Commit 85b99b90 authored by Wim Taymans's avatar Wim Taymans
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gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect...

gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
parent 601b0f1d
2008-08-13 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
2008-08-12 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
 
* ext/x264/gstx264enc.c: (gst_x264_enc_reset),
......
......@@ -119,6 +119,7 @@
#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
#include "gstrtpsession.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
......@@ -317,6 +318,7 @@ struct _GstRtpBinStream
gint64 unix_delta;
/* for lip-sync */
guint64 last_clock_base;
guint64 clock_base;
guint64 clock_base_time;
gint clock_rate;
......@@ -876,13 +878,18 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
else
diff = ostream->ts_offset - ostream->prev_ts_offset;
GST_DEBUG_OBJECT (bin,
"ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ostream->ts_offset,
ostream->prev_ts_offset, diff);
/* only change diff when it changed more than 1 millisecond. This
* compensates for rounding errors in NTP to RTP timestamp
* conversions */
if (diff > GST_MSECOND)
if (diff > GST_MSECOND && diff < (3 * GST_SECOND)) {
g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
ostream->prev_ts_offset = ostream->ts_offset;
ostream->prev_ts_offset = ostream->ts_offset;
}
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
ostream->ssrc, ostream->ts_offset);
......@@ -929,6 +936,9 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
guint32 rtptime;
gboolean have_sr, have_sdes;
gboolean more;
guint64 clock_base;
clock_base = GST_BUFFER_OFFSET (buffer);
stream = gst_pad_get_element_private (pad);
bin = stream->bin;
......@@ -938,6 +948,14 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
if (!gst_rtcp_buffer_validate (buffer))
goto invalid_rtcp;
/* clock base changes when there is a huge gap in the timestamps or seqnum.
* When this happens we don't want to calculate the extended timestamp based
* on the previous one but reset the calculation. */
if (stream->last_clock_base != clock_base) {
stream->last_extrtptime = -1;
stream->last_clock_base = clock_base;
}
have_sr = FALSE;
have_sdes = FALSE;
GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) {
......@@ -989,7 +1007,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
if (type == GST_RTCP_SDES_CNAME) {
stream->clock_base = GST_BUFFER_OFFSET (buffer);
stream->clock_base = clock_base;
stream->clock_base_time = GST_BUFFER_OFFSET_END (buffer);
/* associate the stream to CNAME */
gst_rtp_bin_associate (bin, stream, len, data);
......@@ -1876,6 +1894,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
}
stream->last_clock_base = -1;
if (gst_structure_get_uint (s, "clock-base", &val))
stream->clock_base = val;
else
......
......@@ -237,6 +237,7 @@ struct _GstRtpSessionPrivate
{
GMutex *lock;
GstClock *sysclock;
RTPSession *session;
/* thread for sending out RTCP */
......@@ -1846,3 +1847,9 @@ static void
gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
{
}
void
gst_rtp_session_set_ssrc (GstRtpSession * sess, guint32 ssrc)
{
rtp_session_set_internal_ssrc (sess->priv->session, ssrc);
}
......@@ -75,4 +75,6 @@ struct _GstRtpSessionClass {
GType gst_rtp_session_get_type (void);
void gst_rtp_session_set_ssrc (GstRtpSession *sess, guint32 ssrc);
#endif /* __GST_RTP_SESSION_H__ */
......@@ -916,7 +916,6 @@ check_collision (RTPSession * sess, RTPSource * source,
/* This is not our local source, but lets check if two remote
* source collide
*/
if (rtp) {
if (source->have_rtp_from) {
if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
......@@ -938,8 +937,9 @@ check_collision (RTPSession * sess, RTPSource * source,
return FALSE;
}
}
/* In this case, we have third-party collision or loop */
/* We received RTP or RTCP from this source before but the network address
* changed. In this case, we have third-party collision or loop */
GST_DEBUG ("we have a third-party collision or loop");
/* FIXME: Log 3rd party collision somehow
* Maybe should be done in upper layer, only the SDES can tell us
......@@ -1026,7 +1026,7 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
* rtp_session_get_internal_source:
* @sess: a #RTPSession
*
* Get the internal #RTPSource of @session.
* Get the internal #RTPSource of @sess.
*
* Returns: The internal #RTPSource. g_object_unref() after usage.
*/
......@@ -1042,6 +1042,48 @@ rtp_session_get_internal_source (RTPSession * sess)
return result;
}
/**
* rtp_session_set_internal_ssrc:
* @sess: a #RTPSession
* @ssrc: an SSRC
*
* Set the SSRC of @sess to @ssrc.
*/
void
rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
{
RTP_SESSION_LOCK (sess);
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc));
sess->source->ssrc = ssrc;
rtp_source_reset (sess->source);
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc), sess->source);
RTP_SESSION_UNLOCK (sess);
}
/**
* rtp_session_get_internal_ssrc:
* @sess: a #RTPSession
*
* Get the internal SSRC of @sess.
*
* Returns: The SSRC of the session.
*/
guint32
rtp_session_get_internal_ssrc (RTPSession * sess)
{
guint32 ssrc;
RTP_SESSION_LOCK (sess);
ssrc = sess->source->ssrc;
RTP_SESSION_UNLOCK (sess);
return ssrc;
}
/**
* rtp_session_add_source:
* @sess: a #RTPSession
......@@ -2285,6 +2327,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
}
/* check for outdated collisions */
GST_DEBUG ("checking collision list");
item = g_list_first (sess->conflicting_addresses);
while (item) {
RTPConflictingAddress *known_conflict = item->data;
......@@ -2294,12 +2337,14 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
RTCP_INTERVAL_COLLISION_TIMEOUT)) {
sess->conflicting_addresses =
g_list_delete_link (sess->conflicting_addresses, item);
GST_DEBUG ("collision %p timed out", known_conflict);
g_free (known_conflict);
}
item = next_item;
}
if (sess->change_ssrc) {
GST_DEBUG ("need to change our SSRC (%08x)", sess->source->ssrc);
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc));
......@@ -2313,6 +2358,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
sess->bye_reason = NULL;
sess->sent_bye = FALSE;
sess->change_ssrc = FALSE;
GST_DEBUG ("changed our SSRC to %08x", sess->source->ssrc);
}
RTP_SESSION_UNLOCK (sess);
......
......@@ -264,6 +264,10 @@ gchar* rtp_session_get_sdes_string (RTPSession *sess, GstRTCPSDE
/* handling sources */
RTPSource* rtp_session_get_internal_source (RTPSession *sess);
void rtp_session_set_internal_ssrc (RTPSession *sess, guint32 ssrc);
guint32 rtp_session_get_internal_ssrc (RTPSession *sess);
gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
guint rtp_session_get_num_sources (RTPSession *sess);
guint rtp_session_get_num_active_sources (RTPSession *sess);
......
......@@ -923,11 +923,13 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
} else {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
src->clock_base = -1;
goto bad_sequence;
}
} else {
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
GST_WARNING ("duplicate or reordered packet");
src->clock_base = -1;
}
src->stats.octets_received += arrival->payload_len;
......
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