src->payload = 0;
src->clock_rate = -1;
+ src->clock_base = -1;
src->packets = g_queue_new ();
+ src->seqnum_base = -1;
+ src->last_rtptime = -1;
src->stats.cycles = -1;
src->stats.jitter = 0;
}
/**
+ * rtp_source_update_caps:
+ * @src: an #RTPSource
+ * @caps: a #GstCaps
+ *
+ * Parse @caps and store all relevant information in @source.
+ */
+void
+rtp_source_update_caps (RTPSource * src, GstCaps * caps)
+{
+ GstStructure *s;
+ guint val;
+ gint ival;
+
+ /* nothing changed, return */
+ if (src->caps == caps)
+ return;
+
+ s = gst_caps_get_structure (caps, 0);
+
+ if (gst_structure_get_int (s, "payload", &ival))
+ src->payload = ival;
+ GST_DEBUG ("got payload %d", src->payload);
+
+ gst_structure_get_int (s, "clock-rate", &src->clock_rate);
+ GST_DEBUG ("got clock-rate %d", src->clock_rate);
+
+ if (gst_structure_get_uint (s, "clock-base", &val))
+ src->clock_base = val;
+ GST_DEBUG ("got clock-base %" G_GINT64_FORMAT, src->clock_base);
+
+ if (gst_structure_get_uint (s, "seqnum-base", &val))
+ src->seqnum_base = val;
+ GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
+
+ gst_caps_replace (&src->caps, caps);
+}
+
+/**
* rtp_source_set_callbacks:
* @src: an #RTPSource
* @cb: callback functions
static gint
get_clock_rate (RTPSource * src, guint8 payload)
{
- if (payload != src->payload) {
+ if (src->clock_rate == -1) {
gint clock_rate = -1;
if (src->callbacks.clock_rate)
GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
src->clock_rate = clock_rate;
- src->payload = payload;
}
+ src->payload = payload;
+
return src->clock_rate;
}
+/* Jitter is the variation in the delay of received packets in a flow. It is
+ * measured by comparing the interval when RTP packets were sent to the interval
+ * at which they were received. For instance, if packet #1 and packet #2 leave
+ * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
+ * milliseconds. */
static void
calculate_jitter (RTPSource * src, GstBuffer * buffer,
RTPArrivalStats * arrival)
{
- GstClockTime current;
+ guint64 ntpnstime;
guint32 rtparrival, transit, rtptime;
gint32 diff;
gint clock_rate;
guint8 pt;
/* get arrival time */
- if ((current = arrival->time) == GST_CLOCK_TIME_NONE)
+ if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE)
goto no_time;
pt = gst_rtp_buffer_get_payload_type (buffer);
+ GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
+
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;
rtptime = gst_rtp_buffer_get_timestamp (buffer);
- /* convert arrival time to RTP timestamp units */
- rtparrival = gst_util_uint64_scale_int (current, clock_rate, GST_SECOND);
+ /* no clock-base, take first rtptime as base */
+ if (src->clock_base == -1) {
+ GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
+ src->clock_base = rtptime;
+ }
+
+ /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
+ * care about the absolute value, just the difference. */
+ rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
/* 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.last_rtptime = rtparrival;
- GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %u",
- rtparrival, rtptime, clock_rate, diff, src->stats.jitter);
+ GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
+ rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
return;
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);
+ rtp_source_update_caps (src, GST_BUFFER_CAPS (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;
src->probation--;
src->stats.max_seq = seqnr;
if (src->probation == 0) {
- GST_DEBUG ("probation done!", src->probation);
+ GST_DEBUG ("probation done!");
init_seq (src, seqnr);
} else {
GstBuffer *q;
/* 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) {
}
} else {
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
+ GST_WARNING ("duplicate or reordered packet");
}
src->stats.octets_received += arrival->payload_len;
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 and perform skew correction */
calculate_jitter (src, buffer, arrival);
/* we're ready to push the RTP packet now */
* rtp_source_send_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
+ * @ntpnstime: the NTP time when this buffer was captured in nanoseconds
*
* Send an RTP @buffer originating from @src. This will make @src a sender.
+ * This function takes ownership of @buffer and modifies the SSRC in the RTP
+ * packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
+rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
{
GstFlowReturn result = GST_FLOW_OK;
+ guint len;
+ guint32 rtptime;
+ guint64 ext_rtptime;
+ guint64 ntp_diff, rtp_diff;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ len = gst_rtp_buffer_get_payload_len (buffer);
+
+ rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
+
/* we are a sender now */
src->is_sender = TRUE;
+ /* update stats for the SR */
+ src->stats.packets_sent++;
+ src->stats.octets_sent += len;
+
+ rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ ext_rtptime = src->last_rtptime;
+ ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
+
+ GST_DEBUG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
+ src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
+
+ if (ext_rtptime > src->last_rtptime) {
+ rtp_diff = ext_rtptime - src->last_rtptime;
+ ntp_diff = ntpnstime - src->last_ntpnstime;
+
+ /* calc the diff so we can detect drift at the sender. This can also be used
+ * to guestimate the clock rate if the NTP time is locked to the RTP
+ * timestamps (as is the case when the capture device is providing the clock). */
+ GST_DEBUG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
+ GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
+ }
+
+ /* we keep track of the last received RTP timestamp and the corresponding
+ * NTP timestamp so that we can use this info when constructing SR reports */
+ src->last_rtptime = ext_rtptime;
+ src->last_ntpnstime = ntpnstime;
+
/* push packet */
- if (src->callbacks.push_rtp)
+ if (src->callbacks.push_rtp) {
+ guint32 ssrc;
+
+ ssrc = gst_rtp_buffer_get_ssrc (buffer);
+ if (ssrc != src->ssrc) {
+ /* the SSRC of the packet is not correct, make a writable buffer and
+ * update the SSRC. This could involve a complete copy of the packet when
+ * it is not writable. Usually the payloader will use caps negotiation to
+ * get the correct SSRC from the session manager before pushing anything. */
+ buffer = gst_buffer_make_writable (buffer);
+
+ GST_WARNING ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
+ src->ssrc);
+ gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
+ }
+ GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
+ src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
- else
+ } else {
+ GST_WARNING ("no callback installed, dropping packet");
gst_buffer_unref (buffer);
+ }
return result;
}
/**
* rtp_source_process_sr:
* @src: an #RTPSource
+ * @time: time of packet arrival
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* Update the sender report in @src.
*/
void
-rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
- guint32 packet_count, guint32 octet_count)
+rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
+ guint32 rtptime, guint32 packet_count, guint32 octet_count)
{
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,
- octet_count);
+ GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
+ ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
+ (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
+ packet_count, octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
+ curr->time = time;
/* make current */
src->stats.curr_sr = curridx;
/**
* rtp_source_process_rb:
* @src: an #RTPSource
+ * @time: the current time in nanoseconds since 1970
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* Update the report block in @src.
*/
void
-rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
- guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr)
+rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost,
+ gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr,
+ guint32 dlsr)
{
RTPReceiverReport *curr;
gint curridx;
+ guint32 ntp, A;
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,
- packetslost, exthighestseq, jitter, lsr, dlsr);
+ GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
+ ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
+ src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
+ lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
curridx = src->stats.curr_rr ^ 1;
curr = &src->stats.rr[curridx];
curr->lsr = lsr;
curr->dlsr = dlsr;
+ /* calculate round trip */
+ ntp = (gst_rtcp_unix_to_ntp (time) >> 16) & 0xffffffff;
+ A = ntp - dlsr;
+ A -= lsr;
+ curr->round_trip = A;
+
+ GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
+ A >> 16, A & 0xffff);
+
/* make current */
src->stats.curr_rr = curridx;
}
+
+/**
+ * rtp_source_get_new_sr:
+ * @src: an #RTPSource
+ * @time: the current time in nanoseconds since 1970
+ * @ntptime: the NTP time
+ * @rtptime: the RTP time
+ * @packet_count: the packet count
+ * @octet_count: the octect count
+ *
+ * Get new values to put into a new SR report from this source.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+rtp_source_get_new_sr (RTPSource * src, GstClockTime ntpnstime,
+ guint64 * ntptime, guint32 * rtptime, guint32 * packet_count,
+ guint32 * octet_count)
+{
+ guint64 t_rtp;
+ guint64 t_current_ntp;
+ GstClockTimeDiff diff;
+
+ g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
+
+ /* use the sync params to interpollate the date->time member to rtptime. We
+ * use the last sent timestamp and rtptime as reference points. We assume
+ * that the slope of the rtptime vs timestamp curve is 1, which is certainly
+ * sufficient for the frequency at which we report SR and the rate we send
+ * out RTP packets. */
+ t_rtp = src->last_rtptime;
+
+ GST_DEBUG ("last_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %"
+ G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp);
+
+ if (src->clock_rate != -1) {
+ /* get the diff with the SR time */
+ diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime);
+
+ /* now translate the diff to RTP time, handle positive and negative cases.
+ * If there is no diff, we already set rtptime correctly above. */
+ if (diff > 0) {
+ GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
+ t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
+ } else {
+ diff = -diff;
+ GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
+ t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
+ }
+ } else {
+ GST_WARNING ("no clock-rate, cannot interpollate rtp time");
+ }
+
+ /* convert the NTP time in nanoseconds to 32.32 fixed point */
+ t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
+
+ GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
+ (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
+ (guint32) t_rtp);
+
+ if (ntptime)
+ *ntptime = t_current_ntp;
+ if (rtptime)
+ *rtptime = t_rtp;
+ if (packet_count)
+ *packet_count = src->stats.packets_sent;
+ if (octet_count)
+ *octet_count = src->stats.octets_sent;
+
+ return TRUE;
+}
+
+/**
+ * rtp_source_get_new_rb:
+ * @src: an #RTPSource
+ * @time: the current time in nanoseconds since 1970
+ * @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 on success.
+ */
+gboolean
+rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
+ guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
+ guint32 * jitter, guint32 * lsr, guint32 * dlsr)
+{
+ RTPSourceStats *stats;
+ guint64 extended_max, expected;
+ guint64 expected_interval, received_interval, ntptime;
+ gint64 lost, lost_interval;
+ guint32 fraction, LSR, DLSR;
+ GstClockTime sr_time;
+
+ stats = &src->stats;
+
+ extended_max = stats->cycles + stats->max_seq;
+ expected = extended_max - stats->base_seq + 1;
+
+ GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
+ ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
+ 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;
+ received_interval = stats->packets_received - stats->prev_received;
+ stats->prev_received = stats->packets_received;
+
+ lost_interval = expected_interval - received_interval;
+
+ if (expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+
+ GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
+ /* we scaled the jitter up for additional precision */
+ GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
+ ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
+ extended_max, stats->jitter >> 4);
+
+ if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
+ GstClockTime diff;
+
+ /* LSR is middle 32 bits of the last ntptime */
+ LSR = (ntptime >> 16) & 0xffffffff;
+ diff = time - sr_time;
+ GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
+ /* DLSR, delay since last SR is expressed in 1/65536 second units */
+ DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
+ } else {
+ /* No valid SR received, LSR/DLSR are set to 0 then */
+ GST_DEBUG ("no valid SR received");
+ LSR = 0;
+ DLSR = 0;
+ }
+ GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
+ DLSR >> 16, DLSR & 0xffff);
+
+ if (fractionlost)
+ *fractionlost = fraction;
+ if (packetslost)
+ *packetslost = lost;
+ if (exthighestseq)
+ *exthighestseq = extended_max;
+ if (jitter)
+ *jitter = stats->jitter >> 4;
+ if (lsr)
+ *lsr = LSR;
+ if (dlsr)
+ *dlsr = DLSR;
+
+ return TRUE;
+}
+
+/**
+ * rtp_source_get_last_sr:
+ * @src: an #RTPSource
+ * @time: time of packet arrival
+ * @ntptime: the NTP time
+ * @rtptime: the RTP time
+ * @packet_count: the packet count
+ * @octet_count: the octect count
+ *
+ * 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, GstClockTime * time, guint64 * ntptime,
+ guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
+{
+ 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;
+}
projects / platform / upstream / gst-plugins-good.git / blobdiff