Remove unused function implementations from FakeWebRtcVoiceEngine.

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 127 matching lines @@
   bool experimental_ns_enabled_;
 };
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
-    Channel() {
-      memset(&send_codec, 0, sizeof(send_codec));
-    }
-    bool vad = false;
-    bool codec_fec = false;
-    int max_encoding_bandwidth = 0;
-    bool opus_dtx = false;
-    int cn8_type = 13;
-    int cn16_type = 105;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
-    webrtc::CodecInst send_codec;
     size_t neteq_capacity = 0;
     bool neteq_fast_accelerate = false;
   };
 
   FakeWebRtcVoiceEngine() {
     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()); }
-  bool GetVAD(int channel) {
-    return channels_[channel]->vad;
-  }
-  bool GetOpusDtx(int channel) {
-    return channels_[channel]->opus_dtx;
-  }
-  bool GetCodecFEC(int channel) {
-    return channels_[channel]->codec_fec;
-  }
-  int GetMaxEncodingBandwidth(int channel) {
-    return channels_[channel]->max_encoding_bandwidth;
-  }
-  int GetSendCNPayloadType(int channel, bool wideband) {
-    return (wideband) ?
-        channels_[channel]->cn16_type :
-        channels_[channel]->cn8_type;
-  }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
 
-  int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
-
   int GetAssociateSendChannel(int channel) {
     return channels_[channel]->associate_send_channel;
   }
 
   WEBRTC_STUB(Release, ());
 
   // webrtc::VoEBase
   WEBRTC_STUB(RegisterVoiceEngineObserver, (
       webrtc::VoiceEngineObserver& observer));
   WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
@@ skipping 53 matching lines @@
   WEBRTC_FUNC(AssociateSendChannel, (int channel,
                                      int accociate_send_channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->associate_send_channel = accociate_send_channel;
     return 0;
   }
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
-  WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    // To match the behavior of the real implementation.
-    if (_stricmp(codec.plname, "telephone-event") == 0 ||
-        _stricmp(codec.plname, "audio/telephone-event") == 0 ||
-        _stricmp(codec.plname, "CN") == 0 ||
-        _stricmp(codec.plname, "red") == 0) {
-      return -1;
-    }
-    channels_[channel]->send_codec = codec;
-    ++num_set_send_codecs_;
-    return 0;
-  }
-  WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    codec = channels_[channel]->send_codec;
-    return 0;
-  }
+  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)) {
     WEBRTC_CHECK_CHANNEL(channel);
     Channel* ch = channels_[channel];
     // Check if something else already has this slot.
     if (codec.pltype != -1) {
       for (std::vector<webrtc::CodecInst>::iterator it =
           ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
         if (it->pltype == codec.pltype &&
             _stricmp(it->plname, codec.plname) != 0) {
           return -1;
         }
       }
     }
     // Otherwise try to find this codec and update its payload type.
     int result = -1;  // not found
     for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
          it != ch->recv_codecs.end(); ++it) {
       if (strcmp(it->plname, codec.plname) == 0 &&
           it->plfreq == codec.plfreq &&
           it->channels == codec.channels) {
         it->pltype = codec.pltype;
         result = 0;
       }
     }
     return result;
   }
-  WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type,
-                                     webrtc::PayloadFrequencies frequency)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (frequency == webrtc::kFreq8000Hz) {
-      channels_[channel]->cn8_type = type;
-    } else if (frequency == webrtc::kFreq16000Hz) {
-      channels_[channel]->cn16_type = type;
-    }
-    return 0;
-  }
+  WEBRTC_STUB(SetSendCNPayloadType, (int channel, int type,
+                                     webrtc::PayloadFrequencies frequency));
   WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     Channel* ch = channels_[channel];
     for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
          it != ch->recv_codecs.end(); ++it) {
       if (strcmp(it->plname, codec.plname) == 0 &&
           it->plfreq == codec.plfreq &&
           it->channels == codec.channels &&
           it->pltype != -1) {
         codec.pltype = it->pltype;
         return 0;
       }
     }
     return -1;  // not found
   }
-  WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
-                             bool disableDTX)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (channels_[channel]->send_codec.channels == 2) {
-      // Replicating VoE behavior; VAD cannot be enabled for stereo.
-      return -1;
-    }
-    channels_[channel]->vad = enable;
-    return 0;
-  }
+  WEBRTC_STUB(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
+                             bool disableDTX));
   WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
                              webrtc::VadModes& mode, bool& disabledDTX));
-
-  WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      // TODO(minyue): Excludes other codecs if they support inband FEC.
-      return -1;
-    }
-    channels_[channel]->codec_fec = enable;
-    return 0;
-  }
-  WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    enable = channels_[channel]->codec_fec;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      return -1;
-    }
-    if (frequency_hz <= 8000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
-    else if (frequency_hz <= 12000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
-    else if (frequency_hz <= 16000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
-    else if (frequency_hz <= 24000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
-    else
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      return -1;
-    }
-    channels_[channel]->opus_dtx = enable_dtx;
-    return 0;
-  }
+  WEBRTC_STUB(SetFECStatus, (int channel, bool enable));
+  WEBRTC_STUB(GetFECStatus, (int channel, bool& enable));
+  WEBRTC_STUB(SetOpusMaxPlaybackRate, (int channel, int frequency_hz));
+  WEBRTC_STUB(SetOpusDtx, (int channel, bool enable_dtx));
 
   // webrtc::VoEHardware
   WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
   WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
   WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
@@ skipping 130 matching lines @@
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
 
  private:
   bool inited_ = false;
   int last_channel_ = -1;
   std::map<int, Channel*> channels_;
   bool fail_create_channel_ = false;
-  int num_set_send_codecs_ = 0;  // how many times we call SetSendCodec().
   bool ec_enabled_ = false;
   bool ec_metrics_enabled_ = false;
   bool cng_enabled_ = false;
   bool ns_enabled_ = false;
   bool agc_enabled_ = false;
   bool highpass_filter_enabled_ = false;
   bool stereo_swapping_enabled_ = false;
   bool typing_detection_enabled_ = false;
   webrtc::EcModes ec_mode_ = webrtc::kEcDefault;
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_