Completely flush audio in avformat consumer.

diff --git a/src/modules/avformat/consumer_avformat.c b/src/modules/avformat/consumer_avformat.c
index 811306595a6617e27c1db04c0221ed6cac1de214..964da6ca9e6920b37e4614dddeeadd0532444208 100644 (file)
--- a/src/modules/avformat/consumer_avformat.c
+++ b/src/modules/avformat/consumer_avformat.c
@@ -1974,10 +1974,11 @@ static void *consumer_thread( void *arg )
                        pkt.data = audio_outbuf;
                        pkt.size = 0;
 
-                       if ( fifo &&
-                               ( channels * audio_input_nb_samples < sample_fifo_used( fifo ) / sample_bytes ) )
+                       if ( fifo && sample_fifo_used( fifo ) > 0 )
                        {
-                               sample_fifo_fetch( fifo, audio_buf_1, channels * audio_input_nb_samples * sample_bytes );
+                               // Drain the MLT FIFO
+                               int samples = FFMIN( FFMIN( channels * audio_input_nb_samples, sample_fifo_used( fifo ) / sample_bytes ), AUDIO_ENCODE_BUFFER_SIZE );
+                               sample_fifo_fetch( fifo, audio_buf_1, samples * sample_bytes );
                                void* p = audio_buf_1;
 #if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
                                if ( c->sample_fmt == AV_SAMPLE_FMT_FLTP )
@@ -1991,7 +1992,7 @@ static void *consumer_thread( void *arg )
 #endif
 #if LIBAVCODEC_VERSION_MAJOR >= 55
                                pkt.size = audio_outbuf_size;
-                               audio_avframe->nb_samples = audio_input_nb_samples;
+                               audio_avframe->nb_samples = FFMAX( samples / channels, audio_input_nb_samples );
                                avcodec_fill_audio_frame( audio_avframe, c->channels, c->sample_fmt,
                                        (const uint8_t*) p, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                int got_packet = 0;
@@ -2001,32 +2002,37 @@ static void *consumer_thread( void *arg )
                                else if ( !got_packet )
                                        pkt.size = 0;
 #else
-                               c->frame_size = audio_input_nb_samples;
+                               c->frame_size = FFMAX( samples / channels, audio_input_nb_samples );
                                pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, p );
 #endif
 #if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
                                if ( p != audio_buf_1 )
                                        mlt_pool_release( p );
 #endif
+                               mlt_log_debug( MLT_CONSUMER_SERVICE( consumer ), "flushing audio size %d\n", pkt.size );
                        }
-                       if ( pkt.size <= 0 ) {
+                       else
+                       {
+                               // Drain the codec
+                               if ( pkt.size <= 0 ) {
 #if LIBAVCODEC_VERSION_MAJOR >= 55
-                               pkt.size = audio_outbuf_size;
-                               int got_packet = 0;
-                               int ret = avcodec_encode_audio2( c, &pkt, NULL, &got_packet );
-                               if ( ret < 0 )
-                                       pkt.size = ret;
-                               else if ( !got_packet )
-                                       pkt.size = 0;
+                                       pkt.size = audio_outbuf_size;
+                                       int got_packet = 0;
+                                       int ret = avcodec_encode_audio2( c, &pkt, NULL, &got_packet );
+                                       if ( ret < 0 )
+                                               pkt.size = ret;
+                                       else if ( !got_packet )
+                                               pkt.size = 0;
 #else
-                               pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, NULL );
-                               pkt.pts = c->coded_frame? c->coded_frame->pts : AV_NOPTS_VALUE;
-                               pkt.flags |= PKT_FLAG_KEY;
+                                       pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, NULL );
+                                       pkt.pts = c->coded_frame? c->coded_frame->pts : AV_NOPTS_VALUE;
+                                       pkt.flags |= PKT_FLAG_KEY;
 #endif
+                               }
+                               mlt_log_debug( MLT_CONSUMER_SERVICE( consumer ), "flushing audio size %d\n", pkt.size );
+                               if ( pkt.size <= 0 )
+                                       break;
                        }
-                       mlt_log_debug( MLT_CONSUMER_SERVICE( consumer ), "flushing audio size %d\n", pkt.size );
-                       if ( pkt.size <= 0 )
-                               break;
 
                        // Write the compressed frame in the media file
                        if ( pkt.pts != AV_NOPTS_VALUE )