Fix regressions in audio decoding.

diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c
index 454ba4e77f0295e500036a0d9a7ef4a1f9a17ab3..185515628106dc1d1925d8c1f23e2f870827a944 100644 (file)
--- a/src/modules/avformat/producer_avformat.c
+++ b/src/modules/avformat/producer_avformat.c
@@ -1924,7 +1924,7 @@ static int decode_audio( producer_avformat self, int *ignore, AVPacket pkt, int
 
        while ( ptr && ret >= 0 && len > 0 )
        {
-               int sizeof_sample = sample_bytes( codec_context );
+               int sizeof_sample = resample? sizeof( int16_t ) : sample_bytes( codec_context );
                int data_size = self->audio_buffer_size[ index ];
 
                // Decode the audio
@@ -1948,7 +1948,7 @@ static int decode_audio( producer_avformat self, int *ignore, AVPacket pkt, int
                if ( data_size > 0 )
                {
                        // Figure out how many samples will be needed after resampling
-                       int convert_samples = data_size / codec_context->channels / sizeof_sample;
+                       int convert_samples = data_size / codec_context->channels / sample_bytes( codec_context );
                        int samples_needed = self->resample_factor * convert_samples;
 
                        // Resize audio buffer to prevent overflow
@@ -2084,7 +2084,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                        }
 
                        // Check for audio buffer and create if necessary
-                       self->audio_buffer_size[ index ] = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+                       self->audio_buffer_size[ index ] = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof_sample;
                        self->audio_buffer[ index ] = mlt_pool_alloc( self->audio_buffer_size[ index ] );
 
                        // Check for decoder buffer and create if necessary
@@ -2141,6 +2141,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                }
 
                // Set some additional return values
+               *format = mlt_audio_s16;
                if ( self->audio_index != INT_MAX && !self->audio_resample[ self->audio_index ] )
                {
                        index = self->audio_index;
@@ -2154,7 +2155,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                {
                        // This only works if all audio tracks have the same sample format.
                        for ( index = 0; index < index_max; index++ )
-                               if ( self->audio_codec[ index ] )
+                               if ( self->audio_codec[ index ] && !self->audio_resample[ index ] )
                                {
                                        *format = self->audio_codec[ index ]->sample_fmt == SAMPLE_FMT_S32 ? mlt_audio_s32
                                                : self->audio_codec[ index ]->sample_fmt == SAMPLE_FMT_FLT ? mlt_audio_float