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.
  */
 
 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/audio/time_interval.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
  public:
+  virtual ~AudioSendStream() = default;
+
   struct Stats {
     Stats();
     ~Stats();
 
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
     uint32_t local_ssrc = 0;
     int64_t bytes_sent = 0;
     int32_t packets_sent = 0;
     int32_t packets_lost = -1;
     float fraction_lost = -1.0f;
@@ skipping 96 matching lines @@
   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() {}
+  virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
+
+  virtual RtpState GetRtpState() const = 0;
+
+  virtual const TimeInterval& GetActiveLifetime() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_