Hi, On Tue, Jul 27, 2010 at 10:48 PM, Josh Allmann <[email protected]> wrote: > Re-sending, due to this hunk: > > @@ -154,6 +156,11 @@ static int rtp_write_header(AVFormatContext *s1) > } > case CODEC_ID_AAC: > s->num_frames = 0; > + case CODEC_ID_VORBIS: > + case CODEC_ID_THEORA: > + if(!s->max_frames_per_packet) s->max_frames_per_packet = 15; > + s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); > + s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length > default: > if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { > av_set_pts_info(st, 32, 1, st->codec->sample_rate); > > Apparently AAC and AMR both fall through to the default case. Whether > that's intentional, I don't know, so I moved my stuff to minimize > behavioral changes to existing code.
You could use goto default (where you would normally use "break") so that this works for all codecs without randomness. Ronald _______________________________________________ FFmpeg-soc mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc
