Add overhead per packet observer to the rtp_sender.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 131 matching lines @@
         transport_(),
         kMarkerBit(true) {}
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(
         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_->SetSequenceNumber(kSeqNum);
     rtp_sender_->SetSendPayloadType(kPayload);
     rtp_sender_->SetTimestampOffset(0);
   }
 
   SimulatedClock fake_clock_;
   testing::NiceMock<MockRtcEventLog> mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   testing::StrictMock<MockTransportSequenceNumberAllocator> seq_num_allocator_;
   testing::StrictMock<MockSendPacketObserver> send_packet_observer_;
@@ skipping 272 matching lines @@
 
   ASSERT_EQ(1u, transport_.sent_packets_.size());
   // Verify padding packet timestamp.
   EXPECT_EQ(kTimestamp, transport_.last_sent_packet().Timestamp());
 }
 
 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));
 
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
   EXPECT_CALL(send_packet_observer_,
               OnSendPacket(kTransportSequenceNumber, _, _))
       .Times(1);
   EXPECT_CALL(feedback_observer_,
@@ skipping 22 matching lines @@
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
   EXPECT_CALL(send_packet_observer_,
               OnSendPacket(kTransportSequenceNumber, _, _))
       .Times(1);
 
   SendGenericPayload();
 }
 
 TEST_F(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
-  rtp_sender_.reset(
-      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
-                    nullptr, &seq_num_allocator_, &feedback_observer_, nullptr,
-                    nullptr, nullptr, &mock_rtc_event_log_,
-                    &send_packet_observer_, &retransmission_rate_limiter_));
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
+      &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
 
   uint16_t seq_num = 0;
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
       .Times(1).WillRepeatedly(testing::SaveArg<2>(&seq_num));
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
@@ skipping 259 matching lines @@
   const bool kIsRetransmit = true;
   rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
                                 kIsRetransmit, PacketInfo::kNotAProbe);
   EXPECT_EQ(1, transport_.packets_sent());
 }
 
 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));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
   EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
 
   SendGenericPayload();  // Packet passed to pacer.
   const bool kIsRetransmit = false;
   rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
                                 kIsRetransmit, PacketInfo::kNotAProbe);
   EXPECT_EQ(1, transport_.packets_sent());
 }
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   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_->SetRtxPayloadType(kRtxPayload, kPayload);
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
@@ skipping 97 matching lines @@
 TEST_F(RtpSenderTest, SendFlexfecPackets) {
   constexpr int kMediaPayloadType = 127;
   constexpr int kFlexfecPayloadType = 118;
   constexpr uint32_t kMediaSsrc = 1234;
   constexpr uint32_t kFlexfecSsrc = 5678;
   const std::vector<RtpExtension> kNoRtpExtensions;
   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                                kNoRtpExtensions, &fake_clock_);
 
   // Reset |rtp_sender_| to use FlexFEC.
-  rtp_sender_.reset(
-      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
-                    &flexfec_sender, &seq_num_allocator_, nullptr, nullptr,
-                    nullptr, nullptr, &mock_rtc_event_log_,
-                    &send_packet_observer_, &retransmission_rate_limiter_));
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
+      &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
@@ skipping 27 matching lines @@
   constexpr uint32_t kFlexfecSsrc = 5678;
   const std::vector<RtpExtension> kNoRtpExtensions;
   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                                kNoRtpExtensions, &fake_clock_);
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   &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);
 
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
   rtp_sender_->SetFecParameters(params, params);
@@ skipping 22 matching lines @@
       ++num_calls_;
       ssrc_ = ssrc;
       frame_counts_ = frame_counts;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     FrameCounts frame_counts_;
   } callback;
 
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                  &mock_paced_sender_, nullptr, nullptr,
-                                  nullptr, nullptr, &callback, nullptr, nullptr,
-                                  nullptr, &retransmission_rate_limiter_));
+  rtp_sender_.reset(
+      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
+                    nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
+                    nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
@@ skipping 41 matching lines @@
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     uint32_t total_bitrate_;
     uint32_t retransmit_bitrate_;
   } callback;
   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
   // second averaging window.
   const uint32_t kWindowSizeMs = 1000;
   const uint32_t kPacketInterval = 20;
   const uint32_t kNumPackets =
       (kWindowSizeMs - kPacketInterval) / kPacketInterval;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
@@ skipping 37 matching lines @@
 
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
     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);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
@@ skipping 330 matching lines @@
   EXPECT_EQ(kVideoRotation_0,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 0));
   EXPECT_EQ(kVideoRotation_90,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 1));
   EXPECT_EQ(kVideoRotation_180,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 2));
   EXPECT_EQ(kVideoRotation_270,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 3));
 }
 
+TEST_F(RtpSenderTest, OnOverheadPerPacketChanged) {
+  class MockOverheadPerPacketObserver : public OverheadPerPacketObserver {
+   public:
+    MOCK_METHOD1(OnOverheadPerPacketChange, void(int overhead_per_packet));
+  };
+  MockOverheadPerPacketObserver 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, OnOverheadPerPacketChange(28)).Times(1);
+  rtp_sender_->SetTransportOverhead(28);
+
+  // RTP Overhead is without extensions is 12B.
+  // 28B + 12B = 40B
+  EXPECT_CALL(mock_overhead_observer, OnOverheadPerPacketChange(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, OnOverheadPerPacketChange(48)).Times(1);
+  SendGenericPayload();
+}
+
 }  // namespace webrtc