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/*
* 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 "api/voip/voip_engine_factory.h"
#include <memory>
#include <utility>
#include "api/make_ref_counted.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_audio_encoder_factory.h"
namespace webrtc {
namespace {
using ::testing::NiceMock;
// 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_builder =
std::make_unique<NiceMock<test::MockAudioProcessingBuilder>>();
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
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