Remove channel ids from various interfaces.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  Copyright (c) 2015 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 80 matching lines @@
    *
    * Input:
    *   id : stream id;
    *   stats : pointer to a CallStatistics to store the result.
    *
    * Returns false if the specified stream does not exist, true if succeeds.
    */
   bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
 
   // Inherit from class webrtc::Transport.
-  int SendPacket(int channel, const void *data, size_t len) override;
-  int SendRTCPPacket(int channel, const void *data, size_t len) override;
+  int SendPacket(const void *data, size_t len) override;
+  int SendRTCPPacket(const void *data, size_t len) override;
 
  private:
   struct Packet {
     enum Type { Rtp, Rtcp, } type_;
 
     Packet() : len_(0) {}
-    Packet(Type type, int channel, const void* data, size_t len, uint32 time_ms)
-        : type_(type),
-          channel_(channel),
-          len_(len),
-          send_time_ms_(time_ms) {
+    Packet(Type type, const void* data, size_t len, uint32 time_ms)
+        : type_(type), len_(len), send_time_ms_(time_ms) {
       EXPECT_LE(len_, kMaxPacketSizeByte);
       memcpy(data_, data, len_);
     }
 
-    int channel_;
     uint8_t data_[kMaxPacketSizeByte];
     size_t len_;
     uint32 send_time_ms_;
   };
 
   static bool Run(void* transport) {
     return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
   }
 
   int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
-  void StorePacket(Packet::Type type, int channel, const void* data,
-                   size_t len);
+  void StorePacket(Packet::Type type, const void* data, size_t len);
   void SendPacket(const Packet& packet);
   bool DispatchPackets();
 
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
 
   unsigned int rtt_ms_;
   unsigned int stream_count_;
@@ skipping 17 matching lines @@
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
 
   LoudestFilter loudest_filter_;
 
   const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_