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/*
* Copyright (c) 2021 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_VOIP_TEST_MOCK_VOIP_ENGINE_H_
#define API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
#include <map>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/voip/voip_base.h"
#include "api/voip/voip_codec.h"
#include "api/voip/voip_dtmf.h"
#include "api/voip/voip_engine.h"
#include "api/voip/voip_network.h"
#include "api/voip/voip_statistics.h"
#include "api/voip/voip_volume_control.h"
#include "test/gmock.h"
namespace webrtc {
class MockVoipBase : public VoipBase {
public:
MOCK_METHOD(ChannelId,
CreateChannel,
(Transport*, absl::optional<uint32_t>),
(override));
MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
MOCK_METHOD(VoipResult, StopSend, (ChannelId), (override));
MOCK_METHOD(VoipResult, StartPlayout, (ChannelId), (override));
MOCK_METHOD(VoipResult, StopPlayout, (ChannelId), (override));
};
class MockVoipCodec : public VoipCodec {
public:
MOCK_METHOD(VoipResult,
SetSendCodec,
(ChannelId, int, const SdpAudioFormat&),
(override));
MOCK_METHOD(VoipResult,
SetReceiveCodecs,
(ChannelId, (const std::map<int, SdpAudioFormat>&)),
(override));
};
class MockVoipDtmf : public VoipDtmf {
public:
MOCK_METHOD(VoipResult,
RegisterTelephoneEventType,
(ChannelId, int, int),
(override));
MOCK_METHOD(VoipResult,
SendDtmfEvent,
(ChannelId, DtmfEvent, int),
(override));
};
class MockVoipNetwork : public VoipNetwork {
public:
MOCK_METHOD(VoipResult,
ReceivedRTPPacket,
(ChannelId channel_id, rtc::ArrayView<const uint8_t> rtp_packet),
(override));
MOCK_METHOD(VoipResult,
ReceivedRTCPPacket,
(ChannelId channel_id, rtc::ArrayView<const uint8_t> rtcp_packet),
(override));
};
class MockVoipStatistics : public VoipStatistics {
public:
MOCK_METHOD(VoipResult,
GetIngressStatistics,
(ChannelId, IngressStatistics&),
(override));
MOCK_METHOD(VoipResult,
GetChannelStatistics,
(ChannelId channel_id, ChannelStatistics&),
(override));
};
class MockVoipVolumeControl : public VoipVolumeControl {
public:
MOCK_METHOD(VoipResult, SetInputMuted, (ChannelId, bool), (override));
MOCK_METHOD(VoipResult,
GetInputVolumeInfo,
(ChannelId, VolumeInfo&),
(override));
MOCK_METHOD(VoipResult,
GetOutputVolumeInfo,
(ChannelId, VolumeInfo&),
(override));
};
class MockVoipEngine : public VoipEngine {
public:
VoipBase& Base() override { return base_; }
VoipNetwork& Network() override { return network_; }
VoipCodec& Codec() override { return codec_; }
VoipDtmf& Dtmf() override { return dtmf_; }
VoipStatistics& Statistics() override { return statistics_; }
VoipVolumeControl& VolumeControl() override { return volume_; }
// Direct access to underlying members are required for testing.
MockVoipBase base_;
MockVoipNetwork network_;
MockVoipCodec codec_;
MockVoipDtmf dtmf_;
MockVoipStatistics statistics_;
MockVoipVolumeControl volume_;
};
} // namespace webrtc
#endif // API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
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