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[fix] fix receiver report building and parsing

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This commit is contained in:
dijunkun
2025-03-04 11:17:19 +08:00
parent fba5993866
commit ebfeaf4754
15 changed files with 140 additions and 60 deletions
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2
src/channel/rtp_channel/rtp_audio_sender.h
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@@ -10,6 +10,7 @@
#include <functional>
#include "io_statistics.h"
#include "receiver_report.h"
#include "ringbuffer.h"
#include "rtp_packet.h"
#include "rtp_statistics.h"
@@ -26,6 +27,7 @@ class RtpAudioSender : public ThreadBase {
void Enqueue(std::vector<std::shared_ptr<RtpPacket>> rtp_packets);
void SetSendDataFunc(std::function<int(const char *, size_t)> data_send_func);
uint32_t GetSsrc() { return ssrc_; }
void OnReceiverReport(const ReceiverReport &receiver_report) {}
private:
int SendRtpPacket(std::shared_ptr<RtpPacket> rtp_packet);

2
src/channel/rtp_channel/rtp_data_sender.h
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@@ -10,6 +10,7 @@
#include <functional>
#include "io_statistics.h"
#include "receiver_report.h"
#include "ringbuffer.h"
#include "rtp_packet.h"
#include "rtp_statistics.h"
@@ -26,6 +27,7 @@ class RtpDataSender : public ThreadBase {
void Enqueue(std::vector<std::shared_ptr<RtpPacket>> rtp_packets);
void SetSendDataFunc(std::function<int(const char *, size_t)> data_send_func);
uint32_t GetSsrc() { return ssrc_; }
void OnReceiverReport(const ReceiverReport &receiver_report) {}
private:
private:

106
src/channel/rtp_channel/rtp_video_receiver.cpp
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@@ -101,57 +101,39 @@ void RtpVideoReceiver::InsertRtpPacket(RtpPacket& rtp_packet) {
webrtc::Timestamp now = clock_->CurrentTime();
remote_ssrc_ = rtp_packet.Ssrc();
uint16_t sequence_number = rtp_packet.SequenceNumber();
if (last_receive_time_.has_value() == 0) {
extended_high_seq_num_ = sequence_number;
--cumulative_loss_;
if (!last_receive_time_.has_value()) {
last_extended_high_seq_num_ = sequence_number - 1;
extended_high_seq_num_ = sequence_number - 1;
}
cumulative_loss_ += sequence_number - extended_high_seq_num_;
extended_high_seq_num_ = sequence_number;
// Calculate fraction lost.
int64_t exp_since_last = extended_high_seq_num_ - last_extended_high_seq_num_;
int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
if (exp_since_last > 0 && lost_since_last > 0) {
// Scale 0 to 255, where 255 is 100% loss.
fraction_lost_ = 255 * lost_since_last / exp_since_last;
}
cumulative_lost_ = cumulative_loss_ + cumulative_loss_rtcp_offset_;
if (cumulative_lost_ < 0) {
// Clamp to zero. Work around to accommodate for senders that misbehave with
// negative cumulative loss.
cumulative_lost_ = 0;
cumulative_loss_rtcp_offset_ = -cumulative_loss_;
}
if (cumulative_lost_ > 0x7fffff) {
// Packets lost is a 24 bit signed field, and thus should be clamped, as
// described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
cumulative_lost_ = 0x7fffff;
if (rtp_packet_received.Timestamp() != last_received_timestamp_) {
webrtc::TimeDelta receive_diff = now - *last_receive_time_;
uint32_t receive_diff_rtp =
(receive_diff * rtp_packet_received.payload_type_frequency())
.seconds<uint32_t>();
int32_t time_diff_samples =
receive_diff_rtp -
(rtp_packet_received.Timestamp() - last_received_timestamp_);
ReviseFrequencyAndJitter(rtp_packet_received.payload_type_frequency());
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
jitter_ = jitter_q4_ >> 4;
}
webrtc::TimeDelta receive_diff = now - *last_receive_time_;
uint32_t receive_diff_rtp =
(receive_diff * rtp_packet_received.payload_type_frequency())
.seconds<uint32_t>();
int32_t time_diff_samples =
receive_diff_rtp -
(rtp_packet_received.Timestamp() - last_received_timestamp_);
ReviseFrequencyAndJitter(rtp_packet_received.payload_type_frequency());
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
jitter_ = jitter_q4_ >> 4;
last_extended_high_seq_num_ = extended_high_seq_num_;
last_report_cumulative_loss_ = cumulative_loss_;
last_received_timestamp_ = rtp_packet_received.Timestamp();
last_receive_time_ = now;
@@ -592,24 +574,50 @@ void RtpVideoReceiver::ReviseFrequencyAndJitter(int payload_type_frequency) {
void RtpVideoReceiver::SendRR() {
uint32_t now = CompactNtp(clock_->CurrentNtpTime());
// Calculate fraction lost.
int64_t exp_since_last = extended_high_seq_num_ - last_extended_high_seq_num_;
int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
if (exp_since_last > 0 && lost_since_last > 0) {
// Scale 0 to 255, where 255 is 100% loss.
fraction_lost_ = 255 * lost_since_last / exp_since_last;
} else {
fraction_lost_ = 0;
}
cumulative_lost_ = cumulative_loss_ + cumulative_loss_rtcp_offset_;
if (cumulative_lost_ < 0) {
// Clamp to zero. Work around to accommodate for senders that misbehave with
// negative cumulative loss.
cumulative_lost_ = 0;
cumulative_loss_rtcp_offset_ = -cumulative_loss_;
}
if (cumulative_lost_ > 0x7fffff) {
// Packets lost is a 24 bit signed field, and thus should be clamped, as
// described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
cumulative_lost_ = 0x7fffff;
}
uint32_t receive_time = last_arrival_ntp_timestamp;
uint32_t delay_since_last_sr = now - receive_time;
ReceiverReport rtcp_rr;
RtcpReportBlock report;
report.SetMediaSsrc(ssrc_);
report.SetMediaSsrc(fraction_lost_);
report.SetMediaSsrc(remote_ssrc_);
report.SetFractionLost(fraction_lost_);
report.SetCumulativeLost(cumulative_lost_);
report.SetExtHighestSeqNum(extended_high_seq_num_);
report.SetJitter(jitter_);
report.SetLastSr(last_remote_ntp_timestamp);
report.SetDelayLastSr(delay_since_last_sr);
rtcp_rr.SetSenderSsrc(ssrc_);
rtcp_rr.SetReportBlock(report);
rtcp_rr.Build();
SendRtcpRR(rtcp_rr);
last_extended_high_seq_num_ = extended_high_seq_num_;
last_report_cumulative_loss_ = cumulative_loss_;
}
void RtpVideoReceiver::RtcpThread() {
@@ -626,7 +634,7 @@ void RtpVideoReceiver::RtcpThread() {
auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
now - last_send_rtcp_rr_ts_)
.count();
if (elapsed >= rtcp_rr_interval_ms_) {
if (elapsed >= rtcp_rr_interval_ms_ && last_receive_time_.has_value()) {
SendRR();
last_send_rtcp_rr_ts_ = now;
}

12
src/channel/rtp_channel/rtp_video_sender.cpp
View File

@@ -173,4 +173,16 @@ bool RtpVideoSender::Process() {
}
return true;
}
void RtpVideoSender::OnReceiverReport(const ReceiverReport& receiver_report) {
std::vector<RtcpReportBlock> reports = receiver_report.GetReportBlocks();
for (auto r : reports) {
LOG_WARN(
"r_ssrc [{}], f_lost [{}], c_lost [{}], h_seq [{}], jitter [{}], "
"lsr [{}], dlsr [{}] ",
r.SourceSsrc(), r.FractionLost() / 255.0, r.CumulativeLost(),
r.ExtendedHighSeqNum(), r.Jitter(), r.LastSr(), r.DelaySinceLastSr());
}
}

2
src/channel/rtp_channel/rtp_video_sender.h
View File

@@ -6,6 +6,7 @@
#include "api/clock/clock.h"
#include "clock/system_clock.h"
#include "io_statistics.h"
#include "receiver_report.h"
#include "ringbuffer.h"
#include "rtp_packet.h"
#include "rtp_packet_history.h"
@@ -28,6 +29,7 @@ class RtpVideoSender : public ThreadBase {
void SetOnSentPacketFunc(
std::function<void(const webrtc::RtpPacketToSend &)> on_sent_packet_func);
uint32_t GetSsrc() { return ssrc_; }
void OnReceiverReport(const ReceiverReport &receiver_report);
private:
int SendRtpPacket(