X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fdecavcodec.c;h=9d1474b3455ed6eed040fd028d9215ce4fac5f7e;hb=033e32de9c380f54c7d1362a3979da205ebc3a29;hp=d02ca9d1d69520437aba282568e7665b29e1e17f;hpb=1ac109932e4e44d7d195cfdfc6591b7199a92d9e;p=handbrake-jp%2Fhandbrake-jp-git.git diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index d02ca9d1..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 ); @@ -252,7 +252,7 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, hb_work_private_t * pv = w->private_data; hb_buffer_t * in = *buf_in, * buf, * last = NULL; int pos, len, out_size, i, uncompressed_len; - short buffer[AVCODEC_MAX_AUDIO_FRAME_SIZE]; + short* bufaligned; uint64_t cur; unsigned char *parser_output_buffer; int parser_output_buffer_len; @@ -275,21 +275,24 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, cur = ( in->start < 0 )? pv->pts_next : in->start; + bufaligned = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); 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) { - out_size = sizeof(buffer); - uncompressed_len = avcodec_decode_audio2( pv->context, buffer, - &out_size, - parser_output_buffer, - 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_audio3( pv->context, bufaligned, &out_size, &avp ); } if( out_size ) { @@ -316,7 +319,7 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, pv->context->sample_rate; cur = buf->stop; - s16 = buffer; + s16 = bufaligned; fl32 = (float *) buf->data; for( i = 0; i < out_size / 2; i++ ) { @@ -338,6 +341,7 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, pv->pts_next = cur; + av_free( bufaligned ); return HB_WORK_OK; } @@ -401,25 +405,25 @@ static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, AVCodecParserContext *parser = av_parser_init( codec->id ); AVCodecContext *context = avcodec_alloc_context(); hb_avcodec_open( context, codec ); -#if defined( SYS_CYGWIN ) - uint8_t *buffer = memalign(16, AVCODEC_MAX_AUDIO_FRAME_SIZE); -#else - uint8_t *buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); -#endif + uint8_t *buffer = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; unsigned char *pbuffer; int pos = 0, pbuffer_size; 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; @@ -431,7 +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; @@ -521,12 +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 ); } } @@ -550,13 +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; } @@ -683,8 +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 ) @@ -910,13 +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 @@ -1132,26 +1150,17 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) int16_t *buffer = pv->buffer; if ( buffer == NULL ) { - // XXX ffmpeg bug workaround - // malloc a buffer for the audio decode. On an x86, ffmpeg - // uses mmx/sse instructions on this buffer without checking - // that it's 16 byte aligned and this will cause an abort if - // the buffer is allocated on our stack. Rather than doing - // complicated, machine dependent alignment here we use the - // fact that malloc returns an aligned pointer on most architectures. - - #if defined( SYS_CYGWIN ) - // Cygwin's malloc doesn't appear to return 16-byte aligned memory so use memalign instead. - pv->buffer = memalign(16, AVCODEC_MAX_AUDIO_FRAME_SIZE); - #else - pv->buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); - #endif - + 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; @@ -1176,7 +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. @@ -1210,7 +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 ); } } }