Remove voe::Statistics.

audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 102 matching lines @@
 
   ConfigureStream(this, config, true);
 
   pacer_thread_checker_.DetachFromThread();
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
-  channel_proxy_->DeRegisterExternalTransport();
+  channel_proxy_->RegisterTransport(nullptr);
   channel_proxy_->ResetSenderCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
 const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return config_;
 }
 
@@ skipping 23 matching lines @@
   // using the actual packet size for the configured codec.
   if (first_time || old_config.rtp.nack.rtp_history_ms !=
                         new_config.rtp.nack.rtp_history_ms) {
     channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
                                  new_config.rtp.nack.rtp_history_ms / 20);
   }
 
   if (first_time ||
       new_config.send_transport != old_config.send_transport) {
     if (old_config.send_transport) {
-      channel_proxy->DeRegisterExternalTransport();
+      channel_proxy->RegisterTransport(nullptr);
     }
     if (new_config.send_transport) {
       stream->timed_send_transport_adapter_.reset(new TimedTransport(
           new_config.send_transport, &stream->active_lifetime_));
     } else {
       stream->timed_send_transport_adapter_.reset(nullptr);
     }
-    channel_proxy->RegisterExternalTransport(
+    channel_proxy->RegisterTransport(
         stream->timed_send_transport_adapter_.get());
   }
 
   // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
   // reserved for padding and MUST NOT be used as a local identifier.
   // So it should be safe to use 0 here to indicate "not configured".
   struct ExtensionIds {
     int audio_level = 0;
     int transport_sequence_number = 0;
   };
@@ skipping 464 matching lines @@
     if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc