Remove usage of VoENetwork from VoEWrapper and 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 103 matching lines @@
     return experimental_ns_enabled_;
   }
 
  private:
   bool experimental_ns_enabled_;
 };
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
-      public webrtc::VoEHardware,
-      public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
+      public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
-    bool external_transport = false;
     bool playout = false;
     float volume_scale = 1.0f;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
     bool opus_dtx = false;
     bool red = false;
     bool nack = false;
     int cn8_type = 13;
     int cn16_type = 105;
     int red_type = 117;
     int nack_max_packets = 0;
     uint32_t send_ssrc = 0;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
-    webrtc::PacketTime last_rtp_packet_time;
-    std::list<std::string> packets;
     int neteq_capacity = -1;
     bool neteq_fast_accelerate = false;
   };
 
   FakeWebRtcVoiceEngine() {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
@@ skipping 23 matching lines @@
   }
   int GetMaxEncodingBandwidth(int channel) {
     return channels_[channel]->max_encoding_bandwidth;
   }
   bool GetNACK(int channel) {
     return channels_[channel]->nack;
   }
   int GetNACKMaxPackets(int channel) {
     return channels_[channel]->nack_max_packets;
   }
-  const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
-    RTC_DCHECK(channels_.find(channel) != channels_.end());
-    return channels_[channel]->last_rtp_packet_time;
-  }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
         channels_[channel]->cn8_type;
   }
   int GetSendREDPayloadType(int channel) {
     return channels_[channel]->red_type;
   }
   void set_playout_fail_channel(int channel) {
     playout_fail_channel_ = channel;
@@ skipping 240 matching lines @@
   WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
   WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
   virtual bool BuiltInAECIsAvailable() const { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
   virtual bool BuiltInAGCIsAvailable() const { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   virtual bool BuiltInNSIsAvailable() const { return false; }
 
-  // webrtc::VoENetwork
-  WEBRTC_FUNC(RegisterExternalTransport, (int channel,
-                                          webrtc::Transport& transport)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = true;
-    return 0;
-  }
-  WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = false;
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (!channels_[channel]->external_transport) return -1;
-    channels_[channel]->packets.push_back(
-        std::string(static_cast<const char*>(data), length));
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length,
-                                  const webrtc::PacketTime& packet_time)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (ReceivedRTPPacket(channel, data, length) == -1) {
-      return -1;
-    }
-    channels_[channel]->last_rtp_packet_time = packet_time;
-    return 0;
-  }
-
-  WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
-                                   size_t length));
-
   // webrtc::VoERTP_RTCP
   WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->send_ssrc = ssrc;
     return 0;
   }
   WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
   WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
   WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
       unsigned char id));
@@ skipping 202 matching lines @@
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_