Adds more unit tests for WebRTC.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <vector>
 
@@ skipping 73 matching lines @@
 //  (*) Using SyncSocket for inter-process synchronization with low latency.
 //      The actual data is transferred via SharedMemory. IPC is not involved
 //      in the actual media transfer.
 //
 // Implementation notes:
 //
 //  - This class must be created on the main render thread.
 //  - The webrtc::AudioDeviceModule is reference counted.
 //  - Recording is currently not supported on Mac OS X.
 //
-class WebRtcAudioDeviceImpl
+class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : public webrtc::AudioDeviceModule,
       public AudioDevice::RenderCallback,
       public AudioInputDevice::CaptureCallback,
       public AudioInputDevice::CaptureEventHandler {
  public:
   // Methods called on main render thread.
-  CONTENT_EXPORT WebRtcAudioDeviceImpl();
+  WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
   // We need this one to support runnable method tasks.
   static bool ImplementsThreadSafeReferenceCounting() { return true; }
 
@@ skipping 131 matching lines @@
       const uint32_t samples_per_sec) OVERRIDE;
   virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
   virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
   virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
   virtual int32_t ResetAudioDevice() OVERRIDE;
   virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
   virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
   // Sets the session id.
-  CONTENT_EXPORT void SetSessionId(int session_id);
+  void SetSessionId(int session_id);
 
   // Accessors.
   size_t input_buffer_size() const { return input_buffer_size_; }
   size_t output_buffer_size() const { return output_buffer_size_; }
   int input_channels() const { return input_channels_; }
   int output_channels() const { return output_channels_; }
+  int input_sample_rate() const { return static_cast<int>(input_sample_rate_); }
+  int output_sample_rate() const {
+    return static_cast<int>(output_sample_rate_);
+  }
+  int input_delay_ms() const { return input_delay_ms_; }
+  int output_delay_ms() const { return output_delay_ms_; }
+  bool initialized() const { return initialized_; }
+  bool playing() const { return playing_; }
+  bool recording() const { return recording_; }
 
  private:
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
 
   // Methods called on the main render thread ----------------------------------
   // The following methods are tasks posted on the render thread that needs to
   // be executed on that thread.
   void InitOnRenderThread(int32_t* error, base::WaitableEvent* event);
 
@@ skipping 34 matching lines @@
   scoped_array<int16> output_buffer_;
 
   webrtc::AudioDeviceModule::ErrorCode last_error_;
 
   base::TimeTicks last_process_time_;
 
   // Id of the media session to be started, it tells which device to be used
   // on the input/capture side.
   int session_id_;
 
-  // Protect |recording_|.
+  // Protects |recording_|.
   base::Lock lock_;
 
   int bytes_per_sample_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_