Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.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 224 matching lines @@
 class RtpFeedback
 {
 public:
     virtual ~RtpFeedback() {}
 
     // Receiving payload change or SSRC change. (return success!)
     /*
     *   channels    - number of channels in codec (1 = mono, 2 = stereo)
     */
     virtual int32_t OnInitializeDecoder(
-        const int32_t id,
         const int8_t payloadType,
         const char payloadName[RTP_PAYLOAD_NAME_SIZE],
         const int frequency,
         const uint8_t channels,
         const uint32_t rate) = 0;
 
-    virtual void OnIncomingSSRCChanged( const int32_t id,
-                                        const uint32_t ssrc) = 0;
+    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
 
-    virtual void OnIncomingCSRCChanged( const int32_t id,
-                                        const uint32_t CSRC,
-                                        const bool added) = 0;
+    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
+                                       const bool added) = 0;
 };
 
 class RtpAudioFeedback {
  public:
-
-  virtual void OnPlayTelephoneEvent(const int32_t id,
-                                    const uint8_t event,
+  virtual void OnPlayTelephoneEvent(const uint8_t event,
                                     const uint16_t lengthMs,
                                     const uint8_t volume) = 0;
+
  protected:
   virtual ~RtpAudioFeedback() {}
 };
 
 class RtcpIntraFrameObserver {
  public:
   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
 
   virtual void OnReceivedSLI(uint32_t ssrc,
                              uint8_t picture_id) = 0;
@@ skipping 64 matching lines @@
   virtual int64_t LastProcessedRtt() const = 0;
 
   virtual ~RtcpRttStats() {};
 };
 
 // Null object version of RtpFeedback.
 class NullRtpFeedback : public RtpFeedback {
  public:
   virtual ~NullRtpFeedback() {}
 
-  int32_t OnInitializeDecoder(const int32_t id,
-                              const int8_t payloadType,
+  int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const uint8_t channels,
                               const uint32_t rate) override {
     return 0;
   }
 
-  void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {}
-
-  void OnIncomingCSRCChanged(const int32_t id,
-                             const uint32_t CSRC,
-                             const bool added) override {}
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
 };
 
 // Null object version of RtpData.
 class NullRtpData : public RtpData {
  public:
   virtual ~NullRtpData() {}
 
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
                                 const size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override {
     return 0;
   }
 
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
     return true;
   }
 };
 
 // Null object version of RtpAudioFeedback.
 class NullRtpAudioFeedback : public RtpAudioFeedback {
  public:
   virtual ~NullRtpAudioFeedback() {}
 
-  void OnPlayTelephoneEvent(const int32_t id,
-                            const uint8_t event,
+  void OnPlayTelephoneEvent(const uint8_t event,
                             const uint16_t lengthMs,
                             const uint8_t volume) override {}
 };
 
 // Statistics about packet loss for a single directional connection. All values
 // are totals since the connection initiated.
 struct RtpPacketLossStats {
   // The number of packets lost in events where no adjacent packets were also
   // lost.
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
   uint64_t multiple_packet_loss_packet_count;
 };
 
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_