| /* |
| * 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 "audio/voip/voip_core.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.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/gtest.h" |
| #include "test/mock_transport.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using ::testing::NiceMock; |
| using ::testing::Return; |
| |
| constexpr int kPcmuPayload = 0; |
| |
| class VoipCoreTest : public ::testing::Test { |
| public: |
| const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1}; |
| |
| VoipCoreTest() { audio_device_ = test::MockAudioDeviceModule::CreateNice(); } |
| |
| void SetUp() override { |
| auto encoder_factory = CreateBuiltinAudioEncoderFactory(); |
| auto decoder_factory = CreateBuiltinAudioDecoderFactory(); |
| rtc::scoped_refptr<AudioProcessing> audio_processing = |
| new rtc::RefCountedObject<test::MockAudioProcessing>(); |
| |
| voip_core_ = std::make_unique<VoipCore>(); |
| voip_core_->Init(std::move(encoder_factory), std::move(decoder_factory), |
| CreateDefaultTaskQueueFactory(), audio_device_, |
| std::move(audio_processing)); |
| } |
| |
| std::unique_ptr<VoipCore> voip_core_; |
| NiceMock<MockTransport> transport_; |
| rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_; |
| }; |
| |
| // Validate expected API calls that involves with VoipCore. Some verification is |
| // involved with checking mock audio device. |
| TEST_F(VoipCoreTest, BasicVoipCoreOperation) { |
| // Program mock as non-operational and ready to start. |
| EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(false)); |
| EXPECT_CALL(*audio_device_, InitRecording()).WillOnce(Return(0)); |
| EXPECT_CALL(*audio_device_, InitPlayout()).WillOnce(Return(0)); |
| EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0)); |
| EXPECT_CALL(*audio_device_, StartPlayout()).WillOnce(Return(0)); |
| |
| auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); |
| EXPECT_TRUE(channel); |
| |
| voip_core_->SetSendCodec(*channel, kPcmuPayload, kPcmuFormat); |
| voip_core_->SetReceiveCodecs(*channel, {{kPcmuPayload, kPcmuFormat}}); |
| |
| EXPECT_TRUE(voip_core_->StartSend(*channel)); |
| EXPECT_TRUE(voip_core_->StartPlayout(*channel)); |
| |
| // Program mock as operational that is ready to be stopped. |
| EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true)); |
| EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(true)); |
| EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0)); |
| EXPECT_CALL(*audio_device_, StopPlayout()).WillOnce(Return(0)); |
| |
| EXPECT_TRUE(voip_core_->StopSend(*channel)); |
| EXPECT_TRUE(voip_core_->StopPlayout(*channel)); |
| voip_core_->ReleaseChannel(*channel); |
| } |
| |
| TEST_F(VoipCoreTest, ExpectFailToUseReleasedChannelId) { |
| auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); |
| EXPECT_TRUE(channel); |
| |
| // Release right after creation. |
| voip_core_->ReleaseChannel(*channel); |
| |
| // Now use released channel. |
| |
| // These should be no-op. |
| voip_core_->SetSendCodec(*channel, kPcmuPayload, kPcmuFormat); |
| voip_core_->SetReceiveCodecs(*channel, {{kPcmuPayload, kPcmuFormat}}); |
| |
| EXPECT_FALSE(voip_core_->StartSend(*channel)); |
| EXPECT_FALSE(voip_core_->StartPlayout(*channel)); |
| } |
| |
| } // namespace |
| } // namespace webrtc |