webrtc/api/test/mock_audio_mixer.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

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/*
* Copyright (c) 2016 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 API_TEST_MOCK_AUDIO_MIXER_H_
#define API_TEST_MOCK_AUDIO_MIXER_H_
#include "api/audio/audio_mixer.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockAudioMixer : public AudioMixer {
public:
MOCK_METHOD1(AddSource, bool(Source* audio_source));
MOCK_METHOD1(RemoveSource, void(Source* audio_source));
MOCK_METHOD2(Mix,
void(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing));
};
} // namespace test
} // namespace webrtc
#endif // API_TEST_MOCK_AUDIO_MIXER_H_