X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fdecavcodec.c;h=8b09afcf77dac79aaf135d80368e1b44661037b2;hb=c593146bf3fab6290c71cbbb974e0a756e43f5e0;hp=81ea4996b456b123d099a158347229c7bfdfaba8;hpb=5846d37d995f4b03815c58b736305d6e7f8c51ce;p=handbrake-jp%2Fhandbrake-jp-git.git diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index 81ea4996..8b09afcf 100644 --- a/libhb/decavcodec.c +++ b/libhb/decavcodec.c @@ -60,10 +60,8 @@ */ #include "hb.h" - -#include "libavcodec/avcodec.h" -#include "libavformat/avformat.h" -#include "libswscale/swscale.h" +#include "hbffmpeg.h" +#include "libavcodec/audioconvert.h" static int decavcodecInit( hb_work_object_t *, hb_job_t * ); static int decavcodecWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** ); @@ -190,11 +188,12 @@ static int decavcodecInit( hb_work_object_t * w, hb_job_t * job ) /*XXX*/ if ( codec_id == 0 ) codec_id = CODEC_ID_MP2; + codec = avcodec_find_decoder( codec_id ); pv->parser = av_parser_init( codec_id ); pv->context = avcodec_alloc_context(); - avcodec_open( pv->context, codec ); + hb_avcodec_open( pv->context, codec ); return 0; } @@ -226,7 +225,7 @@ static void decavcodecClose( hb_work_object_t * w ) } if ( pv->context && pv->context->codec ) { - avcodec_close( pv->context ); + hb_avcodec_close( pv->context ); } if ( pv->list ) { @@ -268,6 +267,12 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in, *buf_out = NULL; + if ( in->start < -1 && pv->pts_next <= 0 ) + { + // discard buffers that start before video time 0 + return HB_WORK_OK; + } + cur = ( in->start < 0 )? pv->pts_next : in->start; pos = 0; @@ -355,10 +360,23 @@ static int decavcodecInfo( hb_work_object_t *w, hb_work_info_t *info ) return 0; } +static const int chan2layout[] = { + HB_INPUT_CH_LAYOUT_MONO, // We should allow no audio really. + HB_INPUT_CH_LAYOUT_MONO, + HB_INPUT_CH_LAYOUT_STEREO, + HB_INPUT_CH_LAYOUT_2F1R, + HB_INPUT_CH_LAYOUT_2F2R, + HB_INPUT_CH_LAYOUT_3F2R, + HB_INPUT_CH_LAYOUT_4F2R, + HB_INPUT_CH_LAYOUT_STEREO, + HB_INPUT_CH_LAYOUT_STEREO, +}; + static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, hb_work_info_t *info ) { hb_work_private_t *pv = w->private_data; + int ret = 0; memset( info, 0, sizeof(*info) ); @@ -371,18 +389,52 @@ static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, // now we just return dummy values if there's a codec that will handle it. AVCodec *codec = avcodec_find_decoder( w->codec_param? w->codec_param : CODEC_ID_MP2 ); - if ( codec ) + if ( ! codec ) { - static char codec_name[64]; + // there's no ffmpeg codec for this audio type - give up + return -1; + } - info->name = strncpy( codec_name, codec->name, sizeof(codec_name)-1 ); - info->bitrate = 384000; - info->rate = 48000; - info->rate_base = 1; - info->channel_layout = HB_INPUT_CH_LAYOUT_STEREO; - return 1; + static char codec_name[64]; + info->name = strncpy( codec_name, codec->name, sizeof(codec_name)-1 ); + + 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 + 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 ); + pos += len; + if ( pbuffer_size > 0 ) + { + len = avcodec_decode_audio2( context, (int16_t*)buffer, &out_size, + pbuffer, pbuffer_size ); + if ( len > 0 && context->sample_rate > 0 ) + { + info->bitrate = context->bit_rate; + info->rate = context->sample_rate; + info->rate_base = 1; + info->channel_layout = chan2layout[context->channels & 7]; + ret = 1; + break; + } + } } - return -1; + free( buffer ); + av_parser_close( parser ); + hb_avcodec_close( context ); + return ret; } /* ------------------------------------------------------------- @@ -496,13 +548,22 @@ static void flushDelayQueue( hb_work_private_t *pv ) static int decodeFrame( hb_work_private_t *pv, uint8_t *data, int size ) { - int got_picture; + int got_picture, oldlevel = 0; AVFrame frame; + 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 ) { ++pv->decode_errors; } + if ( global_verbosity_level <= 1 ) + { + av_log_set_level( oldlevel ); + } if( got_picture ) { // ffmpeg makes it hard to attach a pts to a frame. if the MPEG ES @@ -672,17 +733,91 @@ static int decavcodecvInit( hb_work_object_t * w, hb_job_t * job ) pv->context->opaque = pv; pv->context->get_buffer = get_frame_buf; - AVCodec *codec = avcodec_find_decoder( codec_id ); + return 0; +} + +static int next_hdr( hb_buffer_t *in, int offset ) +{ + uint8_t *dat = in->data; + uint16_t last2 = 0xffff; + for ( ; in->size - offset > 1; ++offset ) + { + if ( last2 == 0 && dat[offset] == 0x01 ) + // found an mpeg start code + return offset - 2; + + last2 = ( last2 << 8 ) | dat[offset]; + } + + return -1; +} + +static int find_hdr( hb_buffer_t *in, int offset, uint8_t hdr_type ) +{ + if ( in->size - offset < 4 ) + // not enough room for an mpeg start code + return -1; + + for ( ; ( offset = next_hdr( in, offset ) ) >= 0; ++offset ) + { + if ( in->data[offset+3] == hdr_type ) + // found it + break; + } + return offset; +} + +static int setup_extradata( hb_work_object_t *w, hb_buffer_t *in ) +{ + hb_work_private_t *pv = w->private_data; // we can't call the avstream funcs but the read_header func in the // AVInputFormat may set up some state in the AVContext. In particular // vc1t_read_header allocates 'extradata' to deal with header issues // related to Microsoft's bizarre engineering notions. We alloc a chunk // of space to make vc1 work then associate the codec with the context. - pv->context->extradata_size = 32; - pv->context->extradata = av_malloc(pv->context->extradata_size); - avcodec_open( pv->context, codec ); + if ( w->codec_param != CODEC_ID_VC1 ) + { + // we haven't been inflicted with M$ - allocate a little space as + // a marker and return success. + pv->context->extradata_size = 16; + pv->context->extradata = av_malloc(pv->context->extradata_size); + return 0; + } + // find the start and and of the sequence header + int shdr, shdr_end; + if ( ( shdr = find_hdr( in, 0, 0x0f ) ) < 0 ) + { + // didn't find start of seq hdr + return 1; + } + if ( ( shdr_end = next_hdr( in, shdr + 4 ) ) < 0 ) + { + shdr_end = in->size; + } + shdr_end -= shdr; + + // find the start and and of the entry point header + int ehdr, ehdr_end; + if ( ( ehdr = find_hdr( in, 0, 0x0e ) ) < 0 ) + { + // didn't find start of entry point hdr + return 1; + } + if ( ( ehdr_end = next_hdr( in, ehdr + 4 ) ) < 0 ) + { + ehdr_end = in->size; + } + ehdr_end -= ehdr; + + // found both headers - allocate an extradata big enough to hold both + // then copy them into it. + pv->context->extradata_size = shdr_end + ehdr_end; + pv->context->extradata = av_malloc(pv->context->extradata_size + 8); + memcpy( pv->context->extradata, in->data + shdr, shdr_end ); + memcpy( pv->context->extradata + shdr_end, in->data + ehdr, ehdr_end ); + memset( pv->context->extradata + shdr_end + ehdr_end, 0, 8); return 0; } @@ -705,6 +840,27 @@ static int decavcodecvWork( hb_work_object_t * w, hb_buffer_t ** buf_in, return HB_WORK_DONE; } + // if this is the first frame open the codec (we have to wait for the + // first frame because of M$ VC1 braindamage). + if ( pv->context->extradata_size == 0 ) + { + if ( setup_extradata( w, in ) ) + { + // we didn't find the headers needed to set up extradata. + // the codec will abort if we open it so just free the buf + // and hope we eventually get the info we need. + hb_buffer_close( &in ); + return HB_WORK_OK; + } + AVCodec *codec = avcodec_find_decoder( w->codec_param ); + // There's a mis-feature in ffmpeg that causes the context to be + // incorrectly initialized the 1st time avcodec_open is called. + // If you close it and open a 2nd time, it finishes the job. + hb_avcodec_open( pv->context, codec ); + hb_avcodec_close( pv->context ); + hb_avcodec_open( pv->context, codec ); + } + if( in->start >= 0 ) { pts = in->start; @@ -739,6 +895,12 @@ static int decavcodecvInfo( hb_work_object_t *w, hb_work_info_t *info ) info->rate = 27000000; info->rate_base = (int64_t)context->time_base.num * 27000000LL / context->time_base.den; + if ( context->ticks_per_frame > 1 ) + { + // for ffmpeg 0.5 & later, the H.264 & MPEG-2 time base is + // field rate rather than frame rate so convert back to frames. + 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. */ @@ -804,7 +966,7 @@ static void init_ffmpeg_context( hb_work_object_t *w ) if ( ! pv->context->codec ) { AVCodec *codec = avcodec_find_decoder( pv->context->codec_id ); - avcodec_open( pv->context, codec ); + hb_avcodec_open( pv->context, codec ); } // set up our best guess at the frame duration. // the frame rate in the codec is usually bogus but it's sometimes @@ -969,7 +1131,14 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) // 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. - pv->buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE ); + + #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 + buffer = pv->buffer; } int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; @@ -982,6 +1151,35 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) pos += len; if( out_size > 0 ) { + // We require signed 16-bit ints for the output format. If + // we got something different convert it. + if ( context->sample_fmt != SAMPLE_FMT_S16 ) + { + // Note: av_audio_convert seems to be a work-in-progress but + // looks like it will eventually handle general audio + // mixdowns which would allow us much more flexibility + // in handling multichannel audio in HB. If we were doing + // anything more complicated than a one-for-one format + // conversion we'd probably want to cache the converter + // context in the pv. + int isamp = av_get_bits_per_sample_format( context->sample_fmt ) / 8; + AVAudioConvert *ctx = av_audio_convert_alloc( SAMPLE_FMT_S16, 1, + context->sample_fmt, 1, + NULL, 0 ); + // get output buffer size (in 2-byte samples) then malloc a buffer + out_size = ( out_size * 2 ) / isamp; + buffer = malloc( out_size ); + + // we're doing straight sample format conversion which behaves as if + // there were only one channel. + const void * const ibuf[6] = { pv->buffer }; + void * const obuf[6] = { buffer }; + const int istride[6] = { isamp }; + const int ostride[6] = { 2 }; + + av_audio_convert( ctx, obuf, ostride, ibuf, istride, out_size >> 1 ); + av_audio_convert_free( ctx ); + } hb_buffer_t *buf = hb_buffer_init( 2 * out_size ); // convert from bytes to total samples @@ -1000,6 +1198,12 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size ) fl32[i] = buffer[i]; } hb_list_add( pv->list, buf ); + + // if we allocated a buffer for sample format conversion, free it + if ( buffer != pv->buffer ) + { + free( buffer ); + } } } } @@ -1016,6 +1220,14 @@ static int decavcodecaiWork( hb_work_object_t *w, hb_buffer_t **buf_in, } hb_work_private_t *pv = w->private_data; + + if ( (*buf_in)->start < -1 && pv->pts_next <= 0 ) + { + // discard buffers that start before video time 0 + *buf_out = NULL; + return HB_WORK_OK; + } + if ( ! pv->context ) { init_ffmpeg_context( w );