Fix muxing vorbis into webm/mkv with libavcodec < 55.

[mlt] / src / modules / avformat / consumer_avformat.c

diff --git a/src/modules/avformat/consumer_avformat.c b/src/modules/avformat/consumer_avformat.c
index 295e7fc65ce6411dcb07827aac8fdc37b89c17d8..33b3a93f983e05597ae0830be5ae0503cfc41924 100644 (file)
--- a/src/modules/avformat/consumer_avformat.c
+++ b/src/modules/avformat/consumer_avformat.c
@@ -1178,7 +1178,7 @@ static void *consumer_thread( void *arg )
        int count = 0;
 
        // Allocate the context
-       AVFormatContext *oc = avformat_alloc_context( );
+       AVFormatContext *oc = NULL;
 
        // Streams
        AVStream *video_st = NULL;
@@ -1232,6 +1232,22 @@ static void *consumer_thread( void *arg )
        if ( filename == NULL || !strcmp( filename, "" ) )
                filename = "pipe:";
 
+#if LIBAVUTIL_VERSION_INT >= ((53<<16)+(2<<8)+0)
+       avformat_alloc_output_context2( &oc, fmt, format, filename );
+#else
+       oc = avformat_alloc_context( );
+       oc->oformat = fmt;
+       snprintf( oc->filename, sizeof(oc->filename), "%s", filename );
+
+       if ( oc->oformat && oc->oformat->priv_class && !oc->priv_data && oc->oformat->priv_data_size ) {
+               oc->priv_data = av_mallocz( oc->oformat->priv_data_size );
+               if ( oc->priv_data ) {
+                       *(const AVClass**)oc->priv_data = oc->oformat->priv_class;
+                       av_opt_set_defaults( oc->priv_data );
+               }
+       }
+#endif
+
        // Get the codec ids selected
        audio_codec_id = fmt->audio_codec;
        video_codec_id = fmt->video_codec;
@@ -1310,9 +1326,6 @@ static void *consumer_thread( void *arg )
                }
        }
 
-       oc->oformat = fmt;
-       snprintf( oc->filename, sizeof(oc->filename), "%s", filename );
-
        // Get a frame now, so we can set some AVOptions from properties.
        frame = mlt_consumer_rt_frame( consumer );
 
@@ -1480,7 +1493,7 @@ static void *consumer_thread( void *arg )
        if ( video_st )
                converted_avframe = alloc_picture( video_st->codec->pix_fmt, width, height );
 
-#if LIBAVCODEC_VERSION_MAJOR >= 55
+#if LIBAVCODEC_VERSION_MAJOR >= 54
        // Allocate audio AVFrame
        if ( audio_st[0] )
        {
@@ -1613,11 +1626,13 @@ static void *consumer_thread( void *arg )
                                                        else if ( codec->sample_fmt == AV_SAMPLE_FMT_U8P )
                                                                p = interleaved_to_planar( samples, channels, p, sizeof( uint8_t ) );
 #endif
-#if LIBAVCODEC_VERSION_MAJOR >= 55
+#if LIBAVCODEC_VERSION_MAJOR >= 54
                                                        audio_avframe->nb_samples = FFMAX( samples, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                                        if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                                                audio_avframe->pts = synth_audio_pts;
                                                        synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                                        avcodec_fill_audio_frame( audio_avframe, codec->channels, codec->sample_fmt,
                                                                (const uint8_t*) p, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                                        int got_packet = 0;
@@ -1703,11 +1718,13 @@ static void *consumer_thread( void *arg )
                                                                        dest_offset += current_channels;
                                                                }
                                                        }
-#if LIBAVCODEC_VERSION_MAJOR >= 55
+#if LIBAVCODEC_VERSION_MAJOR >= 54
                                                        audio_avframe->nb_samples = FFMAX( samples, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                                        if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                                                audio_avframe->pts = synth_audio_pts;
                                                        synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                                        avcodec_fill_audio_frame( audio_avframe, codec->channels, codec->sample_fmt,
                                                                (const uint8_t*) audio_buf_2, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                                        int got_packet = 0;
@@ -2013,12 +2030,14 @@ static void *consumer_thread( void *arg )
                                else if ( c->sample_fmt == AV_SAMPLE_FMT_U8P )
                                        p = interleaved_to_planar( audio_input_nb_samples, channels, p, sizeof( uint8_t ) );
 #endif
-#if LIBAVCODEC_VERSION_MAJOR >= 55
+#if LIBAVCODEC_VERSION_MAJOR >= 54
                                pkt.size = audio_outbuf_size;
                                audio_avframe->nb_samples = FFMAX( samples / channels, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                        audio_avframe->pts = synth_audio_pts;
                                synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                avcodec_fill_audio_frame( audio_avframe, c->channels, c->sample_fmt,
                                        (const uint8_t*) p, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                int got_packet = 0;
@@ -2041,7 +2060,7 @@ static void *consumer_thread( void *arg )
                        {
                                // Drain the codec
                                if ( pkt.size <= 0 ) {
-#if LIBAVCODEC_VERSION_MAJOR >= 55
+#if LIBAVCODEC_VERSION_MAJOR >= 54
                                        pkt.size = audio_outbuf_size;
                                        int got_packet = 0;
                                        int ret = avcodec_encode_audio2( c, &pkt, NULL, &got_packet );
@@ -2200,6 +2219,18 @@ on_fatal_error:
                free( full );
                free( cwd );
                remove( "x264_2pass.log.temp" );
+
+               // Recent versions of libavcodec/x264 support passlogfile and need cleanup if specified.
+               if ( !mlt_properties_get( properties, "_logfilename" ) &&
+                     mlt_properties_get( properties, "passlogfile" ) )
+               {
+                       file = mlt_properties_get( properties, "passlogfile" );
+                       remove( file );
+                       full = malloc( strlen( file ) + strlen( ".mbtree" ) + 1 );
+                       sprintf( full, "%s.mbtree", file );
+                       remove( full );
+                       free( full );
+               }
        }
 
        while ( ( frame = mlt_deque_pop_back( queue ) ) )