WebRTC重传包判断
调用处
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bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
int64_t sequence_number,
int64_t now_ms) {
// Check if |packet| is second packet of a stream restart.
if (received_seq_out_of_order_) {
// Count the previous packet as a received; it was postponed below.
--cumulative_loss_;
uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
received_seq_out_of_order_ = absl::nullopt;
if (packet.SequenceNumber() == expected_sequence_number) {
// Ignore sequence number gap caused by stream restart for packet loss
// calculation, by setting received_seq_max_ to the sequence number just
// before the out-of-order seqno. This gives a net zero change of
// |cumulative_loss_|, for the two packets interpreted as a stream reset.
//
// Fraction loss for the next report may get a bit off, since we don't
// update last_report_seq_max_ and last_report_cumulative_loss_ in a
// consistent way.
last_report_seq_max_ = sequence_number - 2;
received_seq_max_ = sequence_number - 2;
return false;
}
}
if (std::abs(sequence_number - received_seq_max_) >
max_reordering_threshold_) {
// Sequence number gap looks too large, wait until next packet to check
// for a stream restart.
received_seq_out_of_order_ = packet.SequenceNumber();
// Postpone counting this as a received packet until we know how to update
// |received_seq_max_|, otherwise we temporarily decrement
// |cumulative_loss_|. The
// ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
// |cumulative_loss_| to be unchanged by the reception of the first packet
// after stream reset.
++cumulative_loss_;
return true;
}
if (sequence_number > received_seq_max_)
return false;
// Old out of order packet, may be retransmit.
if (enable_retransmit_detection_ && IsRetransmitOfOldPacket(packet, now_ms))
receive_counters_.retransmitted.AddPacket(packet);
return true;
}
重传包判断逻辑
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bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
const RtpPacketReceived& packet,
int64_t now_ms) const {
uint32_t frequency_khz = packet.payload_type_frequency() / 1000;
RTC_DCHECK_GT(frequency_khz, 0);
int64_t time_diff_ms = now_ms - last_receive_time_ms_;
// Diff in time stamp since last received in order.
uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_;
uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
int64_t max_delay_ms = 0;
// Jitter standard deviation in samples.
float jitter_std = std::sqrt(static_cast<float>(jitter_q4_ >> 4));
// 2 times the standard deviation => 95% confidence.
// And transform to milliseconds by dividing by the frequency in kHz.
max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
// Min max_delay_ms is 1.
if (max_delay_ms == 0) {
max_delay_ms = 1;
}
return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
}

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