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Code Issues Releases Activity

rtpenc: Always do the default initialization regardless of codecs

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This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).

Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö 2015-02-26 13:33:59 +02:00
parent 11edeaea32
commit f8c01257f9
1 changed files with 11 additions and 10 deletions
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21
libavformat/rtpenc.c
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@ -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;
}