You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
110 lines
3.4 KiB
110 lines
3.4 KiB
/*
|
|
* 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<AudioReceiveStream::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 GetNetworkPipeConfig() 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
|