Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_sender_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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class RTPSenderAudio: public DTMFqueue
 {
 public:
-    RTPSenderAudio(const int32_t id,
-                   Clock* clock,
-                   RTPSender* rtpSender,
-                   RtpAudioFeedback* audio_feedback);
+ RTPSenderAudio(Clock* clock,
+                RTPSender* rtpSender,
+                RtpAudioFeedback* audio_feedback);
     virtual ~RTPSenderAudio();
 
     int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payloadType,
                                  const uint32_t frequency,
                                  const uint8_t channels,
                                  const uint32_t rate,
                                  RtpUtility::Payload*& payload);
 
     int32_t SendAudio(const FrameType frameType,
@@ skipping 27 matching lines @@
     int32_t SendTelephoneEventPacket(bool ended,
                                      int8_t dtmf_payload_type,
                                      uint32_t dtmfTimeStamp,
                                      uint16_t duration,
                                      bool markerBit); // set on first packet in talk burst
 
     bool MarkerBit(const FrameType frameType,
                    const int8_t payloadType);
 
 private:
- const int32_t _id;
  Clock* const _clock;
  RTPSender* const _rtpSender;
  RtpAudioFeedback* const _audioFeedback;
 
  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
 
  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
  // DTMF
  bool _dtmfEventIsOn;
@@ skipping 16 matching lines @@
  int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
  int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
 
  // Audio level indication
  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_