Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtcp_sender.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.
  */
 
@@ skipping 114 matching lines @@
     return packet.BuildExternalBuffer(
         &context_->buffer[context_->position],
         context_->buffer_size - context_->position, this);
   }
 
  private:
   RtcpContext* const context_;
 };
 
 RTCPSender::RTCPSender(
-    int32_t id,
     bool audio,
     Clock* clock,
     ReceiveStatistics* receive_statistics,
     RtcpPacketTypeCounterObserver* packet_type_counter_observer)
-    : id_(id),
-      audio_(audio),
+    : audio_(audio),
       clock_(clock),
       method_(kRtcpOff),
       critical_section_transport_(
           CriticalSectionWrapper::CreateCriticalSection()),
       cbTransport_(nullptr),
 
       critical_section_rtcp_sender_(
           CriticalSectionWrapper::CreateCriticalSection()),
       using_nack_(false),
       sending_(false),
@@ skipping 960 matching lines @@
     delaySinceLastReceivedSR = now-receiveTime;
   }
   report_block->delaySinceLastSR = delaySinceLastReceivedSR;
   report_block->lastSR = feedback_state.remote_sr;
   return true;
 }
 
 int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
   CriticalSectionScoped lock(critical_section_transport_.get());
   if (cbTransport_) {
-    if (cbTransport_->SendRTCPPacket(id_, dataBuffer, length) > 0)
+    if (cbTransport_->SendRTCPPacket(dataBuffer, length) > 0)
       return 0;
   }
   return -1;
 }
 
 void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   csrcs_ = csrcs;
 }
@@ skipping 78 matching lines @@
   return true;
 }
 
 bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
   CriticalSectionScoped lock(critical_section_transport_.get());
   if (!cbTransport_)
     return false;
 
   class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
    public:
-    Sender(Transport* transport, int32_t id)
-        : transport_(transport), id_(id), send_failure_(false) {}
+    Sender(Transport* transport)
+        : transport_(transport), send_failure_(false) {}
 
     void OnPacketReady(uint8_t* data, size_t length) override {
-      if (transport_->SendRTCPPacket(id_, data, length) <= 0)
+      if (transport_->SendRTCPPacket(data, length) <= 0)
         send_failure_ = true;
     }
 
     Transport* const transport_;
-    int32_t id_;
     bool send_failure_;
-  } sender(cbTransport_, id_);
+  } sender(cbTransport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc