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506d4eb7e4
Bug: None Change-Id: If28819bbeebe739f07fcd8d6ea8ab841efc20f75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176562 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31453}
109 lines
3.4 KiB
C++
109 lines
3.4 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "absl/flags/declare.h"
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#include "absl/flags/flag.h"
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#include "api/test/simulated_network.h"
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#include "audio/test/audio_end_to_end_test.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/testsupport/file_utils.h"
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ABSL_DECLARE_FLAG(int, sample_rate_hz);
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ABSL_DECLARE_FLAG(bool, quick);
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namespace webrtc {
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namespace test {
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namespace {
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std::string FileSampleRateSuffix() {
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return std::to_string(absl::GetFlag(FLAGS_sample_rate_hz) / 1000);
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}
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class AudioQualityTest : public AudioEndToEndTest {
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public:
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AudioQualityTest() = default;
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private:
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std::string AudioInputFile() const {
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return test::ResourcePath(
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"voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav");
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}
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std::string AudioOutputFile() const {
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const ::testing::TestInfo* const test_info =
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::testing::UnitTest::GetInstance()->current_test_info();
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return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
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"_" + FileSampleRateSuffix() + ".wav";
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}
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std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override {
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return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
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}
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std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override {
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return TestAudioDeviceModule::CreateBoundedWavFileWriter(
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AudioOutputFile(), absl::GetFlag(FLAGS_sample_rate_hz));
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}
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void PerformTest() override {
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if (absl::GetFlag(FLAGS_quick)) {
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// Let the recording run for a small amount of time to check if it works.
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SleepMs(1000);
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} else {
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AudioEndToEndTest::PerformTest();
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}
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}
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void OnStreamsStopped() override {
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const ::testing::TestInfo* const test_info =
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::testing::UnitTest::GetInstance()->current_test_info();
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// Output information about the input and output audio files so that further
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// processing can be done by an external process.
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printf("TEST %s %s %s\n", test_info->name(), AudioInputFile().c_str(),
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AudioOutputFile().c_str());
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}
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};
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class Mobile2GNetworkTest : public AudioQualityTest {
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void ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) override {
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send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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test::CallTest::kAudioSendPayloadType,
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{"OPUS",
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48000,
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2,
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{{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
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}
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BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
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BuiltInNetworkBehaviorConfig pipe_config;
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pipe_config.link_capacity_kbps = 12;
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pipe_config.queue_length_packets = 1500;
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pipe_config.queue_delay_ms = 400;
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return pipe_config;
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}
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};
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} // namespace
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using LowBandwidthAudioTest = CallTest;
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TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
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AudioQualityTest test;
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RunBaseTest(&test);
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}
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TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
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Mobile2GNetworkTest test;
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RunBaseTest(&test);
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}
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} // namespace test
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} // namespace webrtc
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