diff options
Diffstat (limited to 'codec.c')
| -rw-r--r-- | codec.c | 246 |
1 files changed, 204 insertions, 42 deletions
@@ -33,7 +33,7 @@ /** ** use av_parser to support insane dvb audio streams. */ -#define USE_AVPARSER +#define noUSE_AVPARSER /// compile with passthrough support (experimental) #define USE_PASSTHROUGH @@ -603,8 +603,10 @@ struct _audio_decoder_ AVCodec *AudioCodec; ///< audio codec 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 SampleRate; ///< current stream sample rate int Channels; ///< current stream channels @@ -697,10 +699,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, // we do not send complete frames 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->Channels = 0; audio_decoder->HwSampleRate = 0; @@ -719,10 +723,12 @@ void CodecAudioClose(AudioDecoder * audio_decoder) audio_resample_close(audio_decoder->ReSample); 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) { pthread_mutex_lock(&CodecLockMutex); avcodec_close(audio_decoder->AudioCtx); @@ -808,7 +814,7 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels) ** @param audio_decoder audio decoder data ** @param avpkt audio packet */ -void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) +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))); @@ -844,8 +850,8 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) av_init_packet(dpkt); n = av_parser_parse2(audio_decoder->AudioParser, audio_ctx, &dpkt->data, &dpkt->size, spkt->data + index, spkt->size - index, - !index ? (uint64_t) spkt->pts : AV_NOPTS_VALUE, - !index ? (uint64_t) spkt->dts : AV_NOPTS_VALUE, -1); + !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) { @@ -871,7 +877,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) #else #endif // Update audio clock - if ((uint64_t) dpkt->pts != AV_NOPTS_VALUE) { + if (dpkt->pts != (int64_t) AV_NOPTS_VALUE) { AudioSetClock(dpkt->pts); } // FIXME: must first play remainings bytes, than change and play new. @@ -1059,6 +1065,8 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) #else +#endif + /** ** Decode an audio packet. ** @@ -1074,61 +1082,205 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) AVCodecContext *audio_ctx; int index; -//#define spkt avpkt -#if 1 - 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 audio_ctx = audio_decoder->AudioCtx; index = 0; - while (spkt->size > index) { - int n; + while (avpkt->size > index) { + int l; int buf_sz; - AVPacket dpkt[1]; - - av_init_packet(dpkt); - dpkt->data = spkt->data + index; - dpkt->size = spkt->size - index; buf_sz = sizeof(buf); - n = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt); - if (n < 0) { // no audio frame could be decompressed - Error(_("codec: error audio data at %d\n"), index); + l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *)avpkt); + if (l == AVERROR(EAGAIN)) { + Error(_("codec: latm\n")); break; } -#ifdef DEBUG - Debug(4, "codec/audio: -> %d\n", buf_sz); - if ((unsigned)buf_sz > sizeof(buf)) { - abort(); + if (l < 0) { // no audio frame could be decompressed + Error(_("codec: error audio data at %d\n"), index); + break; } -#endif #ifdef notyetFF_API_OLD_DECODE_AUDIO // FIXME: ffmpeg git comeing int got_frame; - avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt); + avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt); #else #endif - // FIXME: see above, old code removed + // Update audio clock + if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) { + AudioSetClock(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!) + // FIXME: tell ac3 decoder to use downmix + if (audio_decoder->ReSample) { + audio_resample_close(audio_decoder->ReSample); + audio_decoder->ReSample = NULL; + } - index += n; - } + 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; + } -#if 1 - // or av_free_packet, make no difference here - av_destruct_packet(spkt); -#endif -} + // 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 + // avpkt is the original data + buf_sz = 6144; + if (buf_sz < avpkt->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 | (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); + } +#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 (avpkt->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] = (avpkt->size * 8); + buf[7] = (avpkt->size * 8) >> 8; + //buf[8] = 0x0B; + //buf[9] = 0x77; + //printf("%x %x\n", avpkt->data[0],avpkt->data[1]); + // swab? + memcpy(buf + 8, avpkt->data, avpkt->size); + memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->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 (avpkt->size > l) { + Error(_("codec: error more than one frame data\n")); + } + + index += l; + } +} /** ** Flush the audio decoder. @@ -1137,7 +1289,17 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) */ 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); } |