Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_receiver_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_receiver_audio.h"
 
 #include <assert.h>  // assert
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
-    int32_t id, RtpData* data_callback,
+    RtpData* data_callback,
     RtpAudioFeedback* incoming_messages_callback) {
-  return new RTPReceiverAudio(id, data_callback, incoming_messages_callback);
+  return new RTPReceiverAudio(data_callback, incoming_messages_callback);
 }
 
-RTPReceiverAudio::RTPReceiverAudio(const int32_t id,
-                                   RtpData* data_callback,
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
                                    RtpAudioFeedback* incoming_messages_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
-      id_(id),
       last_received_frequency_(8000),
       telephone_event_forward_to_decoder_(false),
       telephone_event_payload_type_(-1),
       cng_nb_payload_type_(-1),
       cng_wb_payload_type_(-1),
       cng_swb_payload_type_(-1),
       cng_fb_payload_type_(-1),
       cng_payload_type_(-1),
       g722_payload_type_(-1),
       last_received_g722_(false),
@@ skipping 212 matching lines @@
 
   if (num_energy_ > 0) {
     memcpy(array_of_energy, current_remote_energy_,
            sizeof(uint8_t) * num_energy_);
   }
   return num_energy_;
 }
 
 int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
     RtpFeedback* callback,
-    int32_t id,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
-  if (-1 == callback->OnInitializeDecoder(id,
-                                          payload_type,
-                                          payload_name,
-                                          specific_payload.Audio.frequency,
-                                          specific_payload.Audio.channels,
-                                          specific_payload.Audio.rate)) {
+  if (-1 ==
+      callback->OnInitializeDecoder(
+          payload_type, payload_name, specific_payload.Audio.frequency,
+          specific_payload.Audio.channels, specific_payload.Audio.rate)) {
     LOG(LS_ERROR) << "Failed to create decoder for payload type: "
                   << payload_name << "/" << static_cast<int>(payload_type);
     return -1;
   }
   return 0;
 }
 
 // We are not allowed to have any critsects when calling data_callback.
 int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
     WebRtcRTPHeader* rtp_header,
@@ skipping 95 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc