#define PIX_FMT_YUYV422 PIX_FMT_YUV422
#endif
-#define AUDIO_ENCODE_BUFFER_SIZE (48000 * 2)
+#define MAX_AUDIO_STREAMS (8)
+#define AUDIO_ENCODE_BUFFER_SIZE (48000 * 2 * MAX_AUDIO_STREAMS)
//
// This structure should be extended and made globally available in mlt
static void *consumer_thread( void *arg );
static void consumer_close( mlt_consumer this );
-/** Initialise the dv consumer.
+/** Initialise the consumer.
*/
mlt_consumer consumer_avformat_init( mlt_profile profile, char *arg )
/** Add an audio output stream
*/
-static AVStream *add_audio_stream( mlt_consumer this, AVFormatContext *oc, int codec_id )
+static AVStream *add_audio_stream( mlt_consumer this, AVFormatContext *oc, int codec_id, int channels )
{
// Get the properties
mlt_properties properties = MLT_CONSUMER_PROPERTIES( this );
// Create a new stream
- AVStream *st = av_new_stream( oc, 1 );
+ AVStream *st = av_new_stream( oc, oc->nb_streams );
// If created, then initialise from properties
if ( st != NULL )
// Set parameters controlled by MLT
c->sample_rate = mlt_properties_get_int( properties, "frequency" );
c->time_base = ( AVRational ){ 1, c->sample_rate };
- c->channels = mlt_properties_get_int( properties, "channels" );
+ c->channels = channels;
if ( mlt_properties_get( properties, "alang" ) != NULL )
strncpy( st->language, mlt_properties_get( properties, "alang" ), sizeof( st->language ) );
mlt_properties properties = MLT_CONSUMER_PROPERTIES( this );
// Create a new stream
- AVStream *st = av_new_stream( oc, 0 );
+ AVStream *st = av_new_stream( oc, oc->nb_streams );
if ( st != NULL )
{
mlt_image_format img_fmt = mlt_image_yuv422;
// For receiving audio samples back from the fifo
- int16_t *buffer = av_malloc( AUDIO_ENCODE_BUFFER_SIZE );
+ int16_t *audio_buf_1 = av_malloc( AUDIO_ENCODE_BUFFER_SIZE );
+ int16_t *audio_buf_2 = NULL;
int count = 0;
// Allocate the context
#endif
// Streams
- AVStream *audio_st = NULL;
AVStream *video_st = NULL;
+ AVStream *audio_st[ MAX_AUDIO_STREAMS ];
+ int is_audio_initialized = 0;
// Time stamps
double audio_pts = 0;
double video_pts = 0;
- // Loop variable
- int i;
-
- // Frames despatched
+ // Frames dispatched
long int frames = 0;
long int total_time = 0;
int audio_codec_id;
int video_codec_id;
+ // Misc
+ char key[27];
+ mlt_properties frame_meta_properties = mlt_properties_new();
+
+ // Initialize audio_st
+ int i = MAX_AUDIO_STREAMS;
+ while ( i-- )
+ audio_st[i] = NULL;
+
// Check for user selected format first
if ( format != NULL )
#if LIBAVFORMAT_VERSION_INT < ((52<<16)+(45<<8)+0)
oc->oformat = fmt;
snprintf( oc->filename, sizeof(oc->filename), "%s", filename );
- // Add audio and video streams
+ // Get the first frame
+ frame = mlt_consumer_rt_frame( this );
+
+ // Add audio and video streams
if ( video_codec_id != CODEC_ID_NONE )
video_st = add_video_stream( this, oc, video_codec_id );
if ( audio_codec_id != CODEC_ID_NONE )
- audio_st = add_audio_stream( this, oc, audio_codec_id );
+ {
+ i = 0;
+ if ( frame )
+ i = mlt_properties_get_int( MLT_FRAME_PROPERTIES(frame), "meta.map.audio.0.channels" );
+ audio_st[0] = add_audio_stream( this, oc, audio_codec_id, i ? i : channels );
+ }
// Set the parameters (even though we have none...)
if ( av_set_parameters(oc, NULL) >= 0 )
if ( video_st && !open_video( oc, video_st ) )
video_st = NULL;
- if ( audio_st )
+ if ( audio_st[0] ) // Add audio streams as needed later
{
- audio_input_frame_size = open_audio( oc, audio_st, audio_outbuf_size );
+ audio_input_frame_size = open_audio( oc, audio_st[0], audio_outbuf_size );
if ( !audio_input_frame_size )
- audio_st = NULL;
+ audio_st[0] = NULL;
}
// Open the output file, if needed
}
}
- // Write the stream header, if any
- if ( mlt_properties_get_int( properties, "running" ) )
+ // Write the stream header now if there are no audio streams.
+ // Otherwise, we wait until we possibly add additional audio streams below.
+ if ( ! audio_st[0] && mlt_properties_get_int( properties, "running" ) )
av_write_header( oc );
}
else
output = alloc_picture( video_st->codec->pix_fmt, width, height );
// Last check - need at least one stream
- if ( audio_st == NULL && video_st == NULL )
+ if ( !audio_st[0] && !video_st )
mlt_properties_set_int( properties, "running", 0 );
// Get the starting time (can ignore the times above)
while( mlt_properties_get_int( properties, "running" ) &&
( !terminated || ( video_st && mlt_deque_count( queue ) ) ) )
{
- // Get the frame
- frame = mlt_consumer_rt_frame( this );
-
// Check that we have a frame to work with
if ( frame != NULL )
{
- // Increment frames despatched
+ // Increment frames dispatched
frames ++;
// Default audio args
terminated = terminate_on_pause && mlt_properties_get_double( frame_properties, "_speed" ) == 0.0;
// Get audio and append to the fifo
- if ( !terminated && audio_st )
+ if ( !terminated && audio_st[0] )
{
samples = mlt_sample_calculator( fps, frequency, count ++ );
mlt_frame_get_audio( frame, (void**) &pcm, &aud_fmt, &frequency, &channels, &samples );
+ // Add audio streams based on channel mapping
+ if ( ! is_audio_initialized )
+ {
+ int j = 0;
+ for ( i = 0; i < MAX_AUDIO_STREAMS && j < channels; i++ )
+ {
+ sprintf( key, "meta.map.audio.%d.channels", i );
+ int map_channels = mlt_properties_get_int( frame_properties, key );
+ if ( !map_channels && i > 0 )
+ map_channels = channels - j;
+ if ( map_channels )
+ {
+ if ( i && ( audio_st[i] = add_audio_stream( this, oc, audio_codec_id, map_channels ) )
+ && ( ! open_audio( oc, audio_st[i], audio_outbuf_size ) ) )
+ audio_st[i] = NULL;
+ }
+ j += i ? map_channels : audio_st[0]->codec->channels;
+ }
+ mlt_properties_pass( frame_meta_properties, frame_properties, "meta.map.audio." );
+
+ // Write the stream header now that we have all streams
+ av_write_header( oc );
+ is_audio_initialized = 1;
+ }
+
// Create the fifo if we don't have one
if ( fifo == NULL )
{
while ( 1 )
{
// Write interleaved audio and video frames
- if ( !video_st || ( video_st && audio_st && audio_pts < video_pts ) )
+ if ( !video_st || ( video_st && audio_st[0] && audio_pts < video_pts ) )
{
if ( ( video_st && terminated ) || ( channels * audio_input_frame_size ) < sample_fifo_used( fifo ) )
{
- AVCodecContext *c = audio_st->codec;
- AVPacket pkt;
int n = FFMIN( FFMIN( channels * audio_input_frame_size, sample_fifo_used( fifo ) ), AUDIO_ENCODE_BUFFER_SIZE );
-
if ( n > 0 )
- sample_fifo_fetch( fifo, buffer, n );
+ sample_fifo_fetch( fifo, audio_buf_1, n );
else
- memset( buffer, 0, AUDIO_ENCODE_BUFFER_SIZE );
-
- av_init_packet( &pkt );
- pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, buffer );
-
- // Write the compressed frame in the media file
- if ( c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE )
+ memset( audio_buf_1, 0, AUDIO_ENCODE_BUFFER_SIZE );
+ samples = n / channels;
+
+ // Extract the audio channels according to channel mapping
+ int j = 0; // channel offset into interleaved source buffer
+ int map_start = 0, map_channels = 0;
+ for ( i = 0; i < MAX_AUDIO_STREAMS && audio_st[i] && j < channels; i++ )
{
- pkt.pts = av_rescale_q( c->coded_frame->pts, c->time_base, audio_st->time_base );
- mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "audio pkt pts %lld frame pts %lld", pkt.pts, c->coded_frame->pts );
- }
- pkt.flags |= PKT_FLAG_KEY;
- pkt.stream_index= audio_st->index;
- pkt.data= audio_outbuf;
+ AVStream *stream = audio_st[i];
+ AVCodecContext *codec = stream->codec;
+ AVPacket pkt;
- if ( pkt.size > 0 )
- if ( av_interleaved_write_frame( oc, &pkt ) != 0 )
- mlt_log_error( MLT_CONSUMER_SERVICE( this ), "error writing audio frame\n" );
+ av_init_packet( &pkt );
- mlt_log_debug( MLT_CONSUMER_SERVICE( this ), " frame_size %d\n", c->frame_size );
- if ( audio_codec_id == CODEC_ID_VORBIS )
- audio_pts = (double)c->coded_frame->pts * av_q2d( audio_st->time_base );
- else
- audio_pts = (double)audio_st->pts.val * av_q2d( audio_st->time_base );
+ // Get the audio channel mapping
+ sprintf( key, "%d.channels", i );
+ map_channels = mlt_properties_get_int( frame_meta_properties, key );
+ sprintf( key, "%d.start", i );
+ if ( mlt_properties_get( frame_meta_properties, key ) )
+ map_start = mlt_properties_get_int( frame_meta_properties, key );
+ else
+ map_start = -1;
+
+ // Optimized for no channel remapping.
+ if ( !map_channels && map_start == -1 )
+ {
+ // Encode the audio.
+ pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, audio_buf_1 );
+ }
+ else
+ {
+ int ch; // channel offset into interleaved dest buffer
+
+ // If last map..channels not specific, use the remaining channels.
+ if ( !map_channels )
+ map_channels = channels - j;
+
+ // Clear or allocate the audio buffer.
+ if ( !audio_buf_2 )
+ audio_buf_2 = av_mallocz( AUDIO_ENCODE_BUFFER_SIZE );
+ else
+ memset( audio_buf_2, 0, AUDIO_ENCODE_BUFFER_SIZE );
+
+ // Interleave the audio buffer with the #channels for this stream.
+ for ( ch = 0; ch < map_channels && j < channels; ch++, j++ )
+ {
+ int16_t *src = audio_buf_1 + ( map_start > -1 ? ( map_start + ch ) : j );
+ int16_t *dest = audio_buf_2 + ch;
+ int s = samples + 1;
+
+ while ( --s ) {
+ *dest = *src;
+ dest += map_channels;
+ src += channels;
+ }
+ }
+
+ // Encode the audio.
+ pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, audio_buf_2 );
+ }
+
+ // Write the compressed frame in the media file
+ if ( codec->coded_frame && codec->coded_frame->pts != AV_NOPTS_VALUE )
+ {
+ pkt.pts = av_rescale_q( codec->coded_frame->pts, codec->time_base, stream->time_base );
+ mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "audio stream %d pkt pts %lld frame pts %lld",
+ stream->index, pkt.pts, codec->coded_frame->pts );
+ }
+ pkt.flags |= PKT_FLAG_KEY;
+ pkt.stream_index = stream->index;
+ pkt.data = audio_outbuf;
+
+ if ( pkt.size > 0 && stream->pts.den )
+ {
+ if ( av_interleaved_write_frame( oc, &pkt ) )
+ mlt_log_error( MLT_CONSUMER_SERVICE( this ), "error writing audio frame %d\n", frames - 1 );
+ }
+
+ mlt_log_debug( MLT_CONSUMER_SERVICE( this ), " frame_size %d\n", codec->frame_size );
+ if ( i == 0 )
+ {
+ if ( audio_codec_id == CODEC_ID_VORBIS )
+ audio_pts = (double)codec->coded_frame->pts * av_q2d( stream->time_base );
+ else
+ audio_pts = (double)stream->pts.val * av_q2d( stream->time_base );
+ }
+ }
}
else
{
break;
}
}
- if ( audio_st )
- mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "audio pts %lld (%f) ", audio_st->pts.val, audio_pts );
+ if ( audio_st[0] )
+ mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "audio pts %lld (%f) ", audio_st[0]->pts.val, audio_pts );
if ( video_st )
mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "video pts %lld (%f) ", video_st->pts.val, video_pts );
mlt_log_debug( MLT_CONSUMER_SERVICE( this ), "\n" );
nanosleep( &t, NULL );
}
}
+
+ // Get the next frame
+ frame = mlt_consumer_rt_frame( this );
}
#ifdef FLUSH
}
#endif
+ // Write the trailer, if any
+ av_write_trailer( oc );
+
// XXX ugly hack to prevent x264 from crashing on multi-threaded encoding
int pass = mlt_properties_get_int( properties, "pass" );
int thread_count = mlt_properties_get_int( properties, "threads" );
// close each codec
if ( video_st && !multithreaded_x264 )
close_video(oc, video_st);
- if (audio_st)
- close_audio(oc, audio_st);
-
- // Write the trailer, if any
- av_write_trailer(oc);
+ for ( i = 0; i < MAX_AUDIO_STREAMS && audio_st[i]; i++ )
+ close_audio( oc, audio_st[i] );
// Free the streams
- for(i = 0; i < oc->nb_streams; i++)
- av_freep(&oc->streams[i]);
+ for ( i = 0; i < oc->nb_streams; i++ )
+ av_freep( &oc->streams[i] );
// Close the output file
- if (!(fmt->flags & AVFMT_NOFILE))
+ if ( !( fmt->flags & AVFMT_NOFILE ) )
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(0<<8)+0)
- url_fclose(oc->pb);
+ url_fclose( oc->pb );
#else
- url_fclose(&oc->pb);
+ url_fclose( &oc->pb );
#endif
// Clean up input and output frames
av_free( input->data[0] );
av_free( input );
av_free( video_outbuf );
- av_free( buffer );
+ av_free( audio_buf_1 );
+ av_free( audio_buf_2 );
// Free the stream
- av_free(oc);
+ av_free( oc );
// Just in case we terminated on pause
mlt_properties_set_int( properties, "running", 0 );
mlt_consumer_stopped( this );
+ mlt_properties_close( frame_meta_properties );
if ( pass == 2 )
{
projects / mlt / commitdiff