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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 6475e00abf204e3327cd2ce6503537951c97c6e8..2cda2d438cb855875f27788a75199400a2d15985 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -637,10 +637,8 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
 
   send_stream->Stop();
 
-  webrtc::internal::AudioSendStream* audio_send_stream =

[eladalon, 2017/07/25 13:35:59]

This is the main motivation for this CL.

-      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();
+  const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
+  suspended_audio_send_ssrcs_[ssrc] = send_stream->GetRtpState();
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
@@ -655,8 +653,8 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
     }
   }
   UpdateAggregateNetworkState();
-  sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
-  delete audio_send_stream;
+  sent_rtp_audio_timer_ms_.Extend(send_stream->GetActiveLifetime());
+  delete send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(