diff options
Diffstat (limited to 'libhb/decavcodec.c')
| -rw-r--r-- | libhb/decavcodec.c | 83 |
1 files changed, 45 insertions, 38 deletions
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index 610ec1120..63b1c6dff 100644 --- a/libhb/decavcodec.c +++ b/libhb/decavcodec.c @@ -74,7 +74,7 @@ static int decavcodecBSInfo( hb_work_object_t *, const hb_buffer_t *, hb_work_in hb_work_object_t hb_decavcodec = { WORK_DECAVCODEC, - "MPGA decoder (libavcodec)", + "Audio decoder (libavcodec)", decavcodecInit, decavcodecWork, decavcodecClose, @@ -1327,23 +1327,6 @@ static void init_ffmpeg_context( hb_work_object_t *w ) pv->brokenByMicrosoft = 1; } -static void prepare_ffmpeg_buffer( hb_buffer_t * in ) -{ - // ffmpeg requires an extra 8 bytes of zero at the end of the buffer and - // will seg fault in odd, data dependent ways if it's not there. (my guess - // is this is a case of a local performance optimization creating a global - // performance degradation since all the time wasted by extraneous data - // copies & memory zeroing has to be huge compared to the minor reduction - // in inner-loop instructions this affords - modern cpus bottleneck on - // memory bandwidth not instruction bandwidth). - if ( in->size + FF_INPUT_BUFFER_PADDING_SIZE > in->alloc ) - { - // have to realloc to add the padding - hb_buffer_realloc( in, in->size + FF_INPUT_BUFFER_PADDING_SIZE ); - } - memset( in->data + in->size, 0, FF_INPUT_BUFFER_PADDING_SIZE ); -} - static int decavcodecviInit( hb_work_object_t * w, hb_job_t * job ) { @@ -1429,7 +1412,6 @@ static int decavcodecviWork( hb_work_object_t * w, hb_buffer_t ** buf_in, pv->new_chap = in->new_chap; pv->chap_time = pts >= 0? pts : pv->pts_next; } - prepare_ffmpeg_buffer( in ); decodeFrame( pv, in->data, in->size, in->sequence, in->start, in->renderOffset ); hb_buffer_close( &in ); *buf_out = link_buf_list( pv ); @@ -1475,7 +1457,7 @@ static hb_buffer_t * downmixAudio( else { buf = hb_buffer_init( nsamples * sizeof(float) ); - memcpy(buf->data, buffer, nsamples * sizeof(float) ); + memcpy( buf->data, buffer, nsamples * sizeof(float) ); } return buf; @@ -1524,7 +1506,34 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat pos += len; if( out_size > 0 ) { - // We require signed 16-bit ints for the output format. If + int isamp = av_get_bytes_per_sample( context->sample_fmt ); + nsamples = out_size / isamp; + double pts_next = pv->pts_next + nsamples * pv->duration; + + // DTS-HD can be passed through to mkv + if( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) + { + // Note that even though we are doing passthru, we had + // to decode so that we know the stop time and the + // pts of the next audio packet. + hb_buffer_t * buf; + + buf = hb_buffer_init( avp.size ); + memcpy( buf->data, avp.data, len ); + buf->start = pv->pts_next; + buf->stop = pts_next; + hb_list_add( pv->list, buf ); + + if ( hb_list_count( audio->priv.ff_audio_list ) == 0 ) + { + pv->pts_next = pts_next; + continue; + } + // Fall through and process the list of other audio + // pipelines that use this ffmpeg audio stream. + } + + // We require signed floats for the output format. If // we got something different convert it. if ( context->sample_fmt != AV_SAMPLE_FMT_FLT ) { @@ -1535,16 +1544,13 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat // anything more complicated than a one-for-one format // conversion we'd probably want to cache the converter // context in the pv. - int isamp; AVAudioConvert *ctx; - isamp = av_get_bytes_per_sample( context->sample_fmt ); ctx = av_audio_convert_alloc( AV_SAMPLE_FMT_FLT, 1, context->sample_fmt, 1, NULL, 0 ); // get output buffer size then malloc a buffer - nsamples = out_size / isamp; buffer = av_malloc( nsamples * sizeof(hb_sample_t) ); // we're doing straight sample format conversion which @@ -1557,16 +1563,11 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat av_audio_convert( ctx, obuf, ostride, ibuf, istride, nsamples ); av_audio_convert_free( ctx ); } - else - { - nsamples = out_size / sizeof(hb_sample_t); - } - hb_buffer_t * buf; - double pts_next = pv->pts_next + nsamples * pv->duration; - buf = downmixAudio( audio, pv, buffer, context->channels, nsamples ); - if ( buf ) + if ( !( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) ) { + hb_buffer_t * buf; + buf = downmixAudio( audio, pv, buffer, context->channels, nsamples ); buf->start = pv->pts_next; buf->stop = pts_next; hb_list_add( pv->list, buf ); @@ -1579,13 +1580,20 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat hb_work_private_t *ff_pv = hb_list_item( pv->ff_audio_list, i ); if ( ff_pv ) { - buf = downmixAudio( ff_audio, ff_pv, buffer, context->channels, nsamples ); - if ( buf ) + hb_buffer_t * buf; + + if ( !( ff_audio->config.out.codec & HB_ACODEC_PASS_FLAG ) ) + { + buf = downmixAudio( ff_audio, ff_pv, buffer, context->channels, nsamples ); + } + else { - buf->start = pv->pts_next; - buf->stop = pts_next; - hb_list_add( ff_pv->list, buf ); + buf = hb_buffer_init( avp.size ); + memcpy( buf->data, avp.data, len ); } + buf->start = pv->pts_next; + buf->stop = pts_next; + hb_list_add( ff_pv->list, buf ); } } pv->pts_next = pts_next; @@ -1639,7 +1647,6 @@ static int decavcodecaiWork( hb_work_object_t *w, hb_buffer_t **buf_in, { pv->pts_next = in->start; } - prepare_ffmpeg_buffer( in ); decodeAudio( w->audio, pv, in->data, in->size, pv->pts_next ); writeAudioFifos( w ); *buf_out = link_buf_list( pv ); |