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52e46247bf
Bug: webrtc:42226242 Change-Id: I54f58eca55bed5db08020129514fb187b9a05d58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359882 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42806}
126 lines
3.8 KiB
C++
126 lines
3.8 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
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#define API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
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#include <cstdint>
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#include <map>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/voip/voip_base.h"
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#include "api/voip/voip_codec.h"
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#include "api/voip/voip_dtmf.h"
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#include "api/voip/voip_engine.h"
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#include "api/voip/voip_network.h"
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#include "api/voip/voip_statistics.h"
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#include "api/voip/voip_volume_control.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockVoipBase : public VoipBase {
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public:
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MOCK_METHOD(ChannelId,
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CreateChannel,
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(Transport*, absl::optional<uint32_t>),
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(override));
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MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
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MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
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MOCK_METHOD(VoipResult, StopSend, (ChannelId), (override));
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MOCK_METHOD(VoipResult, StartPlayout, (ChannelId), (override));
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MOCK_METHOD(VoipResult, StopPlayout, (ChannelId), (override));
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};
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class MockVoipCodec : public VoipCodec {
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public:
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MOCK_METHOD(VoipResult,
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SetSendCodec,
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(ChannelId, int, const SdpAudioFormat&),
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(override));
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MOCK_METHOD(VoipResult,
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SetReceiveCodecs,
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(ChannelId, (const std::map<int, SdpAudioFormat>&)),
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(override));
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};
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class MockVoipDtmf : public VoipDtmf {
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public:
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MOCK_METHOD(VoipResult,
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RegisterTelephoneEventType,
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(ChannelId, int, int),
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(override));
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MOCK_METHOD(VoipResult,
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SendDtmfEvent,
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(ChannelId, DtmfEvent, int),
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(override));
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};
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class MockVoipNetwork : public VoipNetwork {
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public:
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MOCK_METHOD(VoipResult,
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ReceivedRTPPacket,
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(ChannelId channel_id, rtc::ArrayView<const uint8_t> rtp_packet),
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(override));
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MOCK_METHOD(VoipResult,
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ReceivedRTCPPacket,
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(ChannelId channel_id, rtc::ArrayView<const uint8_t> rtcp_packet),
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(override));
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};
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class MockVoipStatistics : public VoipStatistics {
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public:
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MOCK_METHOD(VoipResult,
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GetIngressStatistics,
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(ChannelId, IngressStatistics&),
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(override));
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MOCK_METHOD(VoipResult,
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GetChannelStatistics,
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(ChannelId channel_id, ChannelStatistics&),
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(override));
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};
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class MockVoipVolumeControl : public VoipVolumeControl {
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public:
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MOCK_METHOD(VoipResult, SetInputMuted, (ChannelId, bool), (override));
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MOCK_METHOD(VoipResult,
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GetInputVolumeInfo,
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(ChannelId, VolumeInfo&),
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(override));
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MOCK_METHOD(VoipResult,
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GetOutputVolumeInfo,
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(ChannelId, VolumeInfo&),
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(override));
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};
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class MockVoipEngine : public VoipEngine {
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public:
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VoipBase& Base() override { return base_; }
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VoipNetwork& Network() override { return network_; }
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VoipCodec& Codec() override { return codec_; }
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VoipDtmf& Dtmf() override { return dtmf_; }
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VoipStatistics& Statistics() override { return statistics_; }
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VoipVolumeControl& VolumeControl() override { return volume_; }
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// Direct access to underlying members are required for testing.
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MockVoipBase base_;
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MockVoipNetwork network_;
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MockVoipCodec codec_;
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MockVoipDtmf dtmf_;
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MockVoipStatistics statistics_;
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MockVoipVolumeControl volume_;
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};
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} // namespace webrtc
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#endif // API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
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