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gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.

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Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
This commit is contained in:
Wim Taymans 2007-04-25 13:19:36 +00:00
parent 34534179a2
commit 67c69ca0ea
7 changed files with 183 additions and 42 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

22
ChangeLog
View file

@ -1,3 +1,25 @@
2007-04-25 Wim Taymans <wim@fluendo.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
2007-04-25 Wim Taymans <wim@fluendo.com>
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):

2
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -553,6 +553,8 @@ gst_rtp_jitter_buffer_change_state (GstElement * element,
async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
TRUE);
async_jitter_queue_unlock (priv->queue);
if (ret != GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;

16
gst/rtpmanager/gstrtpsession.c
View file

@ -304,21 +304,27 @@ rtcp_thread (GstRTPSession * rtpsession)
while (!rtpsession->priv->stop_thread) {
gdouble timeout;
GstClockTime target;
GstClockReturn res;
timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
target = gst_clock_get_time (clock);
target += GST_SECOND * timeout;
id = rtpsession->priv->id = gst_clock_new_single_shot_id (clock, target);
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_clock_id_wait (id, NULL);
res = gst_clock_id_wait (id, NULL);
if (res != GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
/* make the session manager produce RTCP, we ignore the result. */
rtp_session_perform_reporting (rtpsession->priv->session);
/* make the session manager produce RTCP, we ignore the result. */
rtp_session_perform_reporting (rtpsession->priv->session);
} else {
GST_DEBUG_OBJECT (rtpsession, "got unscheduled");
}
GST_RTP_SESSION_LOCK (rtpsession);
gst_clock_id_unref (id);

54
gst/rtpmanager/rtpsession.c
View file

@ -893,6 +893,7 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
prevsender = RTP_SOURCE_IS_SENDER (source);
prevactive = RTP_SOURCE_IS_ACTIVE (source);
/* we need to ref so that we can process the CSRCs later */
gst_buffer_ref (buffer);
/* let source process the packet */
@ -982,7 +983,8 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
prevsender = RTP_SOURCE_IS_SENDER (source);
/* first update the source */
rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count);
rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count,
arrival->time);
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
sess->stats.sender_sources++;
@ -1004,7 +1006,7 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
if (ssrc == sess->source->ssrc) {
/* only deal with report blocks for our session, we update the stats of
* the sender of the TCP message. We could also compare our stats against
* the sender of the RTCP message. We could also compare our stats against
* the other sender to see if we are better or worse. */
rtp_source_process_rb (source, fractionlost, packetslost,
exthighestseq, jitter, lsr, dlsr);
@ -1292,6 +1294,7 @@ typedef struct
{
RTPSession *sess;
GstBuffer *rtcp;
GstClockTime time;
GstRTCPPacket packet;
} ReportData;
@ -1322,29 +1325,25 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
}
}
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
/* only report about other sources */
if (source != sess->source) {
/* only report about other sender sources */
if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
RTPSourceStats *stats;
guint32 extended_max, expected;
guint32 expected_interval, received_interval;
guint32 lost, lost_interval, fraction;
guint64 extended_max, expected;
guint64 expected_interval, received_interval, ntptime;
gint64 lost, lost_interval;
guint32 fraction, LSR, DLSR;
GstClockTime time;
stats = &source->stats;
extended_max = (stats->cycles << 16) + stats->max_seq;
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
if (expected > stats->packets_received) {
lost = expected - stats->packets_received;
if (lost > 0x7fffff)
lost = 0x7fffff;
} else {
lost = stats->packets_received - expected;
if (lost > 0x800000)
lost = 0x800000;
else
lost = -lost;
}
GST_DEBUG ("ext_max %d, expected %d, received %d, base_seq %d",
extended_max, expected, stats->packets_received, stats->base_seq);
lost = expected - stats->packets_received;
lost = CLAMP (lost, -0x800000, 0x7fffff);
expected_interval = expected - stats->prev_expected;
stats->prev_expected = expected;
@ -1363,9 +1362,21 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {
/* LSR is middle bits of the last ntptime */
LSR = (ntptime >> 16) & 0xffffffff;
/* DLSR, delay since last SR is expressed in 1/65536 second units */
DLSR = gst_util_uint64_scale_int (data->time - time, 65536, GST_SECOND);
} else {
/* No valid SR received, LSR/DLSR are set to 0 then */
LSR = 0;
DLSR = 0;
}
GST_DEBUG ("LSR %08x, DLSR %08x", LSR, DLSR);
/* packet is not yet filled, add report block for this source. */
gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
extended_max, stats->jitter >> 4, 0, 0);
extended_max, stats->jitter >> 4, LSR, DLSR);
}
}
}
@ -1413,6 +1424,9 @@ rtp_session_perform_reporting (RTPSession * sess)
data.sess = sess;
data.rtcp = NULL;
/* get time so it can be used later */
data.time = sess->callbacks.get_time (sess, sess->user_data);
RTP_SESSION_LOCK (sess);
/* loop over all known sources and do something */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],

120
gst/rtpmanager/rtpsource.c
View file

@ -248,15 +248,19 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
/* transit time is difference with RTP timestamp */
transit = rtparrival - rtptime;
/* get diff with previous transit time */
if (src->stats.transit != -1)
diff = transit - src->stats.transit;
else
/* get ABS diff with previous transit time */
if (src->stats.transit != -1) {
if (transit > src->stats.transit)
diff = transit - src->stats.transit;
else
diff = src->stats.transit - transit;
} else
diff = 0;
src->stats.transit = transit;
if (diff < 0)
diff = -diff;
/* update jitter */
/* update jitter, the value we store is scaled up so we can keep precision. */
src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
src->stats.prev_rtptime = src->stats.last_rtptime;
@ -292,6 +296,8 @@ init_seq (RTPSource * src, guint16 seq)
src->stats.bytes_received = 0;
src->stats.prev_received = 0;
src->stats.prev_expected = 0;
GST_DEBUG ("base_seq %d", seq);
}
/**
@ -319,7 +325,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
seqnr = gst_rtp_buffer_get_seq (buffer);
if (stats->cycles == -1) {
GST_DEBUG ("first buffer");
GST_DEBUG ("received first buffer");
/* first time we heard of this source */
init_seq (src, seqnr);
src->stats.max_seq = seqnr - 1;
@ -366,7 +372,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
/* in order, with permissible gap */
if (seqnr < stats->max_seq) {
/* sequence number wrapped - count another 64K cycle. */
stats->cycles++;
stats->cycles += RTP_SEQ_MOD;
}
stats->max_seq = seqnr;
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
@ -392,8 +398,8 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
src->is_sender = TRUE;
src->validated = TRUE;
GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
src->stats.packets_received, src->stats.octets_received);
GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
/* calculate jitter */
calculate_jitter (src, buffer, arrival);
@ -470,20 +476,21 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
* @time: time of packet arrival
*
* Update the sender report in @src.
*/
void
rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count)
guint32 packet_count, guint32 octet_count, GstClockTime time)
{
RTPSenderReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
", RTP %u, PC %u, OC %u", src->ssrc, ntptime, rtptime, packet_count,
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %u, PC %u, OC %u",
src->ssrc, ntptime >> 32, ntptime & 0xffffffff, rtptime, packet_count,
octet_count);
curridx = src->stats.curr_sr ^ 1;
@ -498,6 +505,7 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
curr->time = time;
/* make current */
src->stats.curr_sr = curridx;
@ -525,7 +533,7 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", src->ssrc, fractionlost,
", PL %u, HS %u, JITTER %u, LSR %08x, DLSR %08x", src->ssrc, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
curridx = src->stats.curr_rr ^ 1;
@ -543,3 +551,85 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
/* make current */
src->stats.curr_rr = curridx;
}
/**
* rtp_source_get_last_sr:
* @src: an #RTPSource
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
* @time: time of packet arrival
*
* Get the values of the last sender report as set with rtp_source_process_sr().
*
* Returns: %TRUE if there was a valid SR report.
*/
gboolean
rtp_source_get_last_sr (RTPSource * src, guint64 * ntptime, guint32 * rtptime,
guint32 * packet_count, guint32 * octet_count, GstClockTime * time)
{
RTPSenderReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.sr[src->stats.curr_sr];
if (!curr->is_valid)
return FALSE;
if (ntptime)
*ntptime = curr->ntptime;
if (rtptime)
*rtptime = curr->rtptime;
if (packet_count)
*packet_count = curr->packet_count;
if (octet_count)
*octet_count = curr->octet_count;
if (time)
*time = curr->time;
return TRUE;
}
/**
* rtp_source_get_last_rb:
* @src: an #RTPSource
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
* Returns: %TRUE if there was a valid SB report.
*/
gboolean
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
guint32 * lsr, guint32 * dlsr)
{
RTPReceiverReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.rr[src->stats.curr_rr];
if (!curr->is_valid)
return FALSE;
if (fractionlost)
*fractionlost = curr->fractionlost;
if (packetslost)
*packetslost = curr->packetslost;
if (exthighestseq)
*exthighestseq = curr->exthighestseq;
if (jitter)
*jitter = curr->jitter;
if (lsr)
*lsr = curr->lsr;
if (dlsr)
*dlsr = curr->dlsr;
return TRUE;
}

8
gst/rtpmanager/rtpsource.h
View file

@ -167,9 +167,15 @@ GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer);
/* RTCP messages */
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count);
guint32 packet_count, guint32 octet_count, GstClockTime time);
void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
gboolean rtp_source_get_last_sr (RTPSource *src, guint64 *ntptime, guint32 *rtptime,
guint32 *packet_count, guint32 *octet_count, GstClockTime *time);
gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
#endif /* __RTP_SOURCE_H__ */

3
gst/rtpmanager/rtpstats.h
View file

@ -34,6 +34,7 @@ typedef struct {
guint32 rtptime;
guint32 packet_count;
guint32 octet_count;
GstClockTime time;
} RTPSenderReport;
/**
@ -100,7 +101,7 @@ typedef struct {
guint32 prev_received;
guint16 max_seq;
guint32 cycles;
guint64 cycles;
guint32 base_seq;
guint32 bad_seq;
guint32 transit;