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

webrtc/call/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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 290 matching lines @@
                          uint32_t csrcs_count,
                          uint8_t* packet,
                          size_t packet_size,
                          Random* prng) {
   RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
   Clock* clock = Clock::GetRealTimeClock();
 
   RTPSender rtp_sender(false,     // bool audio
                        clock,     // Clock* clock
                        nullptr,   // Transport*
-                       nullptr,   // RtpAudioFeedback*
                        nullptr,   // PacedSender*
                        nullptr,   // PacketRouter*
                        nullptr,   // SendTimeObserver*
                        nullptr,   // BitrateStatisticsObserver*
                        nullptr,   // FrameCountObserver*
                        nullptr,   // SendSideDelayObserver*
                        nullptr);  // RtcEventLog*
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {
@@ skipping 363 matching lines @@
   // Enable all header extensions
   uint32_t extensions = (1u << kNumExtensions) - 1;
   uint32_t csrcs_count = 2;
   DropOldEvents(extensions, csrcs_count, 141421356);
   DropOldEvents(extensions, csrcs_count, 173205080);
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG