Low-bandwidth audio testing

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
- *  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
  *  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.
  */
 
-// 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 webrtc {
+namespace test {
+
+// Writes to a WAV file, cutting off silence at the beginning and the end.
+class BoundedWavFileWriter : public test::FakeAudioDevice::Renderer {

[kjellander_webrtc, 2017/03/17 07:11:41]

I think BoundedWavFileWriter is large enough for its own .cc file + unit test.

[oprypin_webrtc, 2017/03/20 10:39:27]

We discussed this and came to conclusion that it's not worth putting it in a
separate .cc file. Moved it to fake_audio_device and added unittest.

[kjellander_webrtc, 2017/03/21 08:31:49]

Actually, I thought we just agreed on where it would be put (webrtc/test) I
still think it would be cleaner to move it to its own .cc file along with the
unit test.

[kjellander_webrtc, 2017/03/21 08:43:41]

I've now been convinced this is fine. Let's keep it here.

+ public:
+  BoundedWavFileWriter(std::string filename, int sampling_frequency_in_hz)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        wav_writer_(filename, sampling_frequency_in_hz, 1),
+        silent_audio_(test::FakeAudioDevice::SamplesPerFrame(
+            sampling_frequency_in_hz), 0),
+        started_writing_(false),
+        trailing_zeros_(0) {}
+
+  int SamplingFrequency() const override {
+    return sampling_frequency_in_hz_;
+  }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    const int16_t kAmplitudeThreshold = 5;
+
+    const int16_t* begin = data.begin();
+    const int16_t* end = data.end();
+    if (!started_writing_) {
+      // Cut off silence at the beginning.
+      while (begin < end) {
+        if (*begin > kAmplitudeThreshold || *begin < -kAmplitudeThreshold) {
+          started_writing_ = true;
+          break;
+        }
+        ++begin;
+      }
+    }
+    if (started_writing_) {
+      // Cut off silence at the end.
+      while (begin < end) {
+        if (*(end - 1) != 0) {
+          break;
+        }
+        --end;
+        ++trailing_zeros_;
+      }
+      if (begin < end) {
+        // If it turns out that the silence was not final, need to write all the
+        // skipped zeros and continue writing audio.
+        while (trailing_zeros_ > 0) {
+          const size_t zeros_to_write = std::min(trailing_zeros_,
+                                                 silent_audio_.size());
+          wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write);
+          trailing_zeros_ -= zeros_to_write;
+        }
+        wav_writer_.WriteSamples(begin, end - begin);
+      }
+    }
+    return true;
+  }
+
+ private:
+  int sampling_frequency_in_hz_;
+  WavWriter wav_writer_;
+  std::vector<int16_t> silent_audio_;
+  bool started_writing_;
+  size_t trailing_zeros_;
+};
+
+
+AudioQualityTest::AudioQualityTest() : EndToEndTest(1000) {}

[kjellander_webrtc, 2017/03/17 07:11:41]

It's not clear what 1000 is here. Move into a constant.

[oprypin_webrtc, 2017/03/17 10:16:51]

Done.

+
+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 std::unique_ptr<test::FakeAudioDevice::Renderer>(
+      new BoundedWavFileWriter(AudioOutputFile(), 16000));

[kjellander_webrtc, 2017/03/17 07:11:41]

Extract into a constant or even a gflags flag, since the python script could
pass it (as it also needs it for PESQ).

[oprypin_webrtc, 2017/03/17 10:16:51]

PESQ has "Sample rate [...] Must select either +8000 or +16000."
I hardcoded this because only 16000 makes sense under these circumstances. It's
good to make it a constant (done).
A flag would also be good in case other tools are used, but can we agree to do
that when the need arises?

[kjellander_webrtc, 2017/03/17 10:37:18]

Yes, let's keep it as a constant.

+}
+
+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) {
+  // Large bitrate by default.
+  send_config->send_codec_spec.codec_inst =
+      CodecInst{120, "OPUS", 48000, 960, 2, 64000};

[kjellander_webrtc, 2017/03/17 07:11:41]

This default is kind of hidden. Can you store it as a const variable for the
class?

[oprypin_webrtc, 2017/03/17 10:16:51]

Done.

+}
+
+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 + 500);

[kjellander_webrtc, 2017/03/17 07:11:41]

Extract 500 into a constant.

[oprypin_webrtc, 2017/03/17 10:16:51]

Done.

+}
+
+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 = test::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;

[stefan-webrtc, 2017/03/20 14:40:10]

I think a longer queue should be used as thats what we've experienced in the
wild

+    pipe_config.queue_delay_ms = 400;
+    return pipe_config;
+  }
+};
+
+TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
+  Mobile2GNetworkTest test;
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc