Add overhead per packet observer to the rtp_sender.

BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2495553002
Cr-Commit-Position: refs/heads/master@{#15124}

webrtc/common_types.h

@@ -299,6 +299,14 @@ class SendPacketObserver {
                             uint32_t ssrc) = 0;
 };
 
+// Callback, used to notify an observer when the overhead per packet
+// has changed.
+class OverheadObserver {
+ public:
+  virtual ~OverheadObserver() = default;
+  virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0;
+};
+
 // ==================================================================
 // Voice specific types
 // ==================================================================

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

@@ -27,6 +27,7 @@
 namespace webrtc {
 
 // Forward declarations.
+class OverheadObserver;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
@@ -89,6 +90,7 @@ class RtpRtcp : public Module {
     RtcEventLog* event_log = nullptr;
     SendPacketObserver* send_packet_observer = nullptr;
     RateLimiter* retransmission_rate_limiter = nullptr;
+    OverheadObserver* overhead_observer = nullptr;
 
    private:
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -82,7 +82,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
                   configuration.send_side_delay_observer,
                   configuration.event_log,
                   configuration.send_packet_observer,
-                  configuration.retransmission_rate_limiter),
+                  configuration.retransmission_rate_limiter,
+                  configuration.overhead_observer),
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
@@ -460,6 +461,7 @@ void ModuleRtpRtcpImpl::SetTransportOverhead(
   RTC_DCHECK_LT(transport_overhead_per_packet, mtu);
   size_t max_payload_length = mtu - transport_overhead_per_packet;
   packet_overhead_ = transport_overhead_per_packet;
+  rtp_sender_.SetTransportOverhead(packet_overhead_);
   rtcp_sender_.SetMaxPayloadLength(max_payload_length);
   rtp_sender_.SetMaxPayloadLength(max_payload_length);
 }

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -77,7 +77,8 @@ RTPSender::RTPSender(
     SendSideDelayObserver* send_side_delay_observer,
     RtcEventLog* event_log,
     SendPacketObserver* send_packet_observer,
-    RateLimiter* retransmission_rate_limiter)
+    RateLimiter* retransmission_rate_limiter,
+    OverheadObserver* overhead_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion?
       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
@@ -119,7 +120,10 @@ RTPSender::RTPSender(
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
-      retransmission_rate_limiter_(retransmission_rate_limiter) {
+      transport_overhead_bytes_per_packet_(0),
+      rtp_overhead_bytes_per_packet_(0),
+      retransmission_rate_limiter_(retransmission_rate_limiter),
+      overhead_observer_(overhead_observer) {
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
   ssrc_rtx_ = ssrc_db_->CreateSSRC();
@@ -571,18 +575,15 @@ size_t RTPSender::DeprecatedSendPadData(size_t bytes,
           kTimestampTicksPerMs * (now_ms - capture_time_ms));
     }
     padding_packet.SetExtension<AbsoluteSendTime>(now_ms);
-
     PacketOptions options;
-    bool has_transport_seq_no =
+    bool has_transport_seq_num =
         UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
-
     padding_packet.SetPadding(padding_bytes_in_packet, &random_);
 
-    if (has_transport_seq_no && transport_feedback_observer_)
+    if (has_transport_seq_num) {
-      transport_feedback_observer_->AddPacket(
+      AddPacketToTransportFeedback(options.packet_id, padding_packet,
-          options.packet_id,
-          padding_packet.payload_size() + padding_packet.padding_size(),
                                    probe_cluster_id);
+    }
 
     if (!SendPacketToNetwork(padding_packet, options))
       break;
@@ -640,6 +641,7 @@ bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
                                     const PacketOptions& options) {
   int bytes_sent = -1;
   if (transport_) {
+    UpdateRtpOverhead(packet);
     bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
                      ? static_cast<int>(packet.size())
                      : -1;
@@ -755,11 +757,8 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
   packet_to_send->SetExtension<AbsoluteSendTime>(now_ms);
 
   PacketOptions options;
-  if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id) &&
+  if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
-      transport_feedback_observer_) {
+    AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
-    transport_feedback_observer_->AddPacket(
-        options.packet_id,
-        packet_to_send->payload_size() + packet_to_send->padding_size(),
                                  probe_cluster_id);
   }
 
@@ -889,10 +888,8 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
   }
 
   PacketOptions options;
-  if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id) &&
+  if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
-      transport_feedback_observer_) {
+    AddPacketToTransportFeedback(options.packet_id, *packet.get(),
-    transport_feedback_observer_->AddPacket(
-        options.packet_id, packet->payload_size() + packet->padding_size(),
                                  PacketInfo::kNotAProbe);
   }
 
@@ -1280,4 +1277,47 @@ RtpState RTPSender::GetRtxRtpState() const {
   return state;
 }
 
+void RTPSender::SetTransportOverhead(int transport_overhead) {
+  if (!overhead_observer_)
+    return;
+  size_t overhead_bytes_per_packet = 0;
+  {
+    rtc::CritScope lock(&send_critsect_);
+    if (transport_overhead_bytes_per_packet_ ==
+        static_cast<size_t>(transport_overhead)) {
+      return;
+    }
+    transport_overhead_bytes_per_packet_ = transport_overhead;
+    overhead_bytes_per_packet =
+        rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
+  }
+  overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
+}
+
+void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
+                                             const RtpPacketToSend& packet,
+                                             int probe_cluster_id) {
+  if (transport_feedback_observer_) {
+    transport_feedback_observer_->AddPacket(
+        packet_id, packet.payload_size() + packet.padding_size(),
+        probe_cluster_id);
+  }
+}
+
+void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
+  if (!overhead_observer_)
+    return;
+  size_t overhead_bytes_per_packet = 0;
+  {
+    rtc::CritScope lock(&send_critsect_);
+    if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
+      return;
+    }
+    rtp_overhead_bytes_per_packet_ = packet.headers_size();
+    overhead_bytes_per_packet =
+        rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
+  }
+  overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
+}
+
 }  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtp_sender.h

@@ -36,6 +36,7 @@
 
 namespace webrtc {
 
+class OverheadObserver;
 class RateLimiter;
 class RtcEventLog;
 class RtpPacketToSend;
@@ -58,7 +59,8 @@ class RTPSender {
             SendSideDelayObserver* send_side_delay_observer,
             RtcEventLog* event_log,
             SendPacketObserver* send_packet_observer,
-            RateLimiter* nack_rate_limiter);
+            RateLimiter* nack_rate_limiter,
+            OverheadObserver* overhead_observer);
 
   ~RTPSender();
 
@@ -214,6 +216,8 @@ class RTPSender {
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
 
+  void SetTransportOverhead(int transport_overhead);
+
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
@@ -259,6 +263,12 @@ class RTPSender {
                       bool is_retransmit);
   bool IsFecPacket(const RtpPacketToSend& packet) const;
 
+  void AddPacketToTransportFeedback(uint16_t packet_id,
+                                    const RtpPacketToSend& packet,
+                                    int probe_cluster_id);
+
+  void UpdateRtpOverhead(const RtpPacketToSend& packet);
+
   Clock* const clock_;
   const int64_t clock_delta_ms_;
   Random random_ GUARDED_BY(send_critsect_);
@@ -327,8 +337,11 @@ class RTPSender {
   uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
   // Mapping rtx_payload_type_map_[associated] = rtx.
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
+  size_t transport_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
+  size_t rtp_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
+  OverheadObserver* overhead_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -150,7 +150,7 @@ class RtpSenderTest : public ::testing::Test {
         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
         nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
         &mock_rtc_event_log_, &send_packet_observer_,
-        &retransmission_rate_limiter_));
+        &retransmission_rate_limiter_, nullptr));
     rtp_sender_->SetSendPayloadType(kPayload);
     rtp_sender_->SetSequenceNumber(kSeqNum);
     rtp_sender_->SetTimestampOffset(0);
@@ -444,7 +444,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
-      &send_packet_observer_, &retransmission_rate_limiter_));
+      &send_packet_observer_, &retransmission_rate_limiter_, nullptr));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
@@ -487,11 +487,11 @@ TEST_F(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
 }
 
 TEST_F(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
-  rtp_sender_.reset(
+  rtp_sender_.reset(new RTPSender(
-      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
+      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
-                    nullptr, &seq_num_allocator_, &feedback_observer_, nullptr,
+      &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, nullptr,
-                    nullptr, nullptr, &mock_rtc_event_log_,
+      &mock_rtc_event_log_, &send_packet_observer_,
-                    &send_packet_observer_, &retransmission_rate_limiter_));
+      &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   rtp_sender_->SetStorePacketsStatus(true, 10);
@@ -775,7 +775,8 @@ TEST_F(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
-      nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_));
+      nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_,
+      nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ -801,7 +802,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
       nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
-      &retransmission_rate_limiter_));
+      &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@@ -919,11 +920,11 @@ TEST_F(RtpSenderTest, SendFlexfecPackets) {
                                kNoRtpExtensions, &fake_clock_);
 
   // Reset |rtp_sender_| to use FlexFEC.
-  rtp_sender_.reset(
+  rtp_sender_.reset(new RTPSender(
-      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
+      false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
-                    &flexfec_sender, &seq_num_allocator_, nullptr, nullptr,
+      &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
-                    nullptr, nullptr, &mock_rtc_event_log_,
+      &mock_rtc_event_log_, &send_packet_observer_,
-                    &send_packet_observer_, &retransmission_rate_limiter_));
+      &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@@ -977,7 +978,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
                                   &flexfec_sender, &seq_num_allocator_, nullptr,
                                   nullptr, nullptr, nullptr,
                                   &mock_rtc_event_log_, &send_packet_observer_,
-                                  &retransmission_rate_limiter_));
+                                  &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@@ -1020,10 +1021,10 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
     FrameCounts frame_counts_;
   } callback;
 
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
+  rtp_sender_.reset(
-                                  &mock_paced_sender_, nullptr, nullptr,
+      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
-                                  nullptr, nullptr, &callback, nullptr, nullptr,
+                    nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
-                                  nullptr, &retransmission_rate_limiter_));
+                    nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
@@ -1085,7 +1086,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   nullptr, nullptr, nullptr, &callback, nullptr,
                                   nullptr, nullptr, nullptr,
-                                  &retransmission_rate_limiter_));
+                                  &retransmission_rate_limiter_, nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
   // number of packets selected so that we fill (but don't overflow) the one
@@ -1143,7 +1144,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
     rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr, nullptr, nullptr,
-                                    &retransmission_rate_limiter_));
+                                    &retransmission_rate_limiter_, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
@@ -1494,4 +1495,52 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCameraAndFlipCVO) {
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 3));
 }
 
+namespace {
+class MockOverheadObserver : public OverheadObserver {
+ public:
+  MOCK_METHOD1(OnOverheadChanged, void(size_t overhead_bytes_per_packet));
+};
+}  // namespace
+
+TEST_F(RtpSenderTest, OnOverheadChanged) {
+  MockOverheadObserver mock_overhead_observer;
+  rtp_sender_.reset(
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+                    &retransmission_rate_limiter_, &mock_overhead_observer));
+
+  // Transport overhead is set to 28B.
+  EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(28)).Times(1);
+  rtp_sender_->SetTransportOverhead(28);
+
+  // RTP overhead is 12B.
+  // 28B + 12B = 40B
+  EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(40)).Times(1);
+  SendGenericPayload();
+
+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
+                                          kTransmissionTimeOffsetExtensionId);
+
+  // TransmissionTimeOffset extension has a size of 8B.
+  // 28B + 12B + 8B = 48B
+  EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(48)).Times(1);
+  SendGenericPayload();
+}
+
+TEST_F(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
+  MockOverheadObserver mock_overhead_observer;
+  rtp_sender_.reset(
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+                    &retransmission_rate_limiter_, &mock_overhead_observer));
+
+  EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
+  rtp_sender_->SetTransportOverhead(28);
+  rtp_sender_->SetTransportOverhead(28);
+
+  EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
+  SendGenericPayload();
+  SendGenericPayload();
+}
+
 }  // namespace webrtc