Add POLQA to low bandwidth audio test

webrtc/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 "gflags/gflags.h"
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace {
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
 // Large bitrate by default.
 const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000};
 
-// The best that can be done with PESQ.
-constexpr int kAudioFileBitRate = 16000;
+DEFINE_int32(bitrate, 16000, "Bitrate of the produced audio file.");

[kjellander_webrtc, 2017/04/07 18:26:12]

This should probably be called sampling_frequency (Hz), since that's what it is
(sorry I didn't spot this earlier). I thought it looked a little strange in
previous reviews but I forgot to follow up.
See
https://cs.chromium.org/chromium/src/third_party/webrtc/test/fake_audio_devic...,
which you call below at line 66.

[oprypin_webrtc, 2017/04/10 12:20:40]

Done.

 }
 
 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_tiny16", "wav");
+  return test::ResourcePath("voice_engine/audio_tiny" +
+                            std::to_string(FLAGS_bitrate / 1000), "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() + ".wav";
+  return webrtc::test::OutputPath() + "LowBandwidth_" +
+      test_info->name() + "_" + std::to_string(FLAGS_bitrate / 1000) + ".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(), kAudioFileBitRate);
+      AudioOutputFile(), FLAGS_bitrate);
 }
 
 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();
@@ skipping 67 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc