Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.

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 123 matching lines @@
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
   }
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
-    rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+    rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
                                     pacer ? &mock_paced_sender_ : nullptr,
                                     &seq_num_allocator_, nullptr, nullptr,
                                     nullptr, nullptr, &mock_rtc_event_log_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtcEventLog mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   MockTransportSequenceNumberAllocator seq_num_allocator_;
@@ skipping 792 matching lines @@
   // Verify transmission time offset. This packet is sent without delay.
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, nullptr, &mock_paced_sender_, nullptr,
+      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
       nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_));
   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 121 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_, nullptr,
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
                                   &mock_paced_sender_, nullptr, nullptr,
                                   nullptr, &callback, nullptr, 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();
@@ skipping 35 matching lines @@
       ssrc_ = ssrc;
       total_stats_ = total_stats;
       retransmit_stats_ = retransmit_stats;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     BitrateStatistics total_stats_;
     BitrateStatistics retransmit_stats_;
   } callback;
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
                                   nullptr, nullptr, nullptr, &callback, nullptr,
                                   nullptr, nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
   const uint32_t kPacketInterval = 20;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
@@ skipping 32 matching lines @@
 
   rtp_sender_.reset();
 }
 
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
-    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
+    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
@@ skipping 307 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc