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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 619 matching lines @@
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
+  const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
-  const uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
   suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
     RTC_DCHECK_EQ(1, num_deleted);
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (AudioReceiveStream* stream : audio_receive_streams_) {
       if (stream->config().rtp.local_ssrc == ssrc) {
         stream->AssociateSendStream(nullptr);
       }
     }
   }
   UpdateAggregateNetworkState();
   sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
-  delete audio_send_stream;
+  delete send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       &audio_receiver_controller_, transport_send_->packet_router(), config,
       config_.audio_state, event_log_);
@@ skipping 748 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc