diff options
Diffstat (limited to 'third_party/libwebrtc/api/voip/test')
3 files changed, 189 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/test/compile_all_headers.cc b/third_party/libwebrtc/api/voip/test/compile_all_headers.cc new file mode 100644 index 0000000000..73a0f0d1c4 --- /dev/null +++ b/third_party/libwebrtc/api/voip/test/compile_all_headers.cc @@ -0,0 +1,14 @@ +/* + * 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. + */ + +// This file verifies that all include files in this directory can be +// compiled without errors or other required includes. + +#include "api/voip/test/mock_voip_engine.h" diff --git a/third_party/libwebrtc/api/voip/test/mock_voip_engine.h b/third_party/libwebrtc/api/voip/test/mock_voip_engine.h new file mode 100644 index 0000000000..74b880d652 --- /dev/null +++ b/third_party/libwebrtc/api/voip/test/mock_voip_engine.h @@ -0,0 +1,124 @@ +/* + * 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_ diff --git a/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc b/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc new file mode 100644 index 0000000000..f967a0ba8f --- /dev/null +++ b/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2020 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. + */ + +#include <utility> + +#include "api/task_queue/default_task_queue_factory.h" +#include "api/voip/voip_engine_factory.h" +#include "modules/audio_device/include/mock_audio_device.h" +#include "modules/audio_processing/include/mock_audio_processing.h" +#include "test/gtest.h" +#include "test/mock_audio_decoder_factory.h" +#include "test/mock_audio_encoder_factory.h" + +namespace webrtc { +namespace { + +// Create voip engine with mock modules as normal use case. +TEST(VoipEngineFactoryTest, CreateEngineWithMockModules) { + VoipEngineConfig config; + config.encoder_factory = rtc::make_ref_counted<MockAudioEncoderFactory>(); + config.decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>(); + config.task_queue_factory = CreateDefaultTaskQueueFactory(); + config.audio_processing = + rtc::make_ref_counted<testing::NiceMock<test::MockAudioProcessing>>(); + config.audio_device_module = test::MockAudioDeviceModule::CreateNice(); + + auto voip_engine = CreateVoipEngine(std::move(config)); + EXPECT_NE(voip_engine, nullptr); +} + +// Create voip engine without setting audio processing as optional component. +TEST(VoipEngineFactoryTest, UseNoAudioProcessing) { + VoipEngineConfig config; + config.encoder_factory = rtc::make_ref_counted<MockAudioEncoderFactory>(); + config.decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>(); + config.task_queue_factory = CreateDefaultTaskQueueFactory(); + config.audio_device_module = test::MockAudioDeviceModule::CreateNice(); + + auto voip_engine = CreateVoipEngine(std::move(config)); + EXPECT_NE(voip_engine, nullptr); +} + +} // namespace +} // namespace webrtc |