Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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 10 matching lines @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 
 // Handles audio RTP packets. This class is thread-safe.
 class RTPReceiverAudio : public RTPReceiverStrategy,
                          public TelephoneEventHandler {
  public:
-  RTPReceiverAudio(const int32_t id,
-                   RtpData* data_callback,
+  RTPReceiverAudio(RtpData* data_callback,
                    RtpAudioFeedback* incoming_messages_callback);
   virtual ~RTPReceiverAudio() {}
 
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
   void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
 
   // Is forwarding of outband telephone events turned on/off?
   bool TelephoneEventForwardToDecoder() const;
 
@@ skipping 24 matching lines @@
 
   bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
 
   int32_t OnNewPayloadTypeCreated(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       int8_t payload_type,
       uint32_t frequency) override;
 
   int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
-      int32_t id,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const override;
 
   // We do not allow codecs to have multiple payload types for audio, so we
   // need to override the default behavior (which is to do nothing).
   void PossiblyRemoveExistingPayloadType(
       RtpUtility::PayloadTypeMap* payload_type_map,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       size_t payload_name_length,
@@ skipping 11 matching lines @@
 
  private:
 
   int32_t ParseAudioCodecSpecific(
       WebRtcRTPHeader* rtp_header,
       const uint8_t* payload_data,
       size_t payload_length,
       const AudioPayload& audio_specific,
       bool is_red);
 
-  int32_t id_;
-
   uint32_t last_received_frequency_;
 
   bool telephone_event_forward_to_decoder_;
   int8_t telephone_event_payload_type_;
   std::set<uint8_t> telephone_event_reported_;
 
   int8_t cng_nb_payload_type_;
   int8_t cng_wb_payload_type_;
   int8_t cng_swb_payload_type_;
   int8_t cng_fb_payload_type_;
   int8_t cng_payload_type_;
 
   // G722 is special since it use the wrong number of RTP samples in timestamp
   // VS. number of samples in the frame
   int8_t g722_payload_type_;
   bool last_received_g722_;
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
 
   RtpAudioFeedback* cb_audio_feedback_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_