// avformat header files
#include <avformat.h>
+//
+// This structure should be extended and made globally available in mlt
+//
+
typedef struct
{
int16_t *buffer;
int size;
int used;
+ double time;
+ int frequency;
+ int channels;
}
*sample_fifo, sample_fifo_s;
-sample_fifo sample_fifo_init( )
+sample_fifo sample_fifo_init( int frequency, int channels )
+{
+ sample_fifo this = calloc( 1, sizeof( sample_fifo_s ) );
+ this->frequency = frequency;
+ this->channels = channels;
+ return this;
+}
+
+// sample_fifo_clear and check are temporarily aborted (not working as intended)
+
+void sample_fifo_clear( sample_fifo this, double time )
+{
+ int words = ( float )( time - this->time ) * this->frequency * this->channels;
+ if ( ( int )( ( float )time * 100 ) < ( int )( ( float )this->time * 100 ) && this->used > words && words > 0 )
+ {
+ memmove( this->buffer, &this->buffer[ words ], ( this->used - words ) * sizeof( int16_t ) );
+ this->used -= words;
+ this->time = time;
+ }
+ else if ( ( int )( ( float )time * 100 ) != ( int )( ( float )this->time * 100 ) )
+ {
+ this->used = 0;
+ this->time = time;
+ }
+}
+
+void sample_fifo_check( sample_fifo this, double time )
{
- return calloc( 1, sizeof( sample_fifo_s ) );
+ if ( this->used == 0 )
+ {
+ if ( ( int )( ( float )time * 100 ) < ( int )( ( float )this->time * 100 ) )
+ this->time = time;
+ }
}
void sample_fifo_append( sample_fifo this, int16_t *samples, int count )
this->used -= count;
memmove( this->buffer, &this->buffer[ count ], this->used * sizeof( int16_t ) );
+ this->time += ( double )count / this->channels / this->frequency;
+
return count;
}
mlt_properties_set( properties, "target", arg );
// sample and frame queue
- mlt_properties_set_data( properties, "sample_fifo", sample_fifo_init( ), 0, ( mlt_destructor )sample_fifo_close, NULL );
mlt_properties_set_data( properties, "frame_queue", mlt_deque_init( ), 0, ( mlt_destructor )mlt_deque_close, NULL );
// Set avformat defaults (all lifted from ffmpeg.c)
int samples = 0;
// AVFormat audio buffer and frame size
- int audio_outbuf_size = 2 * 128 * 1024;
+ int audio_outbuf_size = 10000;
uint8_t *audio_outbuf = av_malloc( audio_outbuf_size );
int audio_input_frame_size = 0;
fmt = guess_format( "mpeg", NULL, NULL );
// We need a filename - default to stdout?
- if ( filename == NULL )
+ if ( filename == NULL || !strcmp( filename, "" ) )
filename = "pipe:";
// Get the codec ids selected
// Open the output file, if needed
if ( !( fmt->flags & AVFMT_NOFILE ) )
{
- if (url_fopen(&oc->pb, filename, URL_RDWR) < 0)
+ if (url_fopen(&oc->pb, filename, URL_WRONLY) < 0)
{
fprintf(stderr, "Could not open '%s'\n", filename);
mlt_properties_set_int( properties, "running", 0 );
{
samples = mlt_sample_calculator( fps, frequency, count );
mlt_frame_get_audio( frame, &pcm, &aud_fmt, &frequency, &channels, &samples );
+
+ // Create the fifo if we don't have one
+ if ( fifo == NULL )
+ {
+ fifo = sample_fifo_init( frequency, channels );
+ mlt_properties_set_data( properties, "sample_fifo", fifo, 0, ( mlt_destructor )sample_fifo_close, NULL );
+ }
+
+ // Append the samples
sample_fifo_append( fifo, pcm, samples * channels );
total_time += ( samples * 1000000 ) / frequency;
}
{
// Compute current audio and video time
if (audio_st)
- audio_pts = (double)audio_st->pts.val * oc->pts_num / oc->pts_den;
+ audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
else
audio_pts = 0.0;
if (video_st)
- video_pts = (double)video_st->pts.val * oc->pts_num / oc->pts_den;
+ video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
else
video_pts = 0.0;
{
if ( channels * audio_input_frame_size < sample_fifo_used( fifo ) )
{
- int out_size;
AVCodecContext *c;
+ AVPacket pkt;
+ av_init_packet( &pkt );
c = &audio_st->codec;
sample_fifo_fetch( fifo, buffer, channels * audio_input_frame_size );
- out_size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, buffer );
-
+ pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, buffer );
// Write the compressed frame in the media file
- if (av_write_frame(oc, audio_st->index, audio_outbuf, out_size) != 0)
+ pkt.pts= c->coded_frame->pts;
+ pkt.flags |= PKT_FLAG_KEY;
+ pkt.stream_index= audio_st->index;
+ pkt.data= audio_outbuf;
+
+ if ( av_write_frame( oc, &pkt ) != 0)
fprintf(stderr, "Error while writing audio frame\n");
}
else
if (oc->oformat->flags & AVFMT_RAWPICTURE)
{
// raw video case. The API will change slightly in the near future for that
- ret = av_write_frame(oc, video_st->index, (uint8_t *)output, sizeof(AVPicture));
+ AVPacket pkt;
+ av_init_packet(&pkt);
+
+ pkt.flags |= PKT_FLAG_KEY;
+ pkt.stream_index= video_st->index;
+ pkt.data= (uint8_t *)output;
+ pkt.size= sizeof(AVPicture);
+
+ ret = av_write_frame(oc, &pkt);
}
else
{
// If zero size, it means the image was buffered
if (out_size != 0)
{
- // write the compressed frame in the media file
- // XXX: in case of B frames, the pts is not yet valid
- ret = av_write_frame( oc, video_st->index, video_outbuf, out_size );
+ AVPacket pkt;
+ av_init_packet( &pkt );
+
+ pkt.pts= c->coded_frame->pts;
+ if(c->coded_frame->key_frame)
+ pkt.flags |= PKT_FLAG_KEY;
+ pkt.stream_index= video_st->index;
+ pkt.data= video_outbuf;
+ pkt.size= out_size;
+
+ // write the compressed frame in the media file
+ ret = av_write_frame(oc, &pkt);
}
}
frame_count++;
if ( real_time_output && frames % 25 == 0 )
{
long passed = time_difference( &ante );
- long pending = ( ( ( long )sample_fifo_used( fifo ) * 1000 ) / frequency ) * 1000;
- passed -= pending;
+ if ( fifo != NULL )
+ {
+ long pending = ( ( ( long )sample_fifo_used( fifo ) * 1000 ) / frequency ) * 1000;
+ passed -= pending;
+ }
if ( passed < total_time )
{
long total = ( total_time - passed );
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