8a9f3a8f53
This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
112 lines
3.5 KiB
C++
112 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 "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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#include "test/video_test_constants.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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// Sleep for whole audio duration which is 5.4 seconds.
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SleepMs(5400);
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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(AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStreamInterface::Config>*
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receive_configs) override {
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send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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test::VideoTestConstants::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 GetSendTransportConfig() 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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