/* * 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 #include "call/audio_send_stream.h" #include "test/gmock.h" namespace webrtc { namespace test { class MockAudioSendStream : public AudioSendStream { public: MOCK_METHOD(const webrtc::AudioSendStream::Config&, GetConfig, (), (const, override)); MOCK_METHOD(void, Reconfigure, (const Config& config), (override)); MOCK_METHOD(void, Start, (), (override)); MOCK_METHOD(void, Stop, (), (override)); // GMock doesn't like move-only types, such as std::unique_ptr. void SendAudioData(std::unique_ptr audio_frame) override { SendAudioDataForMock(audio_frame.get()); } MOCK_METHOD(void, SendAudioDataForMock, (webrtc::AudioFrame*)); MOCK_METHOD( bool, SendTelephoneEvent, (int payload_type, int payload_frequency, int event, int duration_ms), (override)); MOCK_METHOD(void, SetMuted, (bool muted), (override)); MOCK_METHOD(Stats, GetStats, (), (const, override)); MOCK_METHOD(Stats, GetStats, (bool has_remote_tracks), (const, override)); }; } // namespace test } // namespace webrtc #endif // CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_