Remove usage of VoEBase::AssociateSendChannel() from WVoMC.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 39 matching lines @@
 
 #define WEBRTC_FUNC(method, args) int method args override
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
-    int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     size_t neteq_capacity = 0;
     bool neteq_fast_accelerate = false;
   };
 
   explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm) : apm_(apm) {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
 
-  int GetAssociateSendChannel(int channel) {
-    return channels_[channel]->associate_send_channel;
-  }
-
   WEBRTC_STUB(Release, ());
 
   // webrtc::VoEBase
   WEBRTC_STUB(RegisterVoiceEngineObserver, (
       webrtc::VoiceEngineObserver& observer));
   WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
   WEBRTC_FUNC(Init,
               (webrtc::AudioDeviceModule* adm,
                webrtc::AudioProcessing* audioproc,
                const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
@@ skipping 22 matching lines @@
     auto db = webrtc::acm2::RentACodec::Database();
     ch->recv_codecs.assign(db.begin(), db.end());
     ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
     ch->neteq_fast_accelerate =
         config.acm_config.neteq_config.enable_fast_accelerate;
     channels_[++last_channel_] = ch;
     return last_channel_;
   }
   WEBRTC_FUNC(DeleteChannel, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    for (const auto& ch : channels_) {
-      if (ch.second->associate_send_channel == channel) {
-        ch.second->associate_send_channel = -1;
-      }
-    }
     delete channels_[channel];
     channels_.erase(channel);
     return 0;
   }
   WEBRTC_STUB(StartReceive, (int channel));
   WEBRTC_STUB(StartPlayout, (int channel));
   WEBRTC_STUB(StartSend, (int channel));
   WEBRTC_STUB(StopReceive, (int channel));
   WEBRTC_STUB(StopPlayout, (int channel));
   WEBRTC_STUB(StopSend, (int channel));
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
-  WEBRTC_FUNC(AssociateSendChannel, (int channel,
-                                     int accociate_send_channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->associate_send_channel = accociate_send_channel;
-    return 0;
-  }
+  WEBRTC_STUB(AssociateSendChannel, (int channel,
+                                     int accociate_send_channel));
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
   WEBRTC_STUB(SetSendCodec, (int channel, const webrtc::CodecInst& codec));
   WEBRTC_STUB(GetSendCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
   WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetRecPayloadType, (int channel,
                                   const webrtc::CodecInst& codec)) {
@@ skipping 206 matching lines @@
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   webrtc::AudioProcessing* apm_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_