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| author | Martin Storsjö <martin@martin.st> | 2010-01-21 17:10:13 +0000 |
|---|---|---|
| committer | Benoit Fouet <benoit.fouet@free.fr> | 2010-01-21 17:10:13 +0000 |
| commit | 073189917e4dfa0640e8180110b07e130471f37f (patch) | |
| tree | 23b5dd79d8f3f284951eda9b734c2dff851b1366 | |
| parent | 31f6e3c19e1e967d94a616c4fbc585460ded443c (diff) | |
| download | ffmpeg-073189917e4dfa0640e8180110b07e130471f37f.tar.gz | |
Make output-example.c handle AAC audio.
Patch by Martin Storsjö martin martin st
Originally committed as revision 21367 to svn://svn.ffmpeg.org/ffmpeg/trunk
| -rw-r--r-- | libavformat/output-example.c | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/libavformat/output-example.c b/libavformat/output-example.c index 17c97f79db..7eb3352b74 100644 --- a/libavformat/output-example.c +++ b/libavformat/output-example.c @@ -75,6 +75,11 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id) c->bit_rate = 64000; c->sample_rate = 44100; c->channels = 2; + + // some formats want stream headers to be separate + if(oc->oformat->flags & AVFMT_GLOBALHEADER) + c->flags |= CODEC_FLAG_GLOBAL_HEADER; + return st; } @@ -156,7 +161,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) pkt.size= avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples); - if (c->coded_frame->pts != AV_NOPTS_VALUE) + if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base); pkt.flags |= PKT_FLAG_KEY; pkt.stream_index= st->index; |