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

webrtc/call/audio_send_stream.h

 /*
  *  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 111 matching lines @@
       bool transport_cc_enabled = false;
       rtc::Optional<int> cng_payload_type;
       // If unset, use the encoder's default target bitrate.
       rtc::Optional<int> target_bitrate_bps;
     };
 
     rtc::Optional<SendCodecSpec> send_codec_spec;
     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
   };
 
+  virtual ~AudioSendStream() = default;
+
+  virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
+
   // Reconfigure the stream according to the Configuration.
   virtual void Reconfigure(const Config& config) = 0;
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
                                   int event, int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
-
- protected:
-  virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_