Low-bandwidth audio testing

webrtc/audio/test/low_bandwidth_audio_test.cc

Index: webrtc/audio/test/low_bandwidth_audio_test.cc
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc
index c78e8897a824008e6336d676b7a3e5040c1ff5b5..c1b1a02235610b51cfa708aa934eb6183ccc3275 100644
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *  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
@@ -8,9 +8,144 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This is a placeholder for the work oprypin@ is doing on a low-bandwidth
-// audio test executable.
+#include <algorithm>
 
-int main() {
+#include "webrtc/audio/test/low_bandwidth_audio_test.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/run_test.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;
+}
+
+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");
+}
+
+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";
+}
+
+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);
+}
+
+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(
+    Call* sender_call) {
+  return new test::PacketTransport(
+      sender_call, this, test::PacketTransport::kSender,
+      GetNetworkPipeConfig());
+}
+
+test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
+  return new test::PacketTransport(nullptr, this,
+      test::PacketTransport::kReceiver, GetNetworkPipeConfig());
+}
+
+void AudioQualityTest::ModifyAudioConfigs(
+  AudioSendStream::Config* send_config,
+  std::vector<AudioReceiveStream::Config>* receive_configs) {
+  send_config->send_codec_spec.codec_inst = kDefaultCodec;
+}
+
+void AudioQualityTest::PerformTest() {
+  // 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.codec_inst = CodecInst{
+        120,     // pltype
+        "OPUS",  // plname
+        48000,   // plfreq
+        2880,    // pacsize
+        1,       // channels
+        6000     // rate bits/sec
+    };
+  }
+
+  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