Low-bandwidth audio testing

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 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_TEST_CALL_TEST_H_
@@ skipping 65 matching lines @@
                         Transport* send_transport);
 
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
                                              float speed,
                                              int framerate,
                                              int width,
                                              int height);
   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
-  void CreateFakeAudioDevices();
+  void CreateFakeAudioDevices(
+      std::unique_ptr<FakeAudioDevice::Capturer> capturer,
+      std::unique_ptr<FakeAudioDevice::Renderer> renderer);
 
   void CreateVideoStreams();
   void CreateAudioStreams();
   void CreateFlexfecStreams();
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
@@ skipping 56 matching lines @@
   explicit BaseTest(unsigned int timeout_ms);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
   virtual size_t GetNumVideoStreams() const;
   virtual size_t GetNumAudioStreams() const;
   virtual size_t GetNumFlexfecStreams() const;
 
+  virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
+  virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
+  virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
+                                         FakeAudioDevice* recv_audio_device);
+
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
   virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
   virtual test::PacketTransport* CreateReceiveTransport();
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
@@ skipping 13 matching lines @@
       const std::vector<AudioReceiveStream*>& receive_streams);
 
   virtual void ModifyFlexfecConfigs(
       std::vector<FlexfecReceiveStream::Config>* receive_configs);
   virtual void OnFlexfecStreamsCreated(
       const std::vector<FlexfecReceiveStream*>& receive_streams);
 
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 
+  virtual void OnTestFinished();
+
   webrtc::RtcEventLogNullImpl event_log_;
 };
 
 class SendTest : public BaseTest {
  public:
   explicit SendTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 class EndToEndTest : public BaseTest {
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_