videolan/ffmpeg
1
Fork 0
mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-09-30 00:30:24 +02:00
Code Releases Activity

Properly set RTP and NTP timestamps in RTCP SR packets

Browse Source 
Originally committed as revision 10468 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni 2007-09-10 06:58:19 +00:00
parent c64a871234
commit 1b31b02ed1
1 changed files with 14 additions and 7 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

21
libavformat/rtp.c
View File

@ -739,6 +739,7 @@ static int rtp_write_header(AVFormatContext *s1)
s->timestamp = s->base_timestamp; s->timestamp = s->base_timestamp;
s->ssrc = 0; /* FIXME: was random(), what should this be? */ s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1; s->first_packet = 1;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
max_packet_size = url_fget_max_packet_size(&s1->pb); max_packet_size = url_fget_max_packet_size(&s1->pb);
if (max_packet_size <= 12) if (max_packet_size <= 12)
@ -762,6 +763,9 @@ static int rtp_write_header(AVFormatContext *s1)
s->buf_ptr = s->buf; s->buf_ptr = s->buf;
break; break;
default: default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf; s->buf_ptr = s->buf;
break; break;
} }
@ -773,15 +777,22 @@ static int rtp_write_header(AVFormatContext *s1)
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{ {
RTPDemuxContext *s = s1->priv_data; RTPDemuxContext *s = s1->priv_data;
uint32_t rtp_ts;
#if defined(DEBUG) #if defined(DEBUG)
printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
#endif #endif
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
s1->streams[0]->time_base) + s->base_timestamp;
put_byte(&s1->pb, (RTP_VERSION << 6)); put_byte(&s1->pb, (RTP_VERSION << 6));
put_byte(&s1->pb, 200); put_byte(&s1->pb, 200);
put_be16(&s1->pb, 6); /* length in words - 1 */ put_be16(&s1->pb, 6); /* length in words - 1 */
put_be32(&s1->pb, s->ssrc); put_be32(&s1->pb, s->ssrc);
put_be64(&s1->pb, ntp_time); put_be32(&s1->pb, ntp_time / 1000000);
put_be32(&s1->pb, s->timestamp); put_be32(&s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
put_be32(&s1->pb, rtp_ts);
put_be32(&s1->pb, s->packet_count); put_be32(&s1->pb, s->packet_count);
put_be32(&s1->pb, s->octet_count); put_be32(&s1->pb, s->octet_count);
put_flush_packet(&s1->pb); put_flush_packet(&s1->pb);
@ -956,7 +967,6 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
RTPDemuxContext *s = s1->priv_data; RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0]; AVStream *st = s1->streams[0];
int rtcp_bytes; int rtcp_bytes;
int64_t ntp_time;
int size= pkt->size; int size= pkt->size;
uint8_t *buf1= pkt->data; uint8_t *buf1= pkt->data;
@ -968,10 +978,7 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN; RTCP_TX_RATIO_DEN;
if (s->first_packet || rtcp_bytes >= 28) { if (s->first_packet || rtcp_bytes >= 28) {
/* compute NTP time */ rtcp_send_sr(s1, av_gettime());
/* XXX: 90 kHz timestamp hardcoded */
ntp_time = (pkt->pts << 28) / 5625;
rtcp_send_sr(s1, ntp_time);
s->last_octet_count = s->octet_count; s->last_octet_count = s->octet_count;
s->first_packet = 0; s->first_packet = 0;
} }