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| author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-02-27 03:09:44 -0500 |
|---|---|---|
| committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-03-05 13:08:16 -0500 |
| commit | 620b88a302d522f71a6a95aff500cf7bd8520043 (patch) | |
| tree | c4406898b26abfd517ee0b3609243ba34cdea467 /libavformat/swfenc.c | |
| parent | 14aecc50fae6466fe9c16992a2929fb86497f237 (diff) | |
| download | ffmpeg-620b88a302d522f71a6a95aff500cf7bd8520043.tar.gz | |
swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size
This way we can do stream copy without having the demuxer wait until
frame_size has been set.
Diffstat (limited to 'libavformat/swfenc.c')
| -rw-r--r-- | libavformat/swfenc.c | 6 |
1 files changed, 1 insertions, 5 deletions
diff --git a/libavformat/swfenc.c b/libavformat/swfenc.c index f4494cd63a..82ec7ff9bc 100644 --- a/libavformat/swfenc.c +++ b/libavformat/swfenc.c @@ -187,10 +187,6 @@ static int swf_write_header(AVFormatContext *s) AVCodecContext *enc = s->streams[i]->codec; if (enc->codec_type == AVMEDIA_TYPE_AUDIO) { if (enc->codec_id == CODEC_ID_MP3) { - if (!enc->frame_size) { - av_log(s, AV_LOG_ERROR, "audio frame size not set\n"); - return -1; - } swf->audio_enc = enc; swf->audio_fifo= av_fifo_alloc(AUDIO_FIFO_SIZE); if (!swf->audio_fifo) @@ -452,7 +448,7 @@ static int swf_write_audio(AVFormatContext *s, } av_fifo_generic_write(swf->audio_fifo, buf, size, NULL); - swf->sound_samples += enc->frame_size; + swf->sound_samples += av_get_audio_frame_duration(enc, size); /* if audio only stream make sure we add swf frames */ if (!swf->video_enc) |