X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;ds=sidebyside;f=src%2Fmodules%2Favformat%2Fproducer_avformat.c;h=2d0be647bc3c5fe446c602d8cf597cd945eb70ee;hb=d591ca8dbc2658b2bc484043382a58058fd893b1;hp=798b005c578bed7ba53627eac27a24ffce8df795;hpb=beadb0ddd8b26ff6b1d029326a5d121a6bc35cab;p=mlt diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c index 798b005c..2d0be647 100644 --- a/src/modules/avformat/producer_avformat.c +++ b/src/modules/avformat/producer_avformat.c @@ -85,6 +85,7 @@ struct producer_avformat_s int max_channel; int max_frequency; unsigned int invalid_pts_counter; + double resample_factor; }; typedef struct producer_avformat_s *producer_avformat; @@ -154,6 +155,8 @@ mlt_producer producer_avformat_init( mlt_profile profile, char *file ) // Register our get_frame implementation producer->get_frame = producer_get_frame; + + this->resample_factor = 1.0; // Open the file if ( producer_open( this, profile, file ) != 0 ) @@ -695,7 +698,7 @@ static inline void convert_image( AVFrame *frame, uint8_t *buffer, int pix_fmt, AVPicture output; output.data[0] = buffer; output.data[1] = buffer + width * height; - output.data[2] = buffer + ( 3 * width * height ) / 2; + output.data[2] = buffer + ( 5 * width * height ) / 4; output.linesize[0] = width; output.linesize[1] = width >> 1; output.linesize[2] = width >> 1; @@ -739,7 +742,7 @@ static inline void convert_image( AVFrame *frame, uint8_t *buffer, int pix_fmt, AVPicture pict; pict.data[0] = buffer; pict.data[1] = buffer + width * height; - pict.data[2] = buffer + ( 3 * width * height ) / 2; + pict.data[2] = buffer + ( 5 * width * height ) / 4; pict.linesize[0] = width; pict.linesize[1] = width >> 1; pict.linesize[2] = width >> 1; @@ -1216,12 +1219,19 @@ static int video_codec_init( producer_avformat this, int index, mlt_properties p // Determine the fps first from the codec double source_fps = (double) this->video_codec->time_base.den / ( this->video_codec->time_base.num == 0 ? 1 : this->video_codec->time_base.num ); - - // If the muxer reports a frame rate different than the codec - double muxer_fps = av_q2d( stream->r_frame_rate ); - if ( source_fps != muxer_fps ) + + if ( mlt_properties_get( properties, "force_fps" ) ) + { + source_fps = mlt_properties_get_double( properties, "force_fps" ); + stream->time_base = av_d2q( source_fps, 255 ); + } + else + { + // If the muxer reports a frame rate different than the codec + double muxer_fps = av_q2d( stream->r_frame_rate ); // Choose the lesser - the wrong tends to be off by some multiple of 10 - source_fps = muxer_fps < source_fps ? muxer_fps : source_fps; + source_fps = FFMIN( source_fps, muxer_fps ); + } // We'll use fps if it's available if ( source_fps > 0 ) @@ -1363,7 +1373,7 @@ static int seek_audio( producer_avformat this, mlt_position position, double tim return paused; } -static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int samples, double timecode, double source_fps ) +static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int channels, int samples, double timecode, double source_fps ) { // Fetch the audio_format AVFormatContext *context = this->audio_format; @@ -1382,7 +1392,6 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int int16_t *decode_buffer = this->decode_buffer[ index ]; int audio_used = this->audio_used[ index ]; - int channels = codec_context->channels; uint8_t *ptr = pkt->data; int len = pkt->size; int ret = 0; @@ -1411,26 +1420,28 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int // If decoded successfully if ( data_size > 0 ) { + // Figure out how many samples will be needed after resampling + int convert_samples = data_size / codec_context->channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 ); + int samples_needed = lrint( this->resample_factor * convert_samples ); + // Resize audio buffer to prevent overflow - if ( audio_used * channels + data_size > this->audio_buffer_size[ index ] ) + if ( audio_used * channels + samples_needed > this->audio_buffer_size[ index ] ) { - mlt_pool_release( this->audio_buffer[ index ] ); - this->audio_buffer_size[ index ] = audio_used * channels * sizeof(int16_t) + data_size * 2; - audio_buffer = this->audio_buffer[ index ] = mlt_pool_alloc( this->audio_buffer_size[ index ] ); + this->audio_buffer_size[ index ] *= 2; + audio_buffer = this->audio_buffer[ index ] = mlt_pool_realloc( audio_buffer, this->audio_buffer_size[ index ] * sizeof(int16_t) ); } if ( resample ) { // Copy to audio buffer while resampling int16_t *source = decode_buffer; int16_t *dest = &audio_buffer[ audio_used * channels ]; - int convert_samples = data_size / channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 ); audio_used += audio_resample( resample, dest, source, convert_samples ); } else { // Straight copy to audio buffer - memcpy( &audio_buffer[ audio_used * channels ], decode_buffer, data_size ); - audio_used += data_size / channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 ); + memcpy( &audio_buffer[ audio_used * codec_context->channels ], decode_buffer, data_size ); + audio_used += convert_samples; } // Handle ignore @@ -1438,7 +1449,8 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int { *ignore -= 1; audio_used -= samples; - memmove( audio_buffer, &audio_buffer[ samples * channels ], audio_used * sizeof( int16_t ) ); + memmove( audio_buffer, &audio_buffer[ samples * (resample? channels : codec_context->channels) ], + audio_used * sizeof( int16_t ) ); } } } @@ -1486,7 +1498,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format // Fetch the audio_format AVFormatContext *context = this->audio_format; - + // Determine the tracks to use int index = this->audio_index; int index_max = this->audio_index + 1; @@ -1509,6 +1521,12 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format // Check for resample and create if necessary if ( codec_context->channels <= 2 ) { + // Determine by how much resampling will increase number of samples + double resample_factor = this->audio_index == INT_MAX ? 1 : (double) *channels / codec_context->channels; + resample_factor *= (double) *frequency / codec_context->sample_rate; + if ( resample_factor > this->resample_factor ) + this->resample_factor = resample_factor; + // Create the resampler #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0)) this->audio_resample[ index ] = av_audio_resample_init( @@ -1527,8 +1545,8 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format } // Check for audio buffer and create if necessary - this->audio_buffer[ index ] = mlt_pool_alloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) ); this->audio_buffer_size[ index ] = AVCODEC_MAX_AUDIO_FRAME_SIZE; + this->audio_buffer[ index ] = mlt_pool_alloc( this->audio_buffer_size[ index ] * sizeof( int16_t ) ); // Check for decoder buffer and create if necessary this->decode_buffer[ index ] = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) ); @@ -1559,7 +1577,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format // We only deal with audio from the selected audio index if ( ret >= 0 && pkt.data && pkt.size > 0 && ( pkt.stream_index == this->audio_index || ( this->audio_index == INT_MAX && context->streams[ pkt.stream_index ]->codec->codec_type == CODEC_TYPE_AUDIO ) ) ) - ret = decode_audio( this, &ignore, &pkt, *samples, real_timecode, source_fps ); + ret = decode_audio( this, &ignore, &pkt, *channels, *samples, real_timecode, source_fps ); av_free_packet( &pkt ); if ( this->audio_index == INT_MAX && ret >= 0 )