diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index ae1d3e04c6..9a68c6d698 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -176,11 +176,17 @@ static int rtp_write_header(AVFormatContext *s1) } } - avpriv_set_pts_info(st, 32, 1, 90000); + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); + } else { + avpriv_set_pts_info(st, 32, 1, 90000); + } + s->buf_ptr = s->buf; switch(st->codec->codec_id) { case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; + avpriv_set_pts_info(st, 32, 1, 90000); break; case AV_CODEC_ID_MPEG1VIDEO: case AV_CODEC_ID_MPEG2VIDEO: @@ -224,7 +230,7 @@ static int rtp_write_header(AVFormatContext *s1) s->max_frames_per_packet = 15; s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); s->num_frames = 0; - goto defaultcase; + break; case AV_CODEC_ID_ADPCM_G722: /* Due to a historical error, the clock rate for G722 in RTP is * 8000, even if the sample rate is 16000. See RFC 3551. */ @@ -249,7 +255,7 @@ static int rtp_write_header(AVFormatContext *s1) s->max_frames_per_packet = 1; s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, s->max_payload_size / st->codec->block_align); - goto defaultcase; + break; case AV_CODEC_ID_AMR_NB: case AV_CODEC_ID_AMR_WB: if (!s->max_frames_per_packet) @@ -268,18 +274,13 @@ static int rtp_write_header(AVFormatContext *s1) goto fail; } s->num_frames = 0; - goto defaultcase; + break; case AV_CODEC_ID_AAC: s->num_frames = 0; if (!s->max_frames_per_packet) s->max_frames_per_packet = 5; - goto defaultcase; + break; default: -defaultcase: - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); - } - s->buf_ptr = s->buf; break; }