X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fdecavcodec.c;h=9d1474b3455ed6eed040fd028d9215ce4fac5f7e;hb=5d42c76946bc07b78a15aabcc6ebd5b68c07cc12;hp=dae2544c2146b22d9525ba039a0bbb04c7c454e8;hpb=5ab43fbc84cfa7ce16c98049127288574c639a5d;p=handbrake-jp%2Fhandbrake-jp-git.git diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index dae2544c..9d1474b3 100644 --- a/libhb/decavcodec.c +++ b/libhb/decavcodec.c @@ -233,7 +233,7 @@ static void decavcodecClose( hb_work_object_t * w ) } if ( pv->buffer ) { - free( pv->buffer ); + av_free( pv->buffer ); pv->buffer = NULL; } free( pv ); @@ -279,18 +279,20 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, pos = 0; while( pos < in->size ) { - len = av_parser_parse( pv->parser, pv->context, - &parser_output_buffer, &parser_output_buffer_len, - in->data + pos, in->size - pos, cur, cur ); + len = av_parser_parse2( pv->parser, pv->context, + &parser_output_buffer, &parser_output_buffer_len, + in->data + pos, in->size - pos, cur, cur, AV_NOPTS_VALUE ); out_size = 0; uncompressed_len = 0; if (parser_output_buffer_len) { + AVPacket avp; + av_init_packet( &avp ); + avp.data = parser_output_buffer; + avp.size = parser_output_buffer_len; + out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; - uncompressed_len = avcodec_decode_audio2( pv->context, bufaligned, - &out_size, - parser_output_buffer, - parser_output_buffer_len ); + uncompressed_len = avcodec_decode_audio3( pv->context, bufaligned, &out_size, &avp ); } if( out_size ) { @@ -339,7 +341,7 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, pv->pts_next = cur; - free( bufaligned ); + av_free( bufaligned ); return HB_WORK_OK; } @@ -410,14 +412,18 @@ static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, while ( pos < buf->size ) { - int len = av_parser_parse( parser, context, &pbuffer, &pbuffer_size, - buf->data + pos, buf->size - pos, - buf->start, buf->start ); + int len = av_parser_parse2( parser, context, &pbuffer, &pbuffer_size, + buf->data + pos, buf->size - pos, + buf->start, buf->start, AV_NOPTS_VALUE ); pos += len; if ( pbuffer_size > 0 ) { - len = avcodec_decode_audio2( context, (int16_t*)buffer, &out_size, - pbuffer, pbuffer_size ); + AVPacket avp; + av_init_packet( &avp ); + avp.data = pbuffer; + avp.size = pbuffer_size; + + len = avcodec_decode_audio3( context, (int16_t*)buffer, &out_size, &avp ); if ( len > 0 && context->sample_rate > 0 ) { info->bitrate = context->bit_rate; @@ -429,7 +435,7 @@ static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, } } } - free( buffer ); + av_free( buffer ); av_parser_close( parser ); hb_avcodec_close( context ); return ret; @@ -519,12 +525,12 @@ static void log_chapter( hb_work_private_t *pv, int chap_num, int64_t pts ) hb_chapter_t *c = hb_list_item( pv->job->title->list_chapter, chap_num - 1 ); if ( c && c->title ) { - hb_log( "%s: \"%s\" (%d) at frame %u time %lld", + hb_log( "%s: \"%s\" (%d) at frame %u time %"PRId64, pv->context->codec->name, c->title, chap_num, pv->nframes, pts ); } else { - hb_log( "%s: Chapter %d at frame %u time %lld", + hb_log( "%s: Chapter %d at frame %u time %"PRId64, pv->context->codec->name, chap_num, pv->nframes, pts ); } } @@ -548,13 +554,18 @@ static int decodeFrame( hb_work_private_t *pv, uint8_t *data, int size ) { int got_picture, oldlevel = 0; AVFrame frame; + AVPacket avp; if ( global_verbosity_level <= 1 ) { oldlevel = av_log_get_level(); av_log_set_level( AV_LOG_QUIET ); } - if ( avcodec_decode_video( pv->context, &frame, &got_picture, data, size ) < 0 ) + + av_init_packet( &avp ); + avp.data = data; + avp.size = size; + if ( avcodec_decode_video2( pv->context, &frame, &got_picture, &avp ) < 0 ) { ++pv->decode_errors; } @@ -681,8 +692,8 @@ static void decodeVideo( hb_work_private_t *pv, uint8_t *data, int size, do { uint8_t *pout; int pout_len; - int len = av_parser_parse( pv->parser, pv->context, &pout, &pout_len, - data + pos, size - pos, pts, dts ); + int len = av_parser_parse2( pv->parser, pv->context, &pout, &pout_len, + data + pos, size - pos, pts, dts, AV_NOPTS_VALUE ); pos += len; if ( pout_len > 0 ) @@ -908,13 +919,22 @@ static int decavcodecvInfo( hb_work_object_t *w, hb_work_info_t *info ) info->rate_base *= context->ticks_per_frame; } - /* Sometimes there's no pixel aspect set in the source. In that case, - assume a 1:1 PAR. Otherwise, preserve the source PAR. */ - info->pixel_aspect_width = context->sample_aspect_ratio.num ? - context->sample_aspect_ratio.num : 1; - info->pixel_aspect_height = context->sample_aspect_ratio.den ? - context->sample_aspect_ratio.den : 1; - + info->pixel_aspect_width = context->sample_aspect_ratio.num; + info->pixel_aspect_height = context->sample_aspect_ratio.den; + + /* Sometimes there's no pixel aspect set in the source ffmpeg context + * which appears to come from the video stream. In that case, + * try the pixel aspect in AVStream (which appears to come from + * the container). Else assume a 1:1 PAR. */ + if ( info->pixel_aspect_width == 0 || + info->pixel_aspect_height == 0 ) + { + AVStream *st = hb_ffmpeg_avstream( w->codec_param ); + info->pixel_aspect_width = st->sample_aspect_ratio.num ? + st->sample_aspect_ratio.num : 1; + info->pixel_aspect_height = st->sample_aspect_ratio.den ? + st->sample_aspect_ratio.den : 1; + } /* ffmpeg returns the Pixel Aspect Ratio (PAR). Handbrake wants the * Display Aspect Ratio so we convert by scaling by the Storage * Aspect Ratio (w/h). We do the calc in floating point to get the @@ -1133,9 +1153,14 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) pv->buffer = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); buffer = pv->buffer; } + + AVPacket avp; + av_init_packet( &avp ); + avp.data = data + pos; + avp.size = size - pos; + int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; - int len = avcodec_decode_audio2( context, buffer, &out_size, - data + pos, size - pos ); + int len = avcodec_decode_audio3( context, buffer, &out_size, &avp ); if ( len <= 0 ) { return; @@ -1160,7 +1185,7 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) NULL, 0 ); // get output buffer size (in 2-byte samples) then malloc a buffer out_size = ( out_size * 2 ) / isamp; - buffer = malloc( out_size ); + buffer = av_malloc( out_size ); // we're doing straight sample format conversion which behaves as if // there were only one channel. @@ -1194,7 +1219,7 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) // if we allocated a buffer for sample format conversion, free it if ( buffer != pv->buffer ) { - free( buffer ); + av_free( buffer ); } } }