Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/rtp_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 79 matching lines @@
     BitrateStatistics statistics_;
     const BitrateAggregator& aggregator_;
   };
 
   BitrateStatisticsObserver* const callback_;
   BitrateObserver total_bitrate_observer_;
   BitrateObserver retransmit_bitrate_observer_;
   uint32_t ssrc_;
 };
 
-RTPSender::RTPSender(int32_t id,
-                     bool audio,
+RTPSender::RTPSender(bool audio,
                      Clock* clock,
                      Transport* transport,
                      RtpAudioFeedback* audio_feedback,
                      PacedSender* paced_sender,
                      PacketRouter* packet_router,
                      TransportFeedbackObserver* transport_feedback_observer,
                      BitrateStatisticsObserver* bitrate_callback,
                      FrameCountObserver* frame_count_observer,
                      SendSideDelayObserver* send_side_delay_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       bitrates_(new BitrateAggregator(bitrate_callback)),
       total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
-      id_(id),
       audio_configured_(audio),
-      audio_(audio ? new RTPSenderAudio(id, clock, this, audio_feedback)
-                   : nullptr),
+      audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       packet_over_head_(28),
@@ skipping 601 matching lines @@
   if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
                             (rtx & kRtxRetransmitted) > 0, true)) {
     return -1;
   }
   return static_cast<int32_t>(length);
 }
 
 bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
   int bytes_sent = -1;
   if (transport_) {
-    bytes_sent = transport_->SendPacket(id_, packet, size);
+    bytes_sent = transport_->SendPacket(packet, size);
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", size, "sent",
                        bytes_sent);
   // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
   if (bytes_sent <= 0) {
     LOG(LS_WARNING) << "Transport failed to send packet";
     return false;
   }
   return true;
@@ skipping 1148 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc