Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 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 68 matching lines @@
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    VideoRotation rotation) const = 0;
   virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0;
   virtual CVOMode ActivateCVORtpHeaderExtension() = 0;
 };
 
 class RTPSender : public RTPSenderInterface {
  public:
-  RTPSender(int32_t id,
-            bool audio,
+  RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
             PacedSender* paced_sender,
             PacketRouter* packet_router,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer);
   virtual ~RTPSender();
@@ skipping 280 matching lines @@
                       bool is_rtx,
                       bool is_retransmit);
   bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
 
   Clock* clock_;
   int64_t clock_delta_ms_;
 
   rtc::scoped_ptr<BitrateAggregator> bitrates_;
   Bitrate total_bitrate_sent_;
 
-  int32_t id_;
-
   const bool audio_configured_;
   rtc::scoped_ptr<RTPSenderAudio> audio_;
   rtc::scoped_ptr<RTPSenderVideo> video_;
 
   PacedSender* const paced_sender_;
   PacketRouter* const packet_router_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_;
 
@@ skipping 58 matching lines @@
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_