Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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_rtcp_impl.h"
 
 #include <string.h>
 
 #include <set>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 #ifdef _WIN32
 // Disable warning C4355: 'this' : used in base member initializer list.
 #pragma warning(disable : 4355)
 #endif
 
 namespace webrtc {
 
 RtpRtcp::Configuration::Configuration()
-    : id(-1),
-      audio(false),
+    : audio(false),
       receiver_only(false),
       clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
       outgoing_transport(nullptr),
       intra_frame_callback(nullptr),
       bandwidth_callback(nullptr),
       transport_feedback_callback(nullptr),
       rtt_stats(nullptr),
       rtcp_packet_type_counter_observer(nullptr),
       audio_messages(NullObjectRtpAudioFeedback()),
@@ skipping 11 matching lines @@
     // No clock implementation provided, use default clock.
     RtpRtcp::Configuration configuration_copy;
     memcpy(&configuration_copy, &configuration,
            sizeof(RtpRtcp::Configuration));
     configuration_copy.clock = Clock::GetRealTimeClock();
     return new ModuleRtpRtcpImpl(configuration_copy);
   }
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
-    : rtp_sender_(configuration.id,
-                  configuration.audio,
+    : rtp_sender_(configuration.audio,
                   configuration.clock,
                   configuration.outgoing_transport,
                   configuration.audio_messages,
                   configuration.paced_sender,
                   configuration.packet_router,
                   configuration.transport_feedback_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer),
-      rtcp_sender_(configuration.id,
-                   configuration.audio,
+      rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer),
-      rtcp_receiver_(configuration.id,
-                     configuration.clock,
+      rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
                      configuration.intra_frame_callback,
                      this),
       clock_(configuration.clock),
-      id_(configuration.id),
       audio_(configuration.audio),
       collision_detected_(false),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
       packet_overhead_(28),                     // IPV4 UDP.
       padding_index_(static_cast<size_t>(-1)),  // Start padding at first child.
       nack_method_(kNackOff),
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
@@ skipping 895 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc