[fix] fix video frame capture timestamp

src/common/clock/system_clock.cpp

@@ -68,12 +68,12 @@ int64_t SystemClock::CurrentTimeNs() {
   return ticks;
 }
 
+int64_t SystemClock::CurrentTime() { return CurrentTimeNs() / 1000LL; }
+
 int64_t SystemClock::CurrentTimeUs() { return CurrentTimeNs() / 1000LL; }
 
 int64_t SystemClock::CurrentTimeMs() { return CurrentTimeNs() / 1000000LL; }
 
-int64_t SystemClock::CurrentTime() { return CurrentTimeNs() / 1000000000LL; }
-
 int64_t SystemClock::CurrentNtpTime() {
   return ConvertToNtpTime(CurrentTimeNs());
 }

src/common/clock/system_clock.h

@@ -18,8 +18,8 @@ class SystemClock {
   ~SystemClock() = default;
 
   int64_t CurrentTime();
-  int64_t CurrentTimeMs();
   int64_t CurrentTimeUs();
+  int64_t CurrentTimeMs();
   int64_t CurrentTimeNs();
 
   int64_t CurrentNtpTime();

src/frame/video_frame_wrapper.h

@@ -23,10 +23,10 @@ class VideoFrameWrapper : public VideoFrame {
   VideoFrameWrapper() = delete;
   ~VideoFrameWrapper() = default;
 
-  int64_t CaptureTimestamp() { return capture_timestamp_ms_; }
+  int64_t CaptureTimestamp() { return capture_timestamp_us_; }
 
-  void SetCaptureTimestamp(int64_t capture_timestamp_ms) {
+  void SetCaptureTimestamp(int64_t capture_timestamp_us) {
-    capture_timestamp_ms_ = capture_timestamp_ms;
+    capture_timestamp_us_ = capture_timestamp_us;
   }
 
   VideoFrameType FrameType() { return frame_type_; }
@@ -46,7 +46,7 @@ class VideoFrameWrapper : public VideoFrame {
   }
 
  private:
-  int64_t capture_timestamp_ms_ = 0;
+  int64_t capture_timestamp_us_ = 0;
   VideoFrameType frame_type_ = VideoFrameType::kVideoFrameDelta;
   uint32_t encoded_width_ = 0;
   uint32_t encoded_height_ = 0;

src/interface/x.h

@@ -134,7 +134,7 @@ DLLAPI int SendAudioFrame(PeerPtr* peer_ptr, const char* data, size_t size);
 
 DLLAPI int SendDataFrame(PeerPtr* peer_ptr, const char* data, size_t size);
 
-DLLAPI int64_t CurrentNtpTimeMs(PeerPtr* peer_ptr);
+DLLAPI int64_t GetSystemTimeMicros(PeerPtr* peer_ptr);
 
 #ifdef __cplusplus
 }

src/pc/peer_connection.cpp

@@ -390,9 +390,9 @@ int PeerConnection::SendDataFrame(const char *data, size_t size) {
   return 0;
 }
 
-int64_t PeerConnection::CurrentNtpTimeMs() {
+int64_t PeerConnection::GetSystemTimeMicros() {
   if (clock_) {
-    return clock_->CurrentNtpTimeMs();
+    return clock_->CurrentTimeUs();
   }
   return 0;
 }

src/pc/peer_connection.h

@@ -108,7 +108,7 @@ class PeerConnection {
   int SendAudioFrame(const char *data, size_t size);
   int SendDataFrame(const char *data, size_t size);
 
-  int64_t CurrentNtpTimeMs();
+  int64_t GetSystemTimeMicros();
 
  private:
   int Login();

src/rtc/x_inner.cpp

@@ -149,10 +149,10 @@ int SendDataFrame(PeerPtr *peer_ptr, const char *data, size_t size) {
   return 0;
 }
 
-int64_t CurrentNtpTimeMs(PeerPtr *peer_ptr) {
+int64_t GetSystemTimeMicros(PeerPtr *peer_ptr) {
   if (!peer_ptr || !peer_ptr->peer_connection) {
     LOG_ERROR("Peer connection not created");
     return -1;
   }
-  return peer_ptr->peer_connection->CurrentNtpTimeMs();
+  return peer_ptr->peer_connection->GetSystemTimeMicros();
 }

src/rtp/rtp_packetizer/rtp_packetizer.h

@@ -22,11 +22,11 @@ class RtpPacketizer {
   virtual ~RtpPacketizer() = default;
 
   virtual std::vector<std::unique_ptr<RtpPacket>> Build(
-      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) = 0;
 
   virtual std::vector<std::unique_ptr<RtpPacket>> BuildPadding(
-      uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) = 0;
 };
 

src/rtp/rtp_packetizer/rtp_packetizer_av1.cpp

@@ -5,7 +5,7 @@ RtpPacketizerAv1::RtpPacketizerAv1(uint32_t ssrc) {}
 RtpPacketizerAv1::~RtpPacketizerAv1() {}
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerAv1::Build(
-    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
 

src/rtp/rtp_packetizer/rtp_packetizer_av1.h

@@ -16,11 +16,11 @@ class RtpPacketizerAv1 : public RtpPacketizer {
   virtual ~RtpPacketizerAv1();
 
   std::vector<std::unique_ptr<RtpPacket>> Build(
-      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override;
 
   std::vector<std::unique_ptr<RtpPacket>> BuildPadding(
-      uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override {
     return std::vector<std::unique_ptr<RtpPacket>>{};
   };

src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp

@@ -47,7 +47,7 @@ void RtpPacketizerGeneric::AddAbsSendTimeExtension(
 }
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerGeneric::Build(
-    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   uint32_t last_packet_size = payload_size % MAX_NALU_LEN;
   uint32_t packet_num =

src/rtp/rtp_packetizer/rtp_packetizer_generic.h

@@ -16,11 +16,11 @@ class RtpPacketizerGeneric : public RtpPacketizer {
   virtual ~RtpPacketizerGeneric();
 
   std::vector<std::unique_ptr<RtpPacket>> Build(
-      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override;
 
   std::vector<std::unique_ptr<RtpPacket>> BuildPadding(
-      uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override {
     return std::vector<std::unique_ptr<RtpPacket>>{};
   };

src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp

@@ -61,19 +61,19 @@ void RtpPacketizerH264::AddAbsSendTimeExtension(
 }
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::Build(
-    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   if (payload_size <= MAX_NALU_LEN) {
-    return BuildNalu(payload, payload_size, capture_timestamp_ms,
+    return BuildNalu(payload, payload_size, capture_timestamp_us,
                      use_rtp_packet_to_send);
   } else {
-    return BuildFua(payload, payload_size, capture_timestamp_ms,
+    return BuildFua(payload, payload_size, capture_timestamp_us,
                     use_rtp_packet_to_send);
   }
 }
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
-    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
 
@@ -84,7 +84,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
   marker_ = 1;
   payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
   sequence_number_++;
-  timestamp_ = kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
+  timestamp_ = kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_us);
 
   if (!csrc_count_) {
   }
@@ -142,7 +142,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
 }
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
-    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
 
@@ -242,7 +242,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
 }
 
 std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildPadding(
-    uint32_t payload_size, int64_t capture_timestamp_ms,
+    uint32_t payload_size, int64_t capture_timestamp_us,
     bool use_rtp_packet_to_send) {
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
 
@@ -259,7 +259,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildPadding(
     uint8_t payload_type = rtp::PAYLOAD_TYPE(payload_type_ - 1);
     sequence_number_++;
     timestamp_ =
-        kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
+        kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_us);
 
     rtp_packet_frame_.clear();
     rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |

src/rtp/rtp_packetizer/rtp_packetizer_h264.h

@@ -16,20 +16,20 @@ class RtpPacketizerH264 : public RtpPacketizer {
   virtual ~RtpPacketizerH264();
 
   std::vector<std::unique_ptr<RtpPacket>> Build(
-      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override;
 
   std::vector<std::unique_ptr<RtpPacket>> BuildNalu(
-      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint8_t* payload, uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send);
 
   std::vector<std::unique_ptr<RtpPacket>> BuildFua(uint8_t* payload,
                                                    uint32_t payload_size,
-                                                   int64_t capture_timestamp_ms,
+                                                   int64_t capture_timestamp_us,
                                                    bool use_rtp_packet_to_send);
 
   std::vector<std::unique_ptr<RtpPacket>> BuildPadding(
-      uint32_t payload_size, int64_t capture_timestamp_ms,
+      uint32_t payload_size, int64_t capture_timestamp_us,
       bool use_rtp_packet_to_send) override;
 
  private:

src/transport/channel/rtp_video_sender.cpp

@@ -45,7 +45,7 @@ RtpVideoSender::~RtpVideoSender() {
 
 void RtpVideoSender::Enqueue(
     std::vector<std::unique_ptr<RtpPacket>>& rtp_packets,
-    int64_t capture_timestamp_ms) {
+    int64_t capture_timestamp_us) {
   if (!rtp_statistics_) {
     rtp_statistics_ = std::make_unique<RtpStatistics>();
     rtp_statistics_->Start();

src/transport/channel/rtp_video_sender.h

@@ -24,7 +24,7 @@ class RtpVideoSender : public ThreadBase {
 
  public:
   void Enqueue(std::vector<std::unique_ptr<RtpPacket>> &rtp_packets,
-               int64_t capture_timestamp_ms);
+               int64_t capture_timestamp_us);
   void SetSendDataFunc(std::function<int(const char *, size_t)> data_send_func);
   void SetOnSentPacketFunc(
       std::function<void(const webrtc::RtpPacketToSend &)> on_sent_packet_func);

src/transport/channel/video_channel_send.cpp

@@ -57,9 +57,9 @@ void VideoChannelSend::SetEnqueuePacketsFunc(
 }
 
 std::vector<std::unique_ptr<RtpPacket>> VideoChannelSend::GeneratePadding(
-    uint32_t payload_size, int64_t capture_timestamp_ms) {
+    uint32_t payload_size, int64_t capture_timestamp_us) {
   if (rtp_packetizer_) {
-    return rtp_packetizer_->BuildPadding(payload_size, capture_timestamp_ms,
+    return rtp_packetizer_->BuildPadding(payload_size, capture_timestamp_us,
                                          true);
   }
   return std::vector<std::unique_ptr<RtpPacket>>{};

src/transport/channel/video_channel_send.h

@@ -36,7 +36,7 @@ class VideoChannelSend {
           enqueue_packets_func);
 
   std::vector<std::unique_ptr<RtpPacket>> GeneratePadding(
-      uint32_t payload_size, int64_t capture_timestamp_ms);
+      uint32_t payload_size, int64_t capture_timestamp_us);
 
   int64_t GetTransportSeqAndIncrement() {
     int64_t transport_seq = rtp_video_sender_->GetTransportSequenceNumber();

src/transport/ice_transport_controller.cpp

@@ -75,9 +75,9 @@ void IceTransportController::Create(
       });
 
   packet_sender_->SetGeneratePaddingFunc(
-      [this](uint32_t size, int64_t capture_timestamp_ms)
+      [this](uint32_t size, int64_t capture_timestamp_us)
           -> std::vector<std::unique_ptr<RtpPacket>> {
-        return video_channel_send_->GeneratePadding(size, capture_timestamp_ms);
+        return video_channel_send_->GeneratePadding(size, capture_timestamp_us);
       });
 
   audio_channel_send_ = std::make_unique<AudioChannelSend>(
@@ -170,6 +170,7 @@ int IceTransportController::SendVideo(const XVideoFrame* video_frame) {
   new_frame.width = video_frame->width;
   new_frame.height = video_frame->height;
   new_frame.size = video_frame->size;
+  new_frame.timestamp = video_frame->timestamp;
   if (target_width_.has_value() && target_height_.has_value()) {
     if (target_width_.value() < video_frame->width &&
         target_height_.value() < video_frame->height) {

src/transport/packet_sender/packet_sender.h

@@ -20,7 +20,7 @@ class PacketSender {
   virtual int Send() = 0;
   virtual int EnqueueRtpPacket(
       std::vector<std::unique_ptr<RtpPacket>> &rtp_packets,
-      int64_t capture_timestamp_ms) = 0;
+      int64_t capture_timestamp_us) = 0;
 };
 
 #endif

src/transport/packet_sender/packet_sender_imp.cpp

@@ -250,13 +250,13 @@ PacketSenderImp::Stats PacketSenderImp::GetStats() const {
 
 int PacketSenderImp::EnqueueRtpPacket(
     std::vector<std::unique_ptr<RtpPacket>> &rtp_packets,
-    int64_t capture_timestamp_ms) {
+    int64_t capture_timestamp_us) {
   std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> to_send_rtp_packets;
   for (auto &rtp_packet : rtp_packets) {
     std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send(
         static_cast<webrtc::RtpPacketToSend *>(rtp_packet.release()));
     rtp_packet_to_send->set_capture_time(
-        webrtc::Timestamp::Millis(capture_timestamp_ms));
+        webrtc::Timestamp::Micros(capture_timestamp_us));
     rtp_packet_to_send->set_transport_sequence_number(transport_seq_++);
 
     switch (rtp_packet_to_send->PayloadType()) {

src/transport/packet_sender/packet_sender_imp.h

@@ -38,7 +38,7 @@ class PacketSenderImp : public PacketSender,
   int Send() { return 0; }
 
   int EnqueueRtpPacket(std::vector<std::unique_ptr<RtpPacket>>& rtp_packets,
-                       int64_t capture_timestamp_ms);
+                       int64_t capture_timestamp_us);
 
   void SetOnSentPacketFunc(
       std::function<void(const webrtc::RtpPacketToSend&)> on_sent_packet_func) {
@@ -61,12 +61,6 @@ class PacketSenderImp : public PacketSender,
 
       packet->UpdateSequenceNumber(ssrc_seq_[packet->Ssrc()]++);
 
-      webrtc::Timestamp now = clock_->CurrentTime();
-      webrtc::TimeDelta interval = now - last_send_time_;
-      webrtc::TimeDelta delay = now - packet->capture_time();
-      LOG_WARN("interval: {}, delay: {}", interval.ms(), delay.seconds());
-      last_send_time_ = now;
-
       on_sent_packet_func_(*packet);
     }
   }