Simplify book-keeping of lost packets
Update the |cumulative_lost_| counter per received packet. The rules follow from RFC 3550 and are fairly simple: Decrement the counter by one for every received packet. For every in-order packet, i.e., increasing |received_seq_max_|, add that change to |cumulative_lost_|. Net change is zero as long as packets are received in proper sequence. This way, GetStats() always returns an up-to-date value, independent of the timing of RTCP report blocks. For RTCP reports, keep a workaround to never report negative cumulative loss. Bug: webrtc:10679 Change-Id: I47ff3bf266ff2382f405ec9828d34f7fad7068b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150641 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29058}
This commit is contained in:
@ -41,37 +41,37 @@ StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc,
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enable_retransmit_detection_(false),
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jitter_q4_(0),
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cumulative_loss_(0),
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cumulative_loss_rtcp_offset_(0),
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last_receive_time_ms_(0),
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last_received_timestamp_(0),
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received_seq_first_(0),
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received_seq_first_(-1),
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received_seq_max_(-1),
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last_report_inorder_packets_(0),
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last_report_old_packets_(0),
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last_report_cumulative_loss_(0),
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last_report_seq_max_(-1) {}
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StreamStatisticianImpl::~StreamStatisticianImpl() = default;
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void StreamStatisticianImpl::OnRtpPacket(const RtpPacketReceived& packet) {
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UpdateCounters(packet);
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}
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bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
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int64_t sequence_number,
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int64_t now_ms) {
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RTC_DCHECK_EQ(sequence_number,
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seq_unwrapper_.UnwrapWithoutUpdate(packet.SequenceNumber()));
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// Check if |packet| is second packet of a stream restart.
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if (received_seq_out_of_order_) {
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// Count the previous packet as a received; it was postponed below.
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--cumulative_loss_;
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uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
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received_seq_out_of_order_ = absl::nullopt;
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if (packet.SequenceNumber() == expected_sequence_number) {
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// Ignore sequence number gap caused by stream restart for next packet
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// loss calculation.
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last_report_seq_max_ = sequence_number;
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last_report_inorder_packets_ = receive_counters_.transmitted.packets -
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receive_counters_.retransmitted.packets;
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// As final part of stream restart consider |packet| is not out of order.
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// Ignore sequence number gap caused by stream restart for packet loss
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// calculation, by setting received_seq_max_ to the sequence number just
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// before the out-of-order seqno. This gives a net zero change of
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// |cumulative_loss_|, for the two packets interpreted as a stream reset.
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//
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// Fraction loss for the next report may get a bit off, since we don't
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// update last_report_seq_max_ and last_report_cumulative_loss_ in a
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// consistent way.
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last_report_seq_max_ = sequence_number - 2;
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received_seq_max_ = sequence_number - 2;
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return false;
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}
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}
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@ -81,6 +81,13 @@ bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
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// Sequence number gap looks too large, wait until next packet to check
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// for a stream restart.
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received_seq_out_of_order_ = packet.SequenceNumber();
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// Postpone counting this as a received packet until we know how to update
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// |received_seq_max_|, otherwise we temporarily decrement
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// |cumulative_loss_|. The
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// ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
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// |cumulative_loss_| to be unchanged by the reception of the first packet
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// after stream reset.
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++cumulative_loss_;
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return true;
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}
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@ -93,8 +100,7 @@ bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
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return true;
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}
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StreamDataCounters StreamStatisticianImpl::UpdateCounters(
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const RtpPacketReceived& packet) {
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void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) {
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rtc::CritScope cs(&stream_lock_);
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RTC_DCHECK_EQ(ssrc_, packet.Ssrc());
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int64_t now_ms = clock_->TimeInMilliseconds();
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@ -102,17 +108,21 @@ StreamDataCounters StreamStatisticianImpl::UpdateCounters(
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incoming_bitrate_.Update(packet.size(), now_ms);
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receive_counters_.last_packet_received_timestamp_ms = now_ms;
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receive_counters_.transmitted.AddPacket(packet);
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--cumulative_loss_;
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int64_t sequence_number =
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seq_unwrapper_.UnwrapWithoutUpdate(packet.SequenceNumber());
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if (!ReceivedRtpPacket()) {
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received_seq_first_ = sequence_number;
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last_report_seq_max_ = sequence_number - 1;
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received_seq_max_ = sequence_number - 1;
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receive_counters_.first_packet_time_ms = now_ms;
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} else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) {
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return receive_counters_;
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return;
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}
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// In order packet.
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cumulative_loss_ += sequence_number - received_seq_max_;
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received_seq_max_ = sequence_number;
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seq_unwrapper_.UpdateLast(sequence_number);
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@ -125,7 +135,6 @@ StreamDataCounters StreamStatisticianImpl::UpdateCounters(
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}
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last_received_timestamp_ = packet.Timestamp();
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last_receive_time_ms_ = now_ms;
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return receive_counters_;
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}
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void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
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@ -181,7 +190,7 @@ bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
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}
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if (!reset) {
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if (last_report_inorder_packets_ == 0) {
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if (!ReceivedRtpPacket()) {
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// No report.
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return false;
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}
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@ -218,40 +227,24 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
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int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
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RTC_DCHECK_GE(exp_since_last, 0);
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// Number of received RTP packets since last report, counts all packets but
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// not re-transmissions.
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uint32_t rec_since_last = (receive_counters_.transmitted.packets -
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receive_counters_.retransmitted.packets) -
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last_report_inorder_packets_;
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// With NACK we don't know the expected retransmissions during the last
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// second. We know how many "old" packets we have received. We just count
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// the number of old received to estimate the loss, but it still does not
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// guarantee an exact number since we run this based on time triggered by
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// sending of an RTP packet. This should have a minimum effect.
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// With NACK we don't count old packets as received since they are
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// re-transmitted. We use RTT to decide if a packet is re-ordered or
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// re-transmitted.
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uint32_t retransmitted_packets =
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receive_counters_.retransmitted.packets - last_report_old_packets_;
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rec_since_last += retransmitted_packets;
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int32_t missing = 0;
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if (exp_since_last > rec_since_last) {
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missing = (exp_since_last - rec_since_last);
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}
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uint8_t local_fraction_lost = 0;
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if (exp_since_last) {
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int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
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if (exp_since_last > 0 && lost_since_last > 0) {
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// Scale 0 to 255, where 255 is 100% loss.
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local_fraction_lost = static_cast<uint8_t>(255 * missing / exp_since_last);
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stats.fraction_lost =
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static_cast<uint8_t>(255 * lost_since_last / exp_since_last);
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} else {
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stats.fraction_lost = 0;
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}
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stats.fraction_lost = local_fraction_lost;
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// We need a counter for cumulative loss too.
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// TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24.
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cumulative_loss_ += missing;
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stats.packets_lost = cumulative_loss_;
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// TODO(danilchap): Ensure |stats.packets_lost| is clamped to fit in a signed
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// 24-bit value.
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stats.packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_;
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if (stats.packets_lost < 0) {
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// Clamp to zero. Work around to accomodate for senders that misbehave with
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// negative cumulative loss.
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stats.packets_lost = 0;
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cumulative_loss_rtcp_offset_ = -cumulative_loss_;
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}
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stats.extended_highest_sequence_number =
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static_cast<uint32_t>(received_seq_max_);
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// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
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@ -261,9 +254,7 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
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last_reported_statistics_ = stats;
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// Only for report blocks in RTCP SR and RR.
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last_report_inorder_packets_ = receive_counters_.transmitted.packets -
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receive_counters_.retransmitted.packets;
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last_report_old_packets_ = receive_counters_.retransmitted.packets;
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last_report_cumulative_loss_ = cumulative_loss_;
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last_report_seq_max_ = received_seq_max_;
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BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts",
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clock_->TimeInMilliseconds(),
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@ -277,15 +268,16 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
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absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
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rtc::CritScope cs(&stream_lock_);
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if (received_seq_max_ < 0) {
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if (!ReceivedRtpPacket()) {
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return absl::nullopt;
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}
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int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
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if (expected_packets <= 0) {
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return absl::nullopt;
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}
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// Spec allows negative cumulative loss, but implementation uses uint32_t, so
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// this expression is always non-negative.
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if (cumulative_loss_ <= 0) {
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return 0;
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}
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return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets;
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}
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@ -349,7 +341,7 @@ void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
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// this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has
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// it's own locking so don't hold receive_statistics_lock_ (potential
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// deadlock).
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GetOrCreateStatistician(packet.Ssrc())->OnRtpPacket(packet);
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GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet);
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}
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StreamStatisticianImpl* ReceiveStatisticsImpl::GetStatistician(
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@ -24,8 +24,7 @@
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namespace webrtc {
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class StreamStatisticianImpl : public StreamStatistician,
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public RtpPacketSinkInterface {
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class StreamStatisticianImpl : public StreamStatistician {
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public:
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StreamStatisticianImpl(uint32_t ssrc,
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Clock* clock,
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@ -41,12 +40,12 @@ class StreamStatisticianImpl : public StreamStatistician,
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StreamDataCounters GetReceiveStreamDataCounters() const override;
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uint32_t BitrateReceived() const override;
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// Implements RtpPacketSinkInterface
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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void SetMaxReorderingThreshold(int max_reordering_threshold);
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void EnableRetransmitDetection(bool enable);
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// Updates StreamStatistician for incoming packets.
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void UpdateCounters(const RtpPacketReceived& packet);
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private:
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bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet,
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int64_t now_ms) const
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@ -61,11 +60,9 @@ class StreamStatisticianImpl : public StreamStatistician,
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int64_t sequence_number,
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int64_t now_ms)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
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// Updates StreamStatistician for incoming packets.
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StreamDataCounters UpdateCounters(const RtpPacketReceived& packet);
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// Checks if this StreamStatistician received any rtp packets.
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bool ReceivedRtpPacket() const RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_) {
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return received_seq_max_ >= 0;
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return received_seq_first_ >= 0;
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}
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const uint32_t ssrc_;
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@ -78,7 +75,13 @@ class StreamStatisticianImpl : public StreamStatistician,
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// Stats on received RTP packets.
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uint32_t jitter_q4_ RTC_GUARDED_BY(&stream_lock_);
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uint32_t cumulative_loss_ RTC_GUARDED_BY(&stream_lock_);
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// Cumulative loss according to RFC 3550, which may be negative (and often is,
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// if packets are reordered and there are non-RTX retransmissions).
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int32_t cumulative_loss_ RTC_GUARDED_BY(&stream_lock_);
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// Offset added to outgoing rtcp reports, to make ensure that the reported
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// cumulative loss is non-negative. Reports with negative values confuse some
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// senders, in particular, our own loss-based bandwidth estimator.
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int32_t cumulative_loss_rtcp_offset_ RTC_GUARDED_BY(&stream_lock_);
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int64_t last_receive_time_ms_ RTC_GUARDED_BY(&stream_lock_);
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uint32_t last_received_timestamp_ RTC_GUARDED_BY(&stream_lock_);
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@ -94,8 +97,7 @@ class StreamStatisticianImpl : public StreamStatistician,
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StreamDataCounters receive_counters_ RTC_GUARDED_BY(&stream_lock_);
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// Counter values when we sent the last report.
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uint32_t last_report_inorder_packets_ RTC_GUARDED_BY(&stream_lock_);
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uint32_t last_report_old_packets_ RTC_GUARDED_BY(&stream_lock_);
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int32_t last_report_cumulative_loss_ RTC_GUARDED_BY(&stream_lock_);
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int64_t last_report_seq_max_ RTC_GUARDED_BY(&stream_lock_);
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RtcpStatistics last_reported_statistics_ RTC_GUARDED_BY(&stream_lock_);
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};
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