92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
114 lines
3.5 KiB
C++
114 lines
3.5 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 "audio/test/audio_end_to_end_test.h"
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#include "rtc_base/flags.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/testsupport/fileutils.h"
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DEFINE_int(sample_rate_hz, 16000,
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"Sample rate (Hz) of the produced audio files.");
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DEFINE_bool(quick, false,
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"Don't do the full audio recording. "
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"Used to quickly check that the test runs without crashing.");
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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(FLAG_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<test::FakeAudioDevice::Capturer> CreateCapturer() override {
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return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
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}
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std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override {
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return test::FakeAudioDevice::CreateBoundedWavFileWriter(
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AudioOutputFile(), FLAG_sample_rate_hz);
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}
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void PerformTest() override {
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if (FLAG_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(),
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AudioInputFile().c_str(), 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(AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) override {
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send_config->send_codec_spec =
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rtc::Optional<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"},
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{"ptime", "60"},
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{"stereo", "1"}}}});
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}
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FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
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FakeNetworkPipe::Config 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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