Using Rent-A-Codec for static Codec access in WVoE/MC.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 39 matching lines @@
 
 class AudioDeviceModule;
 class AudioRenderer;
 class VoEWrapper;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
 class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
+ public:
+  // Exposed for the WVoE/MC unit test.
+  static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
 
- public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
   explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   bool SetDevices(const Device* in_device, const Device* out_device);
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
 
   const std::vector<AudioCodec>& codecs();
-  bool FindCodec(const AudioCodec& codec);
-  bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
-
   const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel* channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
 
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
@@ skipping 13 matching lines @@
 
   // Starts recording an RtcEventLog using an existing file until 10 minutes
   // pass or the StopRtcEventLog function is called.
   bool StartRtcEventLog(rtc::PlatformFile file);
 
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
 
  private:
   void Construct();
-  void ConstructCodecs();
-  bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   // Given the device type, name, and id, find device id. Return true and
@@ skipping 122 matching lines @@
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
-  bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
@@ skipping 20 matching lines @@
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_