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

webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc

 /*
  *  Copyright (c) 2011 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 30 matching lines @@
 
   TEST_LOG("Changing DTMF payload type.\n");
 
   // Start by modifying the receiving side.
   for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) {
     EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance));
     if (!_stricmp("telephone-event", codec_instance.plname)) {
       codec_instance.pltype = 88;  // Use 88 instead of default 106.
       EXPECT_EQ(0, voe_base_->StopSend(channel_));
       EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
-      EXPECT_EQ(0, voe_base_->StopReceive(channel_));
       EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance));
-      EXPECT_EQ(0, voe_base_->StartReceive(channel_));
       EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
       EXPECT_EQ(0, voe_base_->StartSend(channel_));
       break;
     }
   }
 
   Sleep(500);
 
   // Next, we must modify the sending side as well.
   EXPECT_TRUE(
       channel_proxy_->SetSendTelephoneEventPayloadType(codec_instance.pltype));
 
   RunSixteenDtmfEvents();
 }