -o $@ $^

2
Todo
View File

@ -117,6 +117,8 @@ setup:
unsorted:
stoping vdr while plugin is suspended opens and closes a window.
svdrp prim: support plugin names for device numbers.
hangup PipVideoStream -> Vdpau_get_format -> xcb -> poll
+ lock DecoderLockMutex
future features (not planed for 1.0 - 1.5)

337
codec.c
View File

@ -36,6 +36,8 @@
#define USE_AUDIO_DRIFT_CORRECTION
/// compile AC3 audio drift correction support (experimental)
#define USE_AC3_DRIFT_CORRECTION
/// use ffmpeg libswresample API
#define noUSE_SWRESAMPLE
#include <stdio.h>
#include <unistd.h>
@ -58,6 +60,9 @@
#ifdef USE_VDPAU
#include <libavcodec/vdpau.h>
#endif
#ifdef USE_SWRESAMPLE
#include <libswresample/swresample.h>
#endif
#ifndef __USE_GNU
#define __USE_GNU
@ -629,7 +634,16 @@ struct _audio_decoder_
int HwSampleRate; ///< hw sample rate
int HwChannels; ///< hw channels
#ifndef USE_SWRESAMPLE
ReSampleContext *ReSample; ///< audio resampling context
#endif
#ifdef USE_SWRESAMPLE
#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
struct SwrContext *Resample; ///< audio software resample context
#else
SwrContext *Resample; ///< audio software resample context
#endif
#endif
int64_t LastDelay; ///< last delay
struct timespec LastTime; ///< last time
@ -639,6 +653,7 @@ struct _audio_decoder_
int DriftCorr; ///< audio drift correction value
int DriftFrac; ///< audio drift fraction for ac3
#ifndef USE_SWRESAMPLE
struct AVResampleContext *AvResample; ///< second audio resample context
#define MAX_CHANNELS 8 ///< max number of channels supported
int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers
@ -646,9 +661,12 @@ struct _audio_decoder_
int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples
int RemainSize; ///< size of remain buffer
int RemainCount; ///< number of remaining samples
#endif
};
#ifdef USE_AUDIO_DRIFT_CORRECTION
#define CORRECT_PCM 1 ///< do PCM audio-drift correction
#define CORRECT_AC3 2 ///< do AC3§ audio-drift correction
static char CodecAudioDrift; ///< flag: enable audio-drift correction
#else
static const int CodecAudioDrift = 0;
@ -718,10 +736,16 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
}
if (CodecDownmix) {
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,61,100)
audio_decoder->AudioCtx->request_channels = 2;
#endif
audio_decoder->AudioCtx->request_channel_layout =
AV_CH_LAYOUT_STEREO_DOWNMIX;
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,61,100)
// this has no effect
// audio_decoder->AudioCtx->request_sample_fmt = AV_SAMPLE_FMT_S16;
#endif
pthread_mutex_lock(&CodecLockMutex);
// open codec
#if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,5,0)
@ -768,6 +792,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
void CodecAudioClose(AudioDecoder * audio_decoder)
{
// FIXME: output any buffered data
#ifndef USE_SWRESAMPLE
if (audio_decoder->AvResample) {
int ch;
@ -787,6 +812,12 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
}
#endif
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample) {
swr_free(&audio_decoder->Resample);
}
#endif
if (audio_decoder->AudioCtx) {
pthread_mutex_lock(&CodecLockMutex);
avcodec_close(audio_decoder->AudioCtx);
@ -895,6 +926,8 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
}
}
#ifndef USE_SWRESAMPLE
/**
** Set/update audio pts clock.
**
@ -916,14 +949,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
if (!delay) {
return;
}
clock_gettime(CLOCK_REALTIME, &nowtime);
clock_gettime(CLOCK_MONOTONIC, &nowtime);
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital delay %" PRId64 "ms\n", delay / 90);
Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
delay / 90);
return;
}
// collect over some time
@ -1205,14 +1239,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
}
Error(_("codec: error more than one frame data\n"));
}
#ifdef notyetFF_API_OLD_DECODE_AUDIO
// FIXME: ffmpeg git comeing
int got_frame;
avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
#else
#endif
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
@ -1352,6 +1378,297 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
}
}
#endif
#ifdef USE_SWRESAMPLE
/**
** Set/update audio pts clock.
**
** @param audio_decoder audio decoder data
** @param pts presentation timestamp
*/
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
struct timespec nowtime;
int64_t delay;
int64_t tim_diff;
int64_t pts_diff;
int drift;
int corr;
AudioSetClock(pts);
delay = AudioGetDelay();
if (!delay) {
return;
}
clock_gettime(CLOCK_MONOTONIC, &nowtime);
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
delay / 90);
return;
}
// collect over some time
pts_diff = pts - audio_decoder->LastPTS;
if (pts_diff < 10 * 1000 * 90) {
return;
}
tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
audio_decoder->LastTime.tv_nsec);
drift =
(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
audio_decoder->LastDelay;
// adjust rounding error
nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
if (0) {
Debug(3,
"codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4"
PRId64 "ms %f %d\n", pts_diff / 90, tim_diff / (1000 * 1000),
delay / 90, drift / 90.0, audio_decoder->DriftCorr);
}
// underruns and av_resample have the same time :(((
if (abs(drift) > 10 * 90) {
// drift too big, pts changed?
Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
audio_decoder->DriftCorr, drift / 90);
audio_decoder->LastDelay = 0;
#ifdef DEBUG
corr = 0; // keep gcc happy
#endif
} else {
drift += audio_decoder->Drift;
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
audio_decoder->DriftCorr = -corr;
}
if (audio_decoder->DriftCorr < -20000) { // limit correction
audio_decoder->DriftCorr = -20000;
} else if (audio_decoder->DriftCorr > 20000) {
audio_decoder->DriftCorr = 20000;
}
}
if (audio_decoder->Resample && audio_decoder->DriftCorr) {
int distance;
// try workaround for buggy ffmpeg 0.10
if (abs(audio_decoder->DriftCorr) < 2000) {
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
} else {
distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
}
if (swr_set_compensation(audio_decoder->Resample,
audio_decoder->DriftCorr / 10, distance)) {
Debug(3, "codec/audio: swr_set_compensation failed\n");
}
}
if (1) {
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
}
/**
** 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;
Debug(3, "codec/audio: format change %s %dHz *%d channels %s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
audio_ctx->channels, CodecPassthroughAC3 ? "pass-through" : "");
audio_decoder->SampleRate = audio_ctx->sample_rate;
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
audio_decoder->Channels = audio_ctx->channels;
audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
// 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: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return;
}
if (isAC3) { // no AC3 conversion allowed
return;
}
Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
audio_decoder->Resample =
swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
AV_SAMPLE_FMT_S16, audio_decoder->HwSampleRate,
audio_ctx->channel_layout, audio_ctx->sample_fmt,
audio_ctx->sample_rate, 0, NULL);
if (audio_decoder->Resample) {
swr_init(audio_decoder->Resample);
} else {
Error(_("codec/audio: can't setup resample\n"));
}
}
/**
** Decode an audio packet.
**
** PTS must be handled self.
**
** @note the caller has not aligned avpkt and not cleared the end.
**
** @param audio_decoder audio decoder data
** @param avpkt audio packet
*/
void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
AVCodecContext *audio_ctx;
AVFrame frame;
int got_frame;
int n;
audio_ctx = audio_decoder->AudioCtx;
frame.data[0] = NULL;
n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
(AVPacket *) avpkt);
if (n != avpkt->size) {
if (n == AVERROR(EAGAIN)) {
Error(_("codec/audio: latm\n"));
return;
}
if (n < 0) { // no audio frame could be decompressed
Error(_("codec/audio: bad audio frame\n"));
return;
}
Error(_("codec/audio: error more than one frame data\n"));
}
if (!got_frame) {
Error(_("codec/audio: no frame\n"));
return;
}
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
}
if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
return; // unsupported sample format
}
#ifdef USE_PASSTHROUGH
// SPDIF/HDMI passthrough
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
int16_t spdif[6144 / 2];
int spdif_sz;
// build SPDIF header and append A52 audio to it
// avpkt is the original data
spdif_sz = 6144;
if (spdif_sz < avpkt->size + 8) {
Error(_("codec/audio: decoded data smaller than encoded\n"));
return;
}
// copy original data for output
spdif[0] = htole16(0xF872); // iec 61937 sync word
spdif[1] = htole16(0x4E1F);
spdif[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
spdif[3] = htole16(avpkt->size * 8);
// FIXME: not 100% sure, if endian is correct on not intel hardware
swab(avpkt->data, spdif + 4, avpkt->size);
memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
// don't play with the ac-3 samples
AudioEnqueue(spdif, spdif_sz);
return;
}
#endif
if (0) {
char strbuf[32];
int data_sz;
int plane_sz;
data_sz =
av_samples_get_buffer_size(&plane_sz, audio_ctx->channels,
frame.nb_samples, audio_ctx->sample_fmt, 1);
fprintf(stderr, "codec/audio: sample_fmt %s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt));
av_get_channel_layout_string(strbuf, 32, audio_ctx->channels,
audio_ctx->channel_layout);
fprintf(stderr, "codec/audio: layout %s\n", strbuf);
fprintf(stderr,
"codec/audio: channels %d samples %d plane %d data %d\n",
audio_ctx->channels, frame.nb_samples, plane_sz, data_sz);
}
if (audio_decoder->Resample) {
uint8_t outbuf[8192 * 2 * 8];
uint8_t *out[1];
out[0] = outbuf;
n = swr_convert(audio_decoder->Resample, out,
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(const uint8_t **)frame.extended_data, frame.nb_samples);
if (n > 0) {
CodecReorderAudioFrame((int16_t *) outbuf,
n * 2 * audio_decoder->HwChannels, audio_decoder->HwChannels);
AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
}
return;
}
}
#endif
/**
** Flush the audio decoder.
**