blob: 4eb1166316601d649495197de87e7ce047096232 [file] [log] [blame]
/*
* Copyright (c) 2015 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 CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
#define CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
#include <memory>
#include "call/audio_send_stream.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockAudioSendStream : public AudioSendStream {
public:
MOCK_CONST_METHOD0(GetConfig, const webrtc::AudioSendStream::Config&());
MOCK_METHOD1(Reconfigure, void(const Config& config));
MOCK_METHOD0(Start, void());
MOCK_METHOD0(Stop, void());
// GMock doesn't like move-only types, such as std::unique_ptr.
virtual void SendAudioData(
std::unique_ptr<webrtc::AudioFrame> audio_frame) {
SendAudioDataForMock(audio_frame.get());
}
MOCK_METHOD1(SendAudioDataForMock,
void(webrtc::AudioFrame* audio_frame));
MOCK_METHOD4(SendTelephoneEvent,
bool(int payload_type, int payload_frequency, int event,
int duration_ms));
MOCK_METHOD1(SetMuted, void(bool muted));
MOCK_CONST_METHOD0(GetStats, Stats());
MOCK_CONST_METHOD1(GetStats, Stats(bool has_remote_tracks));
};
} // namespace test
} // namespace webrtc
#endif // CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_