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

webrtc/audio/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 33 matching lines @@
                   rtc::TaskQueue* worker_queue,
                   RtpTransportControllerSendInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats,
                   const rtc::Optional<RtpState>& suspended_rtp_state);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
+  const webrtc::AudioSendStream::Config& GetConfig() const override;

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

I'm not going to move this in the .cc file, though, to avoid messing git-blame
more than I have to. If anyone thinks we'd better move this there, too, I'm open
to that, though.

+  RtpState GetRtpState() const override;
+  const TimeInterval& GetActiveLifetime() const override;
 
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt,
                             int64_t bwe_period_ms) override;
 
   // From PacketFeedbackObserver.
   void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
   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;
-  const TimeInterval& GetActiveLifetime() const;
-
  private:
   class TimedTransport;
 
   VoiceEngine* voice_engine() const;
 
   // These are all static to make it less likely that (the old) config_ is
   // accessed unintentionally.
   static void ConfigureStream(AudioSendStream* stream,
                               const Config& new_config,
                               bool first_time);
@@ skipping 31 matching lines @@
 
   std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
   TimeInterval active_lifetime_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_