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| author | Anton Khirnov <anton@khirnov.net> | 2014-11-15 16:48:59 +0100 |
|---|---|---|
| committer | Anton Khirnov <anton@khirnov.net> | 2015-12-12 21:19:08 +0100 |
| commit | 1f008f34d5b2b5f6217521747e7acfe3efc0e666 (patch) | |
| tree | f7387b8a95cdd2b688470b5f77d9cac63a68a89a | |
| parent | 458e53f51fc75d08df884f8e9eb3d7ded23e97b3 (diff) | |
| download | ffmpeg-1f008f34d5b2b5f6217521747e7acfe3efc0e666.tar.gz | |
rmenc: do not use AVCodecContext.frame_size
It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.
| -rw-r--r-- | libavformat/rmenc.c | 7 |
1 files changed, 5 insertions, 2 deletions
diff --git a/libavformat/rmenc.c b/libavformat/rmenc.c index ad2c17d529..bbfd550a3b 100644 --- a/libavformat/rmenc.c +++ b/libavformat/rmenc.c @@ -185,9 +185,10 @@ static int rv10_write_header(AVFormatContext *ctx, if (stream->enc->codec_type == AVMEDIA_TYPE_AUDIO) { int coded_frame_size, fscode, sample_rate; + int frame_size = av_get_audio_frame_duration(stream->enc, 0); sample_rate = stream->enc->sample_rate; coded_frame_size = (stream->enc->bit_rate * - stream->enc->frame_size) / (8 * sample_rate); + frame_size) / (8 * sample_rate); /* audio codec info */ avio_write(s, ".ra", 3); avio_w8(s, 0xfd); @@ -315,6 +316,7 @@ static int rm_write_header(AVFormatContext *s) for(n=0;n<s->nb_streams;n++) { AVStream *st = s->streams[n]; + int frame_size; s->streams[n]->id = n; codec = s->streams[n]->codec; @@ -327,7 +329,8 @@ static int rm_write_header(AVFormatContext *s) switch(codec->codec_type) { case AVMEDIA_TYPE_AUDIO: rm->audio_stream = stream; - stream->frame_rate = (float)codec->sample_rate / (float)codec->frame_size; + frame_size = av_get_audio_frame_duration(codec, 0); + stream->frame_rate = (float)codec->sample_rate / (float)frame_size; /* XXX: dummy values */ stream->packet_max_size = 1024; stream->nb_packets = 0; |