diff options
| -rw-r--r-- | contrib/ffmpeg/module.defs | 2 | ||||
| -rw-r--r-- | libhb/decavcodec.c | 2 | ||||
| -rw-r--r-- | libhb/stream.c | 19 |
3 files changed, 14 insertions, 9 deletions
diff --git a/contrib/ffmpeg/module.defs b/contrib/ffmpeg/module.defs index 9222b8ccc..304183ffe 100644 --- a/contrib/ffmpeg/module.defs +++ b/contrib/ffmpeg/module.defs @@ -1,7 +1,7 @@ $(eval $(call import.MODULE.defs,FFMPEG,ffmpeg,BZIP2 FAAD2 ZLIB)) $(eval $(call import.CONTRIB.defs,FFMPEG)) -FFMPEG.FETCH.url = http://download.m0k.org/handbrake/contrib/ffmpeg-r25374.tar.bz2 +FFMPEG.FETCH.url = http://download.m0k.org/handbrake/contrib/ffmpeg-r25689.tar.bz2 FFMPEG.CONFIGURE.deps = FFMPEG.CONFIGURE.env = diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index c40ce49ec..2ccac589a 100644 --- a/libhb/decavcodec.c +++ b/libhb/decavcodec.c @@ -1386,7 +1386,7 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat // anything more complicated than a one-for-one format // conversion we'd probably want to cache the converter // context in the pv. - int isamp = av_get_bits_per_sample_format( context->sample_fmt ) / 8; + int isamp = av_get_bits_per_sample_fmt( context->sample_fmt ) / 8; AVAudioConvert *ctx = av_audio_convert_alloc( SAMPLE_FMT_S16, 1, context->sample_fmt, 1, NULL, 0 ); diff --git a/libhb/stream.c b/libhb/stream.c index 204baa7b0..32d407735 100644 --- a/libhb/stream.c +++ b/libhb/stream.c @@ -58,11 +58,7 @@ static const stream2codec_t st2codec[256] = { st(0x0e, N, 0, 0, "ISO 13818-1 auxiliary"), st(0x0f, A, HB_ACODEC_MPGA, CODEC_ID_AAC, "ISO 13818-7 AAC Audio"), st(0x10, V, WORK_DECAVCODECV, CODEC_ID_MPEG4, "MPEG4"), -#if defined(OLD_LATM_PATCH) st(0x11, A, HB_ACODEC_MPGA, CODEC_ID_AAC_LATM, "MPEG4 LATM AAC"), -#else - st(0x11, N, 0, 0, "MPEG4 LATM AAC"), -#endif st(0x12, U, 0, 0, "MPEG4 generic"), st(0x14, N, 0, 0, "ISO 13818-6 DSM-CC download"), @@ -2997,6 +2993,7 @@ static void add_ffmpeg_audio( hb_title_t *title, hb_stream_t *stream, int id ) { AVStream *st = stream->ffmpeg_ic->streams[id]; AVCodecContext *codec = st->codec; + AVMetadataTag *tag; int layout; // scan will ignore any audio without a bitrate. Since we've already @@ -3031,7 +3028,9 @@ static void add_ffmpeg_audio( hb_title_t *title, hb_stream_t *stream, int id ) audio->config.in.channel_layout = layout; } - set_audio_description( audio, lang_for_code2( st->language ) ); + tag = av_metadata_get( st->metadata, "language", NULL, 0 ); + set_audio_description( audio, + lang_for_code2( tag ? tag->value : "und" ) ); hb_list_add( title->list_audio, audio ); } @@ -3199,7 +3198,11 @@ static void add_ffmpeg_subtitle( hb_title_t *title, hb_stream_t *stream, int id return; } - iso639_lang_t *language = lang_for_code2( st->language ); + AVMetadataTag *tag; + iso639_lang_t *language; + + tag = av_metadata_get( st->metadata, "language", NULL, 0 ); + language = lang_for_code2( tag ? tag->value : "und" ); strcpy( subtitle->lang, language->eng_name ); strncpy( subtitle->iso639_2, language->iso639_2, 4 ); @@ -3330,6 +3333,7 @@ static hb_title_t *ffmpeg_title_scan( hb_stream_t *stream ) for( i = 0; i < ic->nb_chapters; i++ ) if( ( m = ic->chapters[i] ) != NULL ) { + AVMetadataTag *tag; hb_chapter_t * chapter; chapter = calloc( sizeof( hb_chapter_t ), 1 ); chapter->index = i+1; @@ -3338,7 +3342,8 @@ static hb_title_t *ffmpeg_title_scan( hb_stream_t *stream ) chapter->hours = chapter->duration / 90000 / 3600; chapter->minutes = ( ( chapter->duration / 90000 ) % 3600 ) / 60; chapter->seconds = ( chapter->duration / 90000 ) % 60; - strcpy( chapter->title, m->title ); + tag = av_metadata_get( m->metadata, "title", NULL, 0 ); + strcpy( chapter->title, tag ? tag->value : "" ); hb_deep_log( 2, "Added chapter %i, name='%s', dur=%"PRIu64", (%02i:%02i:%02i)", chapter->index, chapter->title, chapter->duration, chapter->hours, |