Add overhead to transport feedback observer.

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.
  */
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/stl_util.h"
+#include "webrtc/test/field_trial.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 namespace {
 const int kTransmissionTimeOffsetExtensionId = 1;
 const int kAbsoluteSendTimeExtensionId = 14;
@@ skipping 1483 matching lines @@
 
   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();
 }
 
+TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) {
+  constexpr int kTransportOverheadBytesPerPacket = 28;
+  constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
+  test::ScopedFieldTrials override_field_trials_(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  MockOverheadObserver mock_overhead_observer;
+  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_, &mock_overhead_observer));
+  rtp_sender_->SetSequenceNumber(kSeqNum);
+  rtp_sender_->SetSSRC(kSsrc);
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+  EXPECT_CALL(mock_overhead_observer,
+              OnOverheadChanged(kTransportOverheadBytesPerPacket))
+      .Times(1);
+  rtp_sender_->SetTransportOverhead(kTransportOverheadBytesPerPacket);
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                   kRtpExtensionTransportSequenceNumber,
+                   kTransportSequenceNumberExtensionId));
+
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, kSeqNum, _, _, _));
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);

[stefan-webrtc, 2016/11/28 15:21:11]

Seems like we can skip most of these EXPECT_CALLs, right? The ones on lines
1563-1573 should be enough. Would make it a bit easier to read.

May have to change to a NiceMock

[michaelt, 2016/11/29 08:22:52]

There is still som preparation needed but i could remove quite a bit.

+  const int kProbeClusterId = 1;
+  EXPECT_CALL(feedback_observer_,
+              AddPacket(kTransportSequenceNumber,
+                        sizeof(kPayloadData) + kGenericHeaderLength +
+                            kRtpOverheadBytesPerPacket +
+                            kTransportOverheadBytesPerPacket,
+                        kProbeClusterId))
+      .Times(1);
+  EXPECT_CALL(mock_overhead_observer,
+              OnOverheadChanged(kTransportOverheadBytesPerPacket +
+                                kRtpOverheadBytesPerPacket))
+      .Times(1);
+  SendGenericPayload();
+  rtp_sender_->TimeToSendPacket(
+      kSsrc, kSeqNum, fake_clock_.TimeInMilliseconds(), false, kProbeClusterId);
+}
+
 }  // namespace webrtc