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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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6
src/channel/rtp_channel/rtp_audio_sender.cpp
View File

@@ -30,7 +30,7 @@ void RtpAudioSender::Enqueue(
}
for (auto& rtp_packet : rtp_packets) {
rtp_packe_queue_.push(rtp_packet);
rtp_packet_queue_.push(rtp_packet);
}
}
@@ -139,9 +139,9 @@ bool RtpAudioSender::Process() {
last_send_bytes_ = 0;
for (size_t i = 0; i < 10; i++)
if (!rtp_packe_queue_.isEmpty()) {
if (!rtp_packet_queue_.isEmpty()) {
std::shared_ptr<RtpPacket> rtp_packet;
rtp_packe_queue_.pop(rtp_packet);
rtp_packet_queue_.pop(rtp_packet);
SendRtpPacket(rtp_packet);
}

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

@@ -38,7 +38,7 @@ class RtpAudioSender : public ThreadBase {
private:
std::function<int(const char *, size_t)> data_send_func_ = nullptr;
RingBuffer<std::shared_ptr<RtpPacket>> rtp_packe_queue_;
RingBuffer<std::shared_ptr<RtpPacket>> rtp_packet_queue_;
private:
uint32_t ssrc_ = 0;

6
src/channel/rtp_channel/rtp_data_sender.cpp
View File

@@ -30,7 +30,7 @@ void RtpDataSender::Enqueue(
}
for (auto& rtp_packet : rtp_packets) {
rtp_packe_queue_.push(rtp_packet);
rtp_packet_queue_.push(rtp_packet);
}
}
@@ -139,9 +139,9 @@ bool RtpDataSender::Process() {
last_send_bytes_ = 0;
for (size_t i = 0; i < 10; i++)
if (!rtp_packe_queue_.isEmpty()) {
if (!rtp_packet_queue_.isEmpty()) {
std::shared_ptr<RtpPacket> rtp_packet;
rtp_packe_queue_.pop(rtp_packet);
rtp_packet_queue_.pop(rtp_packet);
SendRtpPacket(rtp_packet);
}

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

@@ -39,7 +39,7 @@ class RtpDataSender : public ThreadBase {
private:
std::function<int(const char *, size_t)> data_send_func_ = nullptr;
RingBuffer<std::shared_ptr<RtpPacket>> rtp_packe_queue_;
RingBuffer<std::shared_ptr<RtpPacket>> rtp_packet_queue_;
private:
uint32_t ssrc_ = 0;

49
src/channel/rtp_channel/rtp_video_sender.cpp
View File

@@ -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);
}
}

11
src/channel/rtp_channel/rtp_video_sender.h
View File

@@ -22,14 +22,16 @@ class RtpVideoSender : public ThreadBase {
virtual ~RtpVideoSender();
public:
void Enqueue(std::vector<std::shared_ptr<RtpPacket>> &rtp_packets);
void Enqueue(std::vector<std::shared_ptr<RtpPacket>> &rtp_packets,
int64_t capture_timestamp);
void SetSendDataFunc(std::function<int(const char *, size_t)> data_send_func);
void SetOnSentPacketFunc(
std::function<void(const webrtc::RtpPacketToSend &)> on_sent_packet_func);
uint32_t GetSsrc() { return ssrc_; }
private:
int SendRtpPacket(std::shared_ptr<RtpPacket> rtp_packet);
int SendRtpPacket(
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send);
int SendRtcpSR(SenderReport &rtcp_sr);
bool CheckIsTimeSendSR();
@@ -41,7 +43,7 @@ class RtpVideoSender : public ThreadBase {
std::function<int(const char *, size_t)> data_send_func_ = nullptr;
std::function<void(const webrtc::RtpPacketToSend &)> on_sent_packet_func_ =
nullptr;
RingBuffer<std::shared_ptr<RtpPacket>> rtp_packe_queue_;
RingBuffer<std::shared_ptr<webrtc::RtpPacketToSend>> rtp_packet_queue_;
private:
uint32_t ssrc_ = 0;
@@ -54,6 +56,9 @@ class RtpVideoSender : public ThreadBase {
uint32_t total_rtp_payload_sent_ = 0;
uint32_t total_rtp_packets_sent_ = 0;
uint32_t last_rtp_timestamp_ = 0;
int64_t last_frame_capture_time_ = 0;
private:
int64_t transport_seq_ = 0;

2
src/channel/video_channel_send.cpp
View File

@@ -63,7 +63,7 @@ int VideoChannelSend::SendVideo(
std::vector<std::shared_ptr<RtpPacket>> rtp_packets =
rtp_packetizer_->Build((uint8_t*)encoded_frame->Buffer(),
(uint32_t)encoded_frame->Size(), true);
rtp_video_sender_->Enqueue(rtp_packets);
rtp_video_sender_->Enqueue(rtp_packets, encoded_frame->CaptureTimestamp());
}
return 0;

2
src/rtp/rtp_packet/rtp_defines.h
View File

@@ -39,5 +39,7 @@ typedef struct {
} FU_HEADER;
typedef enum { UNKNOWN = 0, NALU = 1, FU_A = 28, FU_B = 29 } NAL_UNIT_TYPE;
const int kVideoPayloadTypeFrequency = 90000;
} // namespace rtp
#endif