/*
 *  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 SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_
#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_

#include <string>
#include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"

// This fixture will, in addition to the work done by its superclasses,
// create a channel and prepare playing a file through the fake microphone
// to simulate microphone input. The purpose is to make it convenient
// to write tests that require microphone input.
class BeforeStreamingFixture : public AfterInitializationFixture {
 public:
  BeforeStreamingFixture();
  virtual ~BeforeStreamingFixture();

 protected:
  int             channel_;
  std::string     fake_microphone_input_file_;

  // Shuts off the fake microphone for this test.
  void SwitchToManualMicrophone();

  // Restarts the fake microphone if it's been shut off earlier.
  void RestartFakeMicrophone();

  // Stops all sending and playout.
  void PausePlaying();

  // Resumes all sending and playout.
  void ResumePlaying();

  // Waits until packet_count packetes have been processed by recipient.
  void WaitForTransmittedPackets(int32_t packet_count);

 private:
  void SetUpLocalPlayback();

  LoopBackTransport* transport_;
};


#endif  // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_