2dfa998be2
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d Original change's description: > Prefix flag macros with WEBRTC_. > > Macros defined in rtc_base/flags.h are intended to be used to define > flags in WebRTC's binaries (e.g. tests). > > They are currently not prefixed and this could cause problems with > downstream clients since these names are quite common. > > This CL adds the 'WEBRTC_' prefix to them. > > Generated with: > > for x in DECLARE DEFINE; do > for y in bool int float string FLAG; do > git grep -l "\b$x\_$y\b" | \ > xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g" > done > done > git cl format > > Bug: webrtc:9884 > Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591 > Reviewed-on: https://webrtc-review.googlesource.com/c/106682 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25270} TBR=kwiberg@webrtc.org Bug: webrtc:9884 Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/107161 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25277}
116 lines
3.5 KiB
C++
116 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 "api/test/simulated_network.h"
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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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WEBRTC_DEFINE_int(sample_rate_hz,
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16000,
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"Sample rate (Hz) of the produced audio files.");
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WEBRTC_DEFINE_bool(
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quick,
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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<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(), 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(), 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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