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

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 42a04aee0932b3d06ea0508d1b40373934c02c73..f46608d59afc5e89812f7cf59b58656486be7797 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -50,8 +50,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
   void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
-
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
@@ -73,7 +73,6 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   void OnPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector) override;
 
-  const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
   RtpState GetRtpState() const;