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Code Issues Releases Wiki Activity

Removed old cruft.

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This commit is contained in:
Johns 2012-02-26 14:30:46 +01:00
parent 769f00b4f6
commit e977007dd3
1 changed files with 13 additions and 304 deletions
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307
codec.c
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@ -30,11 +30,6 @@
/// many bugs and incompatiblity in it. Don't use this shit. /// many bugs and incompatiblity in it. Don't use this shit.
/// ///
/**
** use av_parser to support insane dvb audio streams.
*/
#define noUSE_AVPARSER
/// compile with passthrough support (experimental) /// compile with passthrough support (experimental)
#define USE_PASSTHROUGH #define USE_PASSTHROUGH
@ -603,10 +598,6 @@ struct _audio_decoder_
AVCodec *AudioCodec; ///< audio codec AVCodec *AudioCodec; ///< audio codec
AVCodecContext *AudioCtx; ///< audio codec context AVCodecContext *AudioCtx; ///< audio codec context
#ifdef USE_AVPARSER
/// audio parser to support insane dvb streaks
AVCodecParserContext *AudioParser;
#endif
int PassthroughAC3; ///< current ac-3 pass-through int PassthroughAC3; ///< current ac-3 pass-through
int SampleRate; ///< current stream sample rate int SampleRate; ///< current stream sample rate
int Channels; ///< current stream channels int Channels; ///< current stream channels
@ -615,7 +606,6 @@ struct _audio_decoder_
int HwChannels; ///< hw channels int HwChannels; ///< hw channels
ReSampleContext *ReSample; ///< audio resampling context ReSampleContext *ReSample; ///< audio resampling context
}; };
#ifdef USE_PASSTHROUGH #ifdef USE_PASSTHROUGH
@ -667,7 +657,6 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
int codec_id) int codec_id)
{ {
AVCodec *audio_codec; AVCodec *audio_codec;
//AVDictionary *av_dict;
if (name && (audio_codec = avcodec_find_decoder_by_name(name))) { if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: audio decoder '%s' found\n", name); Debug(3, "codec: audio decoder '%s' found\n", name);
@ -694,15 +683,20 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
Fatal(_("codec: can't open audio codec\n")); Fatal(_("codec: can't open audio codec\n"));
} }
#else #else
//av_dict = NULL; if (1) {
AVDictionary *av_dict;
av_dict = NULL;
// FIXME: import settings
//av_dict_set(&av_dict, "dmix_mode", "0", 0); //av_dict_set(&av_dict, "dmix_mode", "0", 0);
//av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0); //av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
//av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0); //av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, NULL) < 0) { if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
pthread_mutex_unlock(&CodecLockMutex); pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open audio codec\n")); Fatal(_("codec: can't open audio codec\n"));
} }
//av_dict_free(&av_dict); av_dict_free(&av_dict);
}
#endif #endif
pthread_mutex_unlock(&CodecLockMutex); pthread_mutex_unlock(&CodecLockMutex);
Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name); Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name);
@ -712,12 +706,6 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
// we do not send complete frames // we do not send complete frames
audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED; audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
} }
#ifdef USE_AVPARSER
if (!(audio_decoder->AudioParser =
av_parser_init(audio_decoder->AudioCtx->codec_id))) {
Fatal(_("codec: can't init audio parser\n"));
}
#endif
audio_decoder->SampleRate = 0; audio_decoder->SampleRate = 0;
audio_decoder->Channels = 0; audio_decoder->Channels = 0;
audio_decoder->HwSampleRate = 0; audio_decoder->HwSampleRate = 0;
@ -736,12 +724,6 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
audio_resample_close(audio_decoder->ReSample); audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL; audio_decoder->ReSample = NULL;
} }
#ifdef USE_AVPARSER
if (audio_decoder->AudioParser) {
av_parser_close(audio_decoder->AudioParser);
audio_decoder->AudioParser = NULL;
}
#endif
if (audio_decoder->AudioCtx) { if (audio_decoder->AudioCtx) {
pthread_mutex_lock(&CodecLockMutex); pthread_mutex_lock(&CodecLockMutex);
avcodec_close(audio_decoder->AudioCtx); avcodec_close(audio_decoder->AudioCtx);
@ -827,268 +809,6 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
} }
} }
#ifdef USE_AVPARSER
/**
** Decode an audio packet.
**
** PTS must be handled self.
**
** @param audio_decoder audio decoder data
** @param avpkt audio packet
*/
void CodecAudioDecodeOld(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
AVCodecContext *audio_ctx;
int index;
//#define spkt avpkt
#if 1 // didn't fix crash in av_parser_parse2
AVPacket spkt[1];
// av_new_packet reserves FF_INPUT_BUFFER_PADDING_SIZE and clears it
if (av_new_packet(spkt, avpkt->size)) {
Error(_("codec: out of memory\n"));
return;
}
memcpy(spkt->data, avpkt->data, avpkt->size);
spkt->pts = avpkt->pts;
spkt->dts = avpkt->dts;
#endif
#ifdef DEBUG
if (!audio_decoder->AudioParser) {
Fatal(_("codec: internal error parser freeded while running\n"));
}
#endif
audio_ctx = audio_decoder->AudioCtx;
index = 0;
while (spkt->size > index) {
int n;
int l;
AVPacket dpkt[1];
av_init_packet(dpkt);
n = av_parser_parse2(audio_decoder->AudioParser, audio_ctx,
&dpkt->data, &dpkt->size, spkt->data + index, spkt->size - index,
!index ? spkt->pts : (int64_t) AV_NOPTS_VALUE,
!index ? spkt->dts : (int64_t) AV_NOPTS_VALUE, -1);
// FIXME: make this a function for both #ifdef cases
if (dpkt->size) {
int buf_sz;
dpkt->pts = audio_decoder->AudioParser->pts;
dpkt->dts = audio_decoder->AudioParser->dts;
buf_sz = sizeof(buf);
l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt);
if (l == AVERROR(EAGAIN)) {
index += n; // this is needed for aac latm
continue;
}
if (l < 0) { // no audio frame could be decompressed
Error(_("codec: error audio data at %d\n"), index);
break;
}
#ifdef notyetFF_API_OLD_DECODE_AUDIO
// FIXME: ffmpeg git comeing
int got_frame;
avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
#else
#endif
// Update audio clock
if (dpkt->pts != (int64_t) AV_NOPTS_VALUE) {
AudioSetClock(dpkt->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;
break;
}
}
}
if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
// need to resample audio
if (audio_decoder->ReSample) {
int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE]
__attribute__ ((aligned(16)));
int outlen;
// FIXME: libav-0.7.2 crash here
outlen =
audio_resample(audio_decoder->ReSample, outbuf, buf,
buf_sz);
#ifdef DEBUG
if (outlen != buf_sz) {
Debug(3, "codec/audio: possible fixed ffmpeg\n");
}
#endif
if (outlen) {
// outlen seems to be wrong in ffmpeg-0.9
outlen /= audio_decoder->Channels *
av_get_bytes_per_sample(audio_ctx->sample_fmt);
outlen *=
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);
}
} 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
// dpkt is the original data
buf_sz = 6144;
if (buf_sz < dpkt->size + 8) {
Error(_
("codec/audio: decoded data smaller than encoded\n"));
break;
}
// 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 | (dpkt->data[5] & 0x07) << 8);
buf[3] = htole16(dpkt->size * 8);
swab(dpkt->data, buf + 4, dpkt->size);
memset(buf + 4 + dpkt->size / 2, 0,
buf_sz - 8 - dpkt->size);
}
#if 0
//
// old experimental code
//
if (1) {
// FIXME: need to detect dts
// copy original data for output
// FIXME: buf is sint
buf[0] = 0x72;
buf[1] = 0xF8;
buf[2] = 0x1F;
buf[3] = 0x4E;
buf[4] = 0x00;
switch (dpkt->size) {
case 512:
buf[5] = 0x0B;
break;
case 1024:
buf[5] = 0x0C;
break;
case 2048:
buf[5] = 0x0D;
break;
default:
Debug(3,
"codec/audio: dts sample burst not supported\n");
buf[5] = 0x00;
break;
}
buf[6] = (dpkt->size * 8);
buf[7] = (dpkt->size * 8) >> 8;
//buf[8] = 0x0B;
//buf[9] = 0x77;
//printf("%x %x\n", dpkt->data[0],dpkt->data[1]);
// swab?
memcpy(buf + 8, dpkt->data, dpkt->size);
memset(buf + 8 + dpkt->size, 0,
buf_sz - 8 - dpkt->size);
} else if (1) {
// FIXME: need to detect mp2
// FIXME: mp2 passthrough
// see softhddev.c version/layer
// 0x04 mpeg1 layer1
// 0x05 mpeg1 layer23
// 0x06 mpeg2 ext
// 0x07 mpeg2.5 layer 1
// 0x08 mpeg2.5 layer 2
// 0x09 mpeg2.5 layer 3
}
// DTS HD?
// True HD?
#endif
#endif
CodecReorderAudioFrame(buf, buf_sz,
audio_decoder->HwChannels);
AudioEnqueue(buf, buf_sz);
}
}
if (dpkt->size > l) {
Error(_("codec: error more than one frame data\n"));
}
}
index += n;
}
#if 1
// or av_free_packet, make no difference here
av_destruct_packet(spkt);
#endif
}
#else
#endif
/** /**
** Decode an audio packet. ** Decode an audio packet.
** **
@ -1301,17 +1021,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
*/ */
void CodecAudioFlushBuffers(AudioDecoder * decoder) void CodecAudioFlushBuffers(AudioDecoder * decoder)
{ {
#ifdef USE_AVPARSER
// FIXME: reset audio parser
if (decoder->AudioParser) {
av_parser_close(decoder->AudioParser);
decoder->AudioParser = NULL;
if (!(decoder->AudioParser =
av_parser_init(decoder->AudioCtx->codec_id))) {
Fatal(_("codec: can't init audio parser\n"));
}
}
#endif
avcodec_flush_buffers(decoder->AudioCtx); avcodec_flush_buffers(decoder->AudioCtx);
} }