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/*
* 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 "absl/flags/declare.h"
#include "absl/flags/flag.h"
#include "api/test/simulated_network.h"
#include "audio/test/audio_end_to_end_test.h"
#include "system_wrappers/include/sleep.h"
#include "test/testsupport/file_utils.h"
ABSL_DECLARE_FLAG(int, sample_rate_hz);
ABSL_DECLARE_FLAG(bool, quick);
namespace webrtc {
namespace test {
namespace {
std::string FileSampleRateSuffix() {
return std::to_string(absl::GetFlag(FLAGS_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(), absl::GetFlag(FLAGS_sample_rate_hz));
}
void PerformTest() override {
if (absl::GetFlag(FLAGS_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<AudioReceiveStreamInterface::Config>*
receive_configs) override {
send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
test::CallTest::kAudioSendPayloadType,
{"OPUS",
48000,
2,
{{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
}
BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
BuiltInNetworkBehaviorConfig 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
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