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

webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.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.
  */
 
@@ skipping 29 matching lines @@
 }
 
 void BeforeStreamingFixture::RestartFakeMicrophone() {
   EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
         channel_, fake_microphone_input_file_.c_str(), true, true));
 }
 
 void BeforeStreamingFixture::PausePlaying() {
   EXPECT_EQ(0, voe_base_->StopSend(channel_));
   EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
-  EXPECT_EQ(0, voe_base_->StopReceive(channel_));
 }
 
 void BeforeStreamingFixture::ResumePlaying() {
-  EXPECT_EQ(0, voe_base_->StartReceive(channel_));
   EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
   EXPECT_EQ(0, voe_base_->StartSend(channel_));
 }
 
 void BeforeStreamingFixture::WaitForTransmittedPackets(int32_t packet_count) {
   transport_->WaitForTransmittedPackets(packet_count);
 }
 
 void BeforeStreamingFixture::SetUpLocalPlayback() {
   transport_ = new LoopBackTransport(voe_network_, channel_);
   EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
 
   webrtc::CodecInst codec;
   codec.channels = 1;
   codec.pacsize = 160;
   codec.plfreq = 8000;
   codec.pltype = 0;
   codec.rate = 64000;
 #if defined(_MSC_VER) && defined(_WIN32)
   _snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE - 1, "PCMU");
 #else
   snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE, "PCMU");
 #endif
   voe_codec_->SetSendCodec(channel_, codec);
 }