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):
fix for pad name change
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;
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);
-
- GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
+ res = gst_clock_id_wait (id, NULL);
+ if (res != GST_CLOCK_UNSCHEDULED) {
+ 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);
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 */
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++;
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);
{
RTPSession *sess;
GstBuffer *rtcp;
+ GstClockTime time;
GstRTCPPacket packet;
} ReportData;
}
}
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;
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);
}
}
}
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],
/* 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;
src->stats.bytes_received = 0;
src->stats.prev_received = 0;
src->stats.prev_expected = 0;
+
+ GST_DEBUG ("base_seq %d", seq);
}
/**
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;
/* 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) {
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);
* @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;
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
+ curr->time = time;
/* make current */
src->stats.curr_sr = curridx;
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;
/* 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;
+}
/* 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__ */
guint32 rtptime;
guint32 packet_count;
guint32 octet_count;
+ GstClockTime time;
} RTPSenderReport;
/**
guint32 prev_received;
guint16 max_seq;
- guint32 cycles;
+ guint64 cycles;
guint32 base_seq;
guint32 bad_seq;
guint32 transit;
projects / platform / upstream / gstreamer.git / commitdiff