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[feat] add timestamp to sender report

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This commit is contained in:
dijunkun
2025-02-21 17:36:32 +08:00
parent a39d0f6652
commit d1177747fd
8 changed files with 48 additions and 32 deletions
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49
src/channel/rtp_channel/rtp_video_sender.cpp
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@@ -44,14 +44,21 @@ RtpVideoSender::~RtpVideoSender() {
}
void RtpVideoSender::Enqueue(
std::vector<std::shared_ptr<RtpPacket>>& rtp_packets) {
std::vector<std::shared_ptr<RtpPacket>>& rtp_packets,
int64_t capture_timestamp) {
if (!rtp_statistics_) {
rtp_statistics_ = std::make_unique<RtpStatistics>();
rtp_statistics_->Start();
}
for (auto& rtp_packet : rtp_packets) {
rtp_packe_queue_.push(std::move(rtp_packet));
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send =
std::dynamic_pointer_cast<webrtc::RtpPacketToSend>(rtp_packet);
rtp_packet_to_send->set_capture_time(
webrtc::Timestamp::Millis(capture_timestamp));
rtp_packet_to_send->set_transport_sequence_number(transport_seq_++);
rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
rtp_packet_queue_.push(std::move(rtp_packet_to_send));
}
}
@@ -65,34 +72,32 @@ void RtpVideoSender::SetOnSentPacketFunc(
on_sent_packet_func_ = on_sent_packet_func;
}
int RtpVideoSender::SendRtpPacket(std::shared_ptr<RtpPacket> rtp_packet) {
int RtpVideoSender::SendRtpPacket(
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send) {
if (!data_send_func_) {
LOG_ERROR("data_send_func_ is nullptr");
return -1;
}
if (on_sent_packet_func_) {
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send =
std::dynamic_pointer_cast<webrtc::RtpPacketToSend>(rtp_packet);
rtp_packet_to_send->set_transport_sequence_number(transport_seq_++);
rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
on_sent_packet_func_(*rtp_packet_to_send);
rtp_packet_history_->AddPacket(rtp_packet_to_send, clock_->CurrentTime());
}
if (0 != data_send_func_((const char*)rtp_packet->Buffer().data(),
rtp_packet->Size())) {
last_rtp_timestamp_ = rtp_packet_to_send->capture_time().ms();
if (0 != data_send_func_((const char*)rtp_packet_to_send->Buffer().data(),
rtp_packet_to_send->Size())) {
// LOG_ERROR("Send rtp packet failed");
return -1;
}
#ifdef SAVE_RTP_SENT_STREAM
fwrite((unsigned char*)rtp_packet->Payload(), 1, rtp_packet->PayloadSize(),
file_rtp_sent_);
fwrite((unsigned char*)rtp_packet_to_send->Payload(), 1,
rtp_packet_to_send->PayloadSize(), file_rtp_sent_);
#endif
last_send_bytes_ += (uint32_t)rtp_packet->Size();
total_rtp_payload_sent_ += (uint32_t)rtp_packet->PayloadSize();
last_send_bytes_ += (uint32_t)rtp_packet_to_send->Size();
total_rtp_payload_sent_ += (uint32_t)rtp_packet_to_send->PayloadSize();
total_rtp_packets_sent_++;
if (io_statistics_) {
@@ -103,13 +108,17 @@ int RtpVideoSender::SendRtpPacket(std::shared_ptr<RtpPacket> rtp_packet) {
if (CheckIsTimeSendSR()) {
SenderReport rtcp_sr;
rtcp_sr.SetSenderSsrc(ssrc_);
rtcp_sr.SetTimestamp(0);
rtcp_sr.SetNtpTimestamp(0);
uint32_t rtp_timestamp =
last_rtp_timestamp_ +
((clock_->CurrentTime().us() + 500) / 1000 - last_frame_capture_time_) *
rtp::kVideoPayloadTypeFrequency;
rtcp_sr.SetTimestamp(rtp_timestamp);
rtcp_sr.SetNtpTimestamp((uint64_t)clock_->CurrentNtpTime());
rtcp_sr.SetSenderPacketCount(total_rtp_packets_sent_);
rtcp_sr.SetSenderOctetCount(total_rtp_payload_sent_);
RtcpReportBlock report;
report.SetMediaSsrc(ssrc_);
report.SetFractionLost(0);
report.SetCumulativeLost(0);
@@ -162,11 +171,11 @@ bool RtpVideoSender::Process() {
last_send_bytes_ = 0;
for (size_t i = 0; i < 10; i++)
if (!rtp_packe_queue_.isEmpty()) {
std::shared_ptr<RtpPacket> rtp_packet;
pop_success = rtp_packe_queue_.pop(rtp_packet);
if (!rtp_packet_queue_.isEmpty()) {
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send;
pop_success = rtp_packet_queue_.pop(rtp_packet_to_send);
if (pop_success) {
SendRtpPacket(rtp_packet);
SendRtpPacket(rtp_packet_to_send);
}
}