Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

webrtc/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 108 matching lines @@
 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_->ResetSenderCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
+const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  return config_;
+}
+
 void AudioSendStream::Reconfigure(
     const webrtc::AudioSendStream::Config& new_config) {
   ConfigureStream(this, new_config, false);
 }
 
 void AudioSendStream::ConfigureStream(
     webrtc::internal::AudioSendStream* stream,
     const webrtc::AudioSendStream::Config& new_config,
     bool first_time) {
   LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString();
@@ skipping 259 matching lines @@
   // the previously sent value is no longer relevant. This will be taken care
   // of with some refactoring which is now being done.
   if (plr) {
     channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
   }
   if (rplr) {
     channel_proxy_->OnRecoverableUplinkPacketLossRate(*rplr);
   }
 }
 
-const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return config_;
-}
-
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   transport_->send_side_cc()->SetTransportOverhead(
       transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
 
 RtpState AudioSendStream::GetRtpState() const {
   return rtp_rtcp_module_->GetRtpState();
 }
@@ skipping 221 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