Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/1934513002 Cr-Commit-Position: refs/heads/master@{#12566}

diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 5343800..13a3696 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h

@@ -121,15 +121,13 @@
 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;
@@ -146,8 +144,6 @@
     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;
   };
@@ -191,10 +187,6 @@
   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 :
@@ -455,40 +447,6 @@
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   bool BuiltInNSIsAvailable() const override { 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);

diff --git a/webrtc/media/engine/webrtcvoe.h b/webrtc/media/engine/webrtcvoe.h
index 2f7b997..c6d1caf 100644
--- a/webrtc/media/engine/webrtcvoe.h
+++ b/webrtc/media/engine/webrtcvoe.h

@@ -74,15 +74,13 @@
  public:
   VoEWrapper()
       : engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
-        base_(engine_), codec_(engine_),
-        hw_(engine_), network_(engine_),
-        rtp_(engine_), volume_(engine_) {
+        base_(engine_), codec_(engine_), hw_(engine_), rtp_(engine_),
+        volume_(engine_) {
   }
   VoEWrapper(webrtc::VoEAudioProcessing* processing,
              webrtc::VoEBase* base,
              webrtc::VoECodec* codec,
              webrtc::VoEHardware* hw,
-             webrtc::VoENetwork* network,
              webrtc::VoERTP_RTCP* rtp,
              webrtc::VoEVolumeControl* volume)
       : engine_(NULL),
@@ -90,7 +88,6 @@
         base_(base),
         codec_(codec),
         hw_(hw),
-        network_(network),
         rtp_(rtp),
         volume_(volume) {
   }
@@ -100,7 +97,6 @@
   webrtc::VoEBase* base() const { return base_.get(); }
   webrtc::VoECodec* codec() const { return codec_.get(); }
   webrtc::VoEHardware* hw() const { return hw_.get(); }
-  webrtc::VoENetwork* network() const { return network_.get(); }
   webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
   webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
   int error() { return base_->LastError(); }
@@ -111,7 +107,6 @@
   scoped_voe_ptr<webrtc::VoEBase> base_;
   scoped_voe_ptr<webrtc::VoECodec> codec_;
   scoped_voe_ptr<webrtc::VoEHardware> hw_;
-  scoped_voe_ptr<webrtc::VoENetwork> network_;
   scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
   scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
 };

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 64f53b7..ed67ce8 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc

@@ -2268,7 +2268,6 @@
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                        packet->cdata(), packet->size(),
                                        webrtc_packet_time);
-
   if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
     return;
   }

diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index a0d7417..c72775a 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc

@@ -58,7 +58,6 @@
                             engine,  // base
                             engine,  // codec
                             engine,  // hw
-                            engine,  // network
                             engine,  // rtp
                             engine) {  // volume
   }