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"
 
+
+DEFINE_int32(sample_rate_hz, 16000,
+             "Sample rate (Hz) of the produced audio files.");
+
 namespace {
+
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
-// The best that can be done with PESQ.
-constexpr int kAudioFileBitRate = 16000;
+std::string FileSampleRateSuffix(int sample_rate_hz = FLAGS_sample_rate_hz) {

[kwiberg-webrtc, 2017/04/12 11:35:06]

You don't use the argument; remove it?

[oprypin_webrtc, 2017/04/12 11:42:32]

Oops. Thanks.

+  return std::to_string(FLAGS_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_tiny16", "wav");
+  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() + ".wav";
+  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(), kAudioFileBitRate);
+      AudioOutputFile(), FLAGS_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();
@@ skipping 67 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc