webrtc/call/test/mock_audio_send_stream.h
Yves Gerey 665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00

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1.5 KiB
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/*
* 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_