Fix audio decoding when AVPacket has >1 frame.

This was most obvious on FLAC.

diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c
index 861e753d41d43847d4ccc63f439e2323810cab4a..bee8dc1929e7932f8142f1d9a035d5177ff9fe21 100644 (file)
--- a/src/modules/avformat/producer_avformat.c
+++ b/src/modules/avformat/producer_avformat.c
@@ -1524,13 +1524,13 @@ 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 channels, int samples, double timecode, double fps )
+static int decode_audio( producer_avformat this, int *ignore, AVPacket pkt, int channels, int samples, double timecode, double fps )
 {
        // Fetch the audio_format
        AVFormatContext *context = this->audio_format;
 
        // Get the current stream index
-       int index = pkt->stream_index;
+       int index = pkt.stream_index;
 
        // Get codec context
        AVCodecContext *codec_context = this->audio_codec[ index ];
@@ -1543,8 +1543,8 @@ 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 ];
-       uint8_t *ptr = pkt->data;
-       int len = pkt->size;
+       uint8_t *ptr = pkt.data;
+       int len = pkt.size;
        int ret = 0;
 
        while ( ptr && ret >= 0 && len > 0 )
@@ -1553,7 +1553,7 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int
 
                // Decode the audio
 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(26<<8)+0))
-               ret = avcodec_decode_audio3( codec_context, decode_buffer, &data_size, pkt );
+               ret = avcodec_decode_audio3( codec_context, decode_buffer, &data_size, &pkt );
 #elif (LIBAVCODEC_VERSION_INT >= ((51<<16)+(29<<8)+0))
                ret = avcodec_decode_audio2( codec_context, decode_buffer, &data_size, ptr, len );
 #else
@@ -1565,8 +1565,8 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int
                        break;
                }
 
-               len -= ret;
-               ptr += ret;
+               pkt.size = len -= ret;
+               pkt.data = ptr += ret;
 
                // If decoded successfully
                if ( data_size > 0 )
@@ -1607,9 +1607,9 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int
        }
 
        // If we're behind, ignore this packet
-       if ( pkt->pts >= 0 )
+       if ( pkt.pts >= 0 )
        {
-               double current_pts = av_q2d( context->streams[ index ]->time_base ) * pkt->pts;
+               double current_pts = av_q2d( context->streams[ index ]->time_base ) * pkt.pts;
                int req_position = ( int )( timecode * fps + 0.5 );
                int int_position = ( int )( current_pts * fps + 0.5 );
                if ( context->start_time != AV_NOPTS_VALUE )
@@ -1743,7 +1743,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                                 ( this->audio_index == INT_MAX && context->streams[ pkt.stream_index ]->codec->codec_type == CODEC_TYPE_AUDIO ) ) )
                        {
                                int channels2 = this->audio_index == INT_MAX ? this->audio_codec[pkt.stream_index]->channels : *channels;
-                               ret = decode_audio( this, &ignore, &pkt, channels2, *samples, real_timecode, fps );
+                               ret = decode_audio( this, &ignore, pkt, channels2, *samples, real_timecode, fps );
                        }
                        av_free_packet( &pkt );