| /* |
| * Copyright (c) 2017 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/test/simulated_network.h" |
| #include "audio/test/audio_end_to_end_test.h" |
| #include "rtc_base/flags.h" |
| #include "system_wrappers/include/sleep.h" |
| #include "test/testsupport/fileutils.h" |
| |
| DEFINE_int(sample_rate_hz, |
| 16000, |
| "Sample rate (Hz) of the produced audio files."); |
| |
| DEFINE_bool(quick, |
| false, |
| "Don't do the full audio recording. " |
| "Used to quickly check that the test runs without crashing."); |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| std::string FileSampleRateSuffix() { |
| return std::to_string(FLAG_sample_rate_hz / 1000); |
| } |
| |
| class AudioQualityTest : public AudioEndToEndTest { |
| public: |
| AudioQualityTest() = default; |
| |
| private: |
| std::string AudioInputFile() const { |
| return test::ResourcePath( |
| "voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav"); |
| } |
| |
| std::string AudioOutputFile() const { |
| const ::testing::TestInfo* const test_info = |
| ::testing::UnitTest::GetInstance()->current_test_info(); |
| return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() + |
| "_" + FileSampleRateSuffix() + ".wav"; |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override { |
| return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile()); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override { |
| return TestAudioDeviceModule::CreateBoundedWavFileWriter( |
| AudioOutputFile(), FLAG_sample_rate_hz); |
| } |
| |
| void PerformTest() override { |
| if (FLAG_quick) { |
| // Let the recording run for a small amount of time to check if it works. |
| SleepMs(1000); |
| } else { |
| AudioEndToEndTest::PerformTest(); |
| } |
| } |
| |
| void OnStreamsStopped() override { |
| const ::testing::TestInfo* const test_info = |
| ::testing::UnitTest::GetInstance()->current_test_info(); |
| |
| // Output information about the input and output audio files so that further |
| // processing can be done by an external process. |
| printf("TEST %s %s %s\n", test_info->name(), AudioInputFile().c_str(), |
| AudioOutputFile().c_str()); |
| } |
| }; |
| |
| class Mobile2GNetworkTest : public AudioQualityTest { |
| void ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) override { |
| send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( |
| test::CallTest::kAudioSendPayloadType, |
| {"OPUS", |
| 48000, |
| 2, |
| {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}}); |
| } |
| |
| DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override { |
| DefaultNetworkSimulationConfig pipe_config; |
| pipe_config.link_capacity_kbps = 12; |
| pipe_config.queue_length_packets = 1500; |
| pipe_config.queue_delay_ms = 400; |
| return pipe_config; |
| } |
| }; |
| } // namespace |
| |
| using LowBandwidthAudioTest = CallTest; |
| |
| TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) { |
| AudioQualityTest test; |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) { |
| Mobile2GNetworkTest test; |
| RunBaseTest(&test); |
| } |
| } // namespace test |
| } // namespace webrtc |