diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc | 51 |
1 files changed, 51 insertions, 0 deletions
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 |