Remove voe::Channel::StopReceive() and associated logic.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/base/checks.h"
@@ skipping 110 matching lines @@
   if (audio_send_stream_) {
     fake_send_audio_device_->Start();
     audio_send_stream_->Start();
     EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
   }
   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
     audio_recv_stream->Start();
   if (!audio_receive_streams_.empty()) {
     fake_recv_audio_device_->Start();
     EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
-    EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
   }
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Start();
 }
 
 void CallTest::Stop() {
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Stop();
   if (!audio_receive_streams_.empty()) {
     fake_recv_audio_device_->Stop();
-    EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
     EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
   }
   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
     audio_recv_stream->Stop();
   if (audio_send_stream_) {
     fake_send_audio_device_->Stop();
     EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
     audio_send_stream_->Stop();
   }
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
@@ skipping 284 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc