return payload_type;
}
-static inline uint32_t decode_be32(const uint8_t *p)
-{
- return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
-}
-
-static inline uint64_t decode_be64(const uint8_t *p)
-{
- return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
-}
-
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
if (buf[1] != 200)
return -1;
- s->last_rtcp_ntp_time = decode_be64(buf + 8);
+ s->last_rtcp_ntp_time = AV_RB64(buf + 8);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
- s->last_rtcp_timestamp = decode_be32(buf + 16);
+ s->last_rtcp_timestamp = AV_RB32(buf + 16);
return 0;
}
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
case CODEC_ID_H264:
- st->need_parsing = 1;
+ st->need_parsing = AVSTREAM_PARSE_FULL;
break;
default:
break;
infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
/* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
- In my test, the faad decoder doesnt behave correctly when sending each AU one by one
+ In my test, the FAAD decoder does not behave correctly when sending each AU one by one
but does when sending the whole as one big packet... */
infos->au_headers[0].size = 0;
infos->au_headers[0].index = 0;
return -1;
}
payload_type = buf[1] & 0x7f;
- seq = (buf[2] << 8) | buf[3];
- timestamp = decode_be32(buf + 4);
- ssrc = decode_be32(buf + 8);
+ seq = AV_RB16(buf + 2);
+ timestamp = AV_RB32(buf + 4);
+ ssrc = AV_RB32(buf + 8);
/* store the ssrc in the RTPDemuxContext */
s->ssrc = ssrc;
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
- h = decode_be32(buf);
+ h = AV_RB32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
- h = decode_be32(buf);
+ h = AV_RB32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
max_packet_size = url_fget_max_packet_size(&s1->pb);
if (max_packet_size <= 12)
- return AVERROR_IO;
+ return AVERROR(EIO);
s->max_payload_size = max_packet_size - 12;
switch(st->codec->codec_id) {
return 0;
}
-static int rtp_write_trailer(AVFormatContext *s1)
-{
- // RTPDemuxContext *s = s1->priv_data;
- return 0;
-}
-
AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet,
- rtp_write_trailer,
};