Improved pass-through (PCM+EAC3) support.

This commit is contained in:
Johns
2013-02-11 16:53:51 +01:00
parent 145d65ff01
commit 2cd667fb44
9 changed files with 451 additions and 269 deletions

426
codec.c
View File

@@ -1,7 +1,7 @@
///
/// @file codec.c @brief Codec functions
///
/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
/// Copyright (c) 2009 - 2013 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
@@ -30,13 +30,13 @@
/// many bugs and incompatiblity in it. Don't use this shit.
///
/// compile with passthrough support (stable, ac3 only)
/// compile with pass-through support (stable, AC-3, E-AC-3 only)
#define USE_PASSTHROUGH
/// compile audio drift correction support (experimental)
/// compile audio drift correction support (very experimental)
#define USE_AUDIO_DRIFT_CORRECTION
/// compile AC3 audio drift correction support (experimental)
/// compile AC-3 audio drift correction support (very experimental)
#define USE_AC3_DRIFT_CORRECTION
/// use ffmpeg libswresample API
/// use ffmpeg libswresample API (autodected, Makefile)
#define noUSE_SWRESAMPLE
#include <stdio.h>
@@ -633,7 +633,7 @@ struct _audio_decoder_
AVCodec *AudioCodec; ///< audio codec
AVCodecContext *AudioCtx; ///< audio codec context
int PassthroughAC3; ///< current ac-3 pass-through
char Passthrough; ///< current pass-through flags
int SampleRate; ///< current stream sample rate
int Channels; ///< current stream channels
@@ -651,6 +651,10 @@ struct _audio_decoder_
#endif
#endif
uint16_t Spdif[24576 / 2]; ///< SPDIF output buffer
int SpdifIndex; ///< index into SPDIF output buffer
int SpdifCount; ///< SPDIF repeat counter
int64_t LastDelay; ///< last delay
struct timespec LastTime; ///< last time
int64_t LastPTS; ///< last PTS
@@ -670,24 +674,32 @@ struct _audio_decoder_
#endif
};
///
/// IEC Data type enumeration.
///
enum IEC61937
{
IEC61937_AC3 = 0x01, ///< AC-3 data
// FIXME: more data types
IEC61937_EAC3 = 0x15, ///< E-AC-3 data
};
#ifdef USE_AUDIO_DRIFT_CORRECTION
#define CORRECT_PCM 1 ///< do PCM audio-drift correction
#define CORRECT_AC3 2 ///< do AC3<EFBFBD> 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;
#endif
#ifdef USE_PASSTHROUGH
//static char CodecPassthroughPCM; ///< pass pcm through (unsupported)
static char CodecPassthroughAC3; ///< pass ac3 through
//static char CodecPassthroughDTS; ///< pass dts through (unsupported)
//static char CodecPassthroughMPA; ///< pass mpa through (unsupported)
///
/// Pass-through flags: CodecPCM, CodecAC3, CodecEAC3, ...
///
static char CodecPassthrough;
#else
static const int CodecPassthroughAC3 = 0;
static const int CodecPassthrough = 0;
#endif
static char CodecDownmix; ///< enable ac-3 downmix
static char CodecDownmix; ///< enable AC-3 decoder downmix
/**
** Allocate a new audio decoder context.
@@ -840,7 +852,7 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
void CodecSetAudioDrift(int mask)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
CodecAudioDrift = mask & 3;
CodecAudioDrift = mask & (CORRECT_PCM | CORRECT_AC3);
#endif
(void)mask;
}
@@ -848,12 +860,12 @@ void CodecSetAudioDrift(int mask)
/**
** Set audio pass-through.
**
** @param mask enable mask (PCM, AC3)
** @param mask enable mask (PCM, AC3, EAC3)
*/
void CodecSetAudioPassthrough(int mask)
{
#ifdef USE_PASSTHROUGH
CodecPassthroughAC3 = mask & 1 ? 1 : 0;
CodecPassthrough = mask & (CodecPCM | CodecAC3 | CodecEAC3);
#endif
(void)mask;
}
@@ -932,6 +944,178 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
}
}
/**
** Handle audio format changes helper.
**
** @param audio_decoder audio decoder data
** @param[out] passthrough pass-through output
*/
static int CodecAudioUpdateHelper(AudioDecoder * audio_decoder,
int *passthrough)
{
const AVCodecContext *audio_ctx;
int err;
audio_ctx = audio_decoder->AudioCtx;
Debug(3, "codec/audio: format change %s %dHz *%d channels%s%s%s%s%s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
audio_ctx->channels, CodecPassthrough & CodecPCM ? " PCM" : "",
CodecPassthrough & CodecMPA ? " MPA" : "",
CodecPassthrough & CodecAC3 ? " AC3" : "",
CodecPassthrough & CodecEAC3 ? " EAC3" : "",
CodecPassthrough ? " pass-through" : "");
*passthrough = 0;
audio_decoder->SampleRate = audio_ctx->sample_rate;
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
audio_decoder->Channels = audio_ctx->channels;
audio_decoder->HwChannels = audio_ctx->channels;
audio_decoder->Passthrough = CodecPassthrough;
// SPDIF/HDMI pass-through
if ((CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3)
|| (CodecPassthrough & CodecEAC3
&& audio_ctx->codec_id == CODEC_ID_EAC3)) {
audio_decoder->HwChannels = 2;
audio_decoder->SpdifIndex = 0; // reset buffer
audio_decoder->SpdifCount = 0;
*passthrough = 1;
}
// channels not support?
if ((err =
AudioSetup(&audio_decoder->HwSampleRate,
&audio_decoder->HwChannels, *passthrough))) {
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return err;
}
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);
return 0;
}
/**
** Audio pass-through decoder helper.
**
** @param audio_decoder audio decoder data
** @param avpkt undecoded audio packet
*/
static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder,
const AVPacket * avpkt)
{
#ifdef USE_PASSTHROUGH
const AVCodecContext *audio_ctx;
audio_ctx = audio_decoder->AudioCtx;
// SPDIF/HDMI passthrough
if (CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
uint16_t *spdif;
int spdif_sz;
spdif = audio_decoder->Spdif;
spdif_sz = 6144;
#ifdef USE_AC3_DRIFT_CORRECTION
// FIXME: this works with some TVs/AVReceivers
// FIXME: write burst size drift correction, which should work with all
if (CodecAudioDrift & CORRECT_AC3) {
int x;
x = (audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * spdif_sz)) / (10 *
audio_decoder->HwSampleRate * 100);
audio_decoder->DriftFrac =
(audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * spdif_sz)) % (10 *
audio_decoder->HwSampleRate * 100);
// round to word border
x *= audio_decoder->HwChannels * 4;
if (x < -64) { // limit correction
x = -64;
} else if (x > 64) {
x = 64;
}
spdif_sz += x;
}
#endif
// build SPDIF header and append A52 audio to it
// avpkt is the original data
if (spdif_sz < avpkt->size + 8) {
Error(_("codec/audio: decoded data smaller than encoded\n"));
return -1;
}
spdif[0] = htole16(0xF872); // iec 61937 sync word
spdif[1] = htole16(0x4E1F);
spdif[2] = htole16(IEC61937_AC3 | (avpkt->data[5] & 0x07) << 8);
spdif[3] = htole16(avpkt->size * 8);
// copy original data for output
// FIXME: not 100% sure, if endian is correct on not intel hardware
swab(avpkt->data, spdif + 4, avpkt->size);
// FIXME: don't need to clear always
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 1;
}
if (CodecPassthrough & CodecEAC3 && audio_ctx->codec_id == CODEC_ID_EAC3) {
uint16_t *spdif;
int spdif_sz;
int repeat;
// build SPDIF header and append A52 audio to it
// avpkt is the original data
spdif = audio_decoder->Spdif;
spdif_sz = 6144;
// 24576 = 4 * 6144
if (spdif_sz < audio_decoder->SpdifIndex + avpkt->size + 8) {
Error(_("codec/audio: decoded data smaller than encoded\n"));
return -1;
}
// check if we must pack multiple packets
repeat = 1;
if ((avpkt->data[4] & 0xc0) != 0xc0) { // fscod
static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
// fscod2
repeat = eac3_repeat[(avpkt->data[4] & 0x30) >> 4];
}
//fprintf(stderr, "repeat %d\n", repeat);
// copy original data for output
// pack upto repeat EAC-3 pakets into one IEC 61937 burst
// FIXME: not 100% sure, if endian is correct on not intel hardware
swab(avpkt->data, spdif + 4 + audio_decoder->SpdifIndex, avpkt->size);
audio_decoder->SpdifIndex += avpkt->size;
if (++audio_decoder->SpdifCount < repeat) {
return 1;
}
spdif[0] = htole16(0xF872); // iec 61937 sync word
spdif[1] = htole16(0x4E1F);
spdif[2] = htole16(IEC61937_EAC3);
spdif[3] = htole16(audio_decoder->SpdifIndex * 8);
memset(spdif + 4 + audio_decoder->SpdifIndex / 2, 0,
spdif_sz - 8 - audio_decoder->SpdifIndex);
// don't play with the eac-3 samples
AudioEnqueue(spdif, spdif_sz);
audio_decoder->SpdifIndex = 0;
audio_decoder->SpdifCount = 0;
return 1;
}
#endif
return 0;
}
#ifndef USE_SWRESAMPLE
/**
@@ -1007,8 +1191,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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)) {
if ((CodecAudioDrift & CORRECT_AC3) && (!CodecPassthroughAC3
|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
&& (!CodecPassthroughEAC3
|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
@@ -1045,14 +1231,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
** Handle audio format changes.
**
** @param audio_decoder audio decoder data
**
** @note this is the old not good supported version
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
int passthrough;
const AVCodecContext *audio_ctx;
int err;
int isAC3;
// FIXME: use swr_convert from swresample (only in ffmpeg!)
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
@@ -1064,28 +1251,8 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
}
audio_ctx = audio_decoder->AudioCtx;
Debug(3, "codec/audio: format change %dHz %d channels %s\n",
audio_ctx->sample_rate, audio_ctx->channels,
CodecPassthroughAC3 ? "pass-through" : "");
if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {
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: resample %dHz *%d -> %dHz *%d\n",
audio_ctx->sample_rate, audio_ctx->channels,
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
@@ -1101,19 +1268,21 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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;
}
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
}
// prepare audio drift resample
#ifdef USE_AUDIO_DRIFT_CORRECTION
if ((CodecAudioDrift & 1) && !isAC3) {
if (CodecAudioDrift & CORRECT_PCM) {
if (audio_decoder->AvResample) {
Error(_("codec/audio: overwrite resample\n"));
}
@@ -1144,7 +1313,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
if ((CodecAudioDrift & 1) && audio_decoder->AvResample) {
if ((CodecAudioDrift & CORRECT_PCM) && 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];
@@ -1205,12 +1374,16 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
n *= 2;
n *= audio_decoder->HwChannels;
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
}
AudioEnqueue(buf, n);
return;
}
#endif
CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
}
AudioEnqueue(data, count);
}
@@ -1232,6 +1405,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
buf_sz = sizeof(buf);
l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
if (avpkt->size != l) {
@@ -1250,7 +1424,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// FIXME: must first play remainings bytes, than change and play new.
if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
if (audio_decoder->Passthrough != CodecPassthrough
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
@@ -1283,48 +1457,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioEnqueue(audio_decoder, outbuf, outlen);
}
} else {
#ifdef USE_PASSTHROUGH
// SPDIF/HDMI passthrough
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
// build SPDIF header and append A52 audio to it
// avpkt is the original data
buf_sz = 6144;
#ifdef USE_AC3_DRIFT_CORRECTION
if (CodecAudioDrift & 2) {
int x;
x = (audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * buf_sz)) / (10 *
audio_decoder->HwSampleRate * 100);
audio_decoder->DriftFrac =
(audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * buf_sz)) % (10 *
audio_decoder->HwSampleRate * 100);
x *= audio_decoder->HwChannels * 4;
if (x < -64) { // limit correction
x = -64;
} else if (x > 64) {
x = 64;
}
buf_sz += x;
}
#endif
if (buf_sz < avpkt->size + 8) {
Error(_
("codec/audio: decoded data smaller than encoded\n"));
return;
}
// copy original data for output
// FIXME: not 100% sure, if endian is correct
buf[0] = htole16(0xF872); // iec 61937 sync word
buf[1] = htole16(0x4E1F);
buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
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);
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
#if 0
@@ -1377,7 +1510,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
}
// DTS HD?
// True HD?
#endif
#endif
CodecAudioEnqueue(audio_decoder, buf, buf_sz);
}
@@ -1461,8 +1593,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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)) {
if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
&& (!(CodecPassthrough & CodecEAC3)
|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
@@ -1504,49 +1638,27 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
int passthrough;
const AVCodecContext *audio_ctx;
int err;
int isAC3;
if (CodecAudioUpdateHelper(audio_decoder, &passthrough)) {
// FIXME: handle swresample format conversions.
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
}
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;
#ifdef DEBUG
if (audio_ctx->sample_fmt == AV_SAMPLE_FMT_S16
&& audio_ctx->sample_rate == audio_decoder->HwSampleRate
&& !CodecAudioDrift) {
// FIXME: use Resample only, when it is needed!
fprintf(stderr, "no resample needed\n");
}
// 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);
#endif
audio_decoder->Resample =
swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
@@ -1579,6 +1691,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
frame.data[0] = NULL;
n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
(AVPacket *) avpkt);
@@ -1602,42 +1715,20 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
if (audio_decoder->Passthrough != CodecPassthrough
|| 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);
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
#endif
if (0) {
char strbuf[32];
int data_sz;
@@ -1665,12 +1756,19 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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);
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf,
n * 2 * audio_decoder->HwChannels,
audio_decoder->HwChannels);
}
AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
}
return;
}
#ifdef DEBUG
// should be never reached
fprintf(stderr, "oops\n");
#endif
}
#endif