audio/test/low_bandwidth_audio_test.cc

/*
 *  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 <algorithm>

#include "webrtc/audio/test/low_bandwidth_audio_test.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"


DEFINE_int(sample_rate_hz, 16000,
           "Sample rate (Hz) of the produced audio files.");

DEFINE_bool(quick, false,
            "Don't do the full audio recording. "
            "Used to quickly check that the test runs without crashing.");

namespace {

// Wait half a second between stopping sending and stopping receiving audio.
constexpr int kExtraRecordTimeMs = 500;

std::string FileSampleRateSuffix() {
  return std::to_string(FLAG_sample_rate_hz / 1000);
}

}  // namespace

namespace webrtc {
namespace test {

AudioQualityTest::AudioQualityTest()
    : EndToEndTest(CallTest::kDefaultTimeoutMs) {}

size_t AudioQualityTest::GetNumVideoStreams() const {
  return 0;
}
size_t AudioQualityTest::GetNumAudioStreams() const {
  return 1;
}
size_t AudioQualityTest::GetNumFlexfecStreams() const {
  return 0;
}

std::string AudioQualityTest::AudioInputFile() {
  return test::ResourcePath("voice_engine/audio_tiny" + FileSampleRateSuffix(),
                            "wav");
}

std::string AudioQualityTest::AudioOutputFile() {
  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<test::FakeAudioDevice::Capturer>
    AudioQualityTest::CreateCapturer() {
  return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
}

std::unique_ptr<test::FakeAudioDevice::Renderer>
    AudioQualityTest::CreateRenderer() {
  return test::FakeAudioDevice::CreateBoundedWavFileWriter(
      AudioOutputFile(), FLAG_sample_rate_hz);
}

void AudioQualityTest::OnFakeAudioDevicesCreated(
    test::FakeAudioDevice* send_audio_device,
    test::FakeAudioDevice* recv_audio_device) {
  send_audio_device_ = send_audio_device;
}

FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
  return FakeNetworkPipe::Config();
}

test::PacketTransport* AudioQualityTest::CreateSendTransport(
    SingleThreadedTaskQueueForTesting* task_queue,
    Call* sender_call) {
  return new test::PacketTransport(
      task_queue, sender_call, this, test::PacketTransport::kSender,
      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}

test::PacketTransport* AudioQualityTest::CreateReceiveTransport(
    SingleThreadedTaskQueueForTesting* task_queue) {
  return new test::PacketTransport(
      task_queue, nullptr, this, test::PacketTransport::kReceiver,
      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}

void AudioQualityTest::ModifyAudioConfigs(
  AudioSendStream::Config* send_config,
  std::vector<AudioReceiveStream::Config>* receive_configs) {
  // Large bitrate by default.
  const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
                                              {{"stereo", "1"}});
  send_config->send_codec_spec =
      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
          {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
}

void AudioQualityTest::PerformTest() {
  if (FLAG_quick) {
    // Let the recording run for a small amount of time to check if it works.
    SleepMs(1000);
  } else {
    // Wait until the input audio file is done...
    send_audio_device_->WaitForRecordingEnd();
    // and some extra time to account for network delay.
    SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
  }
}

void AudioQualityTest::OnTestFinished() {
  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());
}


using LowBandwidthAudioTest = CallTest;

TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
  AudioQualityTest test;
  RunBaseTest(&test);
}


class Mobile2GNetworkTest : public AudioQualityTest {
  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
      std::vector<AudioReceiveStream::Config>* receive_configs) override {
    send_config->send_codec_spec =
        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
            {test::CallTest::kAudioSendPayloadType,
             {"OPUS",
              48000,
              2,
              {{"maxaveragebitrate", "6000"},
               {"ptime", "60"},
               {"stereo", "1"}}}});
  }

  FakeNetworkPipe::Config GetNetworkPipeConfig() override {
    FakeNetworkPipe::Config pipe_config;
    pipe_config.link_capacity_kbps = 12;
    pipe_config.queue_length_packets = 1500;
    pipe_config.queue_delay_ms = 400;
    return pipe_config;
  }
};

TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
  Mobile2GNetworkTest test;
  RunBaseTest(&test);
}

}  // namespace test
}  // namespace webrtc