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/*
* 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_