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gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first

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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
This commit is contained in:
Wim Taymans 2008-04-25 08:15:58 +00:00 committed by Tim-Philipp Müller
parent 3c5cf0cd38
commit e779adca69
5 changed files with 11 additions and 1 deletions
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3
gst/rtpmanager/gstrtpbin.c
View file

@ -306,6 +306,7 @@ struct _GstRtpBinStream
/* for lip-sync */
guint64 clock_base;
guint64 clock_base_time;
gint clock_rate;
gint64 ts_offset;
gint64 prev_ts_offset;
@ -785,6 +786,7 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
stream->local_unix =
gst_util_uint64_scale_int (stream->local_rtp, GST_SECOND,
stream->clock_rate);
stream->local_unix += stream->clock_base_time;
/* calculate delta between server and receiver */
stream->unix_delta = stream->last_unix - stream->local_unix;
@ -942,6 +944,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
if (type == GST_RTCP_SDES_CNAME) {
stream->clock_base = GST_BUFFER_OFFSET (buffer);
stream->clock_base_time = GST_BUFFER_OFFSET_END (buffer);
/* associate the stream to CNAME */
gst_rtp_bin_associate (bin, stream, len, data);
}

2
gst/rtpmanager/rtpsession.c
View file

@ -1233,6 +1233,7 @@ update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
/* get time of arrival */
g_get_current_time (&current);
arrival->time = GST_TIMEVAL_TO_TIME (current);
arrival->timestamp = GST_BUFFER_TIMESTAMP (buffer);
arrival->ntpnstime = ntpnstime;
/* get packet size including header overhead */
@ -1434,6 +1435,7 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
return;
GST_BUFFER_OFFSET (packet->buffer) = source->clock_base;
GST_BUFFER_OFFSET_END (packet->buffer) = source->clock_base_time;
prevsender = RTP_SOURCE_IS_SENDER (source);

2
gst/rtpmanager/rtpsource.c
View file

@ -171,6 +171,7 @@ rtp_source_init (RTPSource * src)
src->payload = 0;
src->clock_rate = -1;
src->clock_base = -1;
src->clock_base_time = -1;
src->packets = g_queue_new ();
src->seqnum_base = -1;
src->last_rtptime = -1;
@ -772,6 +773,7 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
if (src->clock_base == -1) {
GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
src->clock_base = rtptime;
src->clock_base_time = arrival->timestamp;
}
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't

1
gst/rtpmanager/rtpsource.h
View file

@ -134,6 +134,7 @@ struct _RTPSource {
gint clock_rate;
gint32 seqnum_base;
gint64 clock_base;
guint64 clock_base_time;
GstClockTime bye_time;
GstClockTime last_activity;

4
gst/rtpmanager/rtpstats.h
View file

@ -56,7 +56,8 @@ typedef struct {
/**
* RTPArrivalStats:
* @time: arrival time of a packet
* @time: arrival time of a packet according to the system clock
* @timestamp: arrival time of a packet as buffer timestamp
* @address: address of the sender of the packet
* @bytes: bytes of the packet including lowlevel overhead
* @payload_len: bytes of the RTP payload
@ -65,6 +66,7 @@ typedef struct {
*/
typedef struct {
GstClockTime time;
GstClockTime timestamp;
guint64 ntpnstime;
gboolean have_address;
GstNetAddress address;