Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
 #include <assert.h> //assert
 #include <string.h> //memcpy
 
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
-RTPSenderAudio::RTPSenderAudio(const int32_t id,
-                               Clock* clock,
+RTPSenderAudio::RTPSenderAudio(Clock* clock,
                                RTPSender* rtpSender,
-                               RtpAudioFeedback* audio_feedback) :
-    _id(id),
-    _clock(clock),
-    _rtpSender(rtpSender),
-    _audioFeedback(audio_feedback),
-    _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
-    _packetSizeSamples(160),
-    _dtmfEventIsOn(false),
-    _dtmfEventFirstPacketSent(false),
-    _dtmfPayloadType(-1),
-    _dtmfTimestamp(0),
-    _dtmfKey(0),
-    _dtmfLengthSamples(0),
-    _dtmfLevel(0),
-    _dtmfTimeLastSent(0),
-    _dtmfTimestampLastSent(0),
-    _REDPayloadType(-1),
-    _inbandVADactive(false),
-    _cngNBPayloadType(-1),
-    _cngWBPayloadType(-1),
-    _cngSWBPayloadType(-1),
-    _cngFBPayloadType(-1),
-    _lastPayloadType(-1),
-    _audioLevel_dBov(0) {
-}
+                               RtpAudioFeedback* audio_feedback)
+    : _clock(clock),
+      _rtpSender(rtpSender),
+      _audioFeedback(audio_feedback),
+      _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+      _packetSizeSamples(160),
+      _dtmfEventIsOn(false),
+      _dtmfEventFirstPacketSent(false),
+      _dtmfPayloadType(-1),
+      _dtmfTimestamp(0),
+      _dtmfKey(0),
+      _dtmfLengthSamples(0),
+      _dtmfLevel(0),
+      _dtmfTimeLastSent(0),
+      _dtmfTimestampLastSent(0),
+      _REDPayloadType(-1),
+      _inbandVADactive(false),
+      _cngNBPayloadType(-1),
+      _cngWBPayloadType(-1),
+      _cngSWBPayloadType(-1),
+      _cngFBPayloadType(-1),
+      _lastPayloadType(-1),
+      _audioLevel_dBov(0) {}
 
 RTPSenderAudio::~RTPSenderAudio() {
 }
 
 int RTPSenderAudio::AudioFrequency() const {
   return kDtmfFrequencyHz;
 }
 
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
@@ skipping 136 matching lines @@
         _dtmfEventFirstPacketSent = false;
         _dtmfKey = key;
         _dtmfLengthSamples = (kDtmfFrequencyHz / 1000) * dtmfLengthMS;
         dtmfToneStarted = true;
         _dtmfEventIsOn = true;
       }
     }
   }
   if (dtmfToneStarted) {
     if (_audioFeedback)
-      _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
+      _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
   }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it
   if (_dtmfEventIsOn) {
     if (frameType == kFrameEmpty) {
       // kFrameEmpty is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
       if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
@@ skipping 263 matching lines @@
         retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
                                            kAllowRetransmission,
                                            PacedSender::kHighPriority);
         sendCount--;
 
     }while (sendCount > 0 && retVal == 0);
 
     return retVal;
 }
 }  // namespace webrtc