Add the ability to read/write to WAV files in FakeAudioDevice

webrtc/test/file_audio_device.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 "webrtc/test/file_audio_device.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFrameLengthMs = 10;
+constexpr int kFramesPerSecond = 1000 / kFrameLengthMs;
+
+int NumSamplesPerFrame(int sample_rate_hz) {
+  return rtc::CheckedDivExact(sample_rate_hz, kFramesPerSecond);
+}
+
+}  // namespace
+namespace test {
+
+FileReaderAudioDevice::FileReaderAudioDevice(std::string filename, float speed,
+                                             int sample_rate_hz)
+    : filename_(filename),
+      sample_rate_hz_(sample_rate_hz),
+      speed_(speed),
+      audio_callback_(nullptr),
+      wav_reader_(nullptr),

[kwiberg-webrtc, 2017/03/07 10:11:57]

unique_ptrs are null by default, so you don't need this line.

+      playout_buffer_(NumSamplesPerFrame(sample_rate_hz), 0),
+      tick_(EventTimerWrapper::Create()),
+      thread_(FileReaderAudioDevice::Run, this, "FileReaderAudioDevice"),
+      done_reading_(true, true) {
+  RTC_CHECK(
+      sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+      sample_rate_hz == 32000 || sample_rate_hz == 44100 ||
+      sample_rate_hz == 48000);
+}
+
+FileReaderAudioDevice::~FileReaderAudioDevice() {
+  StopRecording();
+}
+
+int32_t FileReaderAudioDevice::StartRecording() {
+  rtc::CritScope cs(&lock_);
+  RTC_CHECK(!wav_reader_);
+  wav_reader_.reset(new WavReader(filename_));
+  RTC_CHECK(wav_reader_->sample_rate() == sample_rate_hz_);
+  RTC_CHECK(wav_reader_->num_channels() == 1);

[kwiberg-webrtc, 2017/03/07 10:11:57]

RTC_CHECK_EQ for these two.

[oprypin_webrtc, 2017/03/09 08:23:24]

Done.

+
+  RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
+  done_reading_.Reset();
+  return 0;
+}
+
+int32_t FileReaderAudioDevice::StopRecording() {
+  rtc::CritScope cs(&lock_);
+  wav_reader_.reset();
+  thread_.Stop();
+  return 0;
+}
+
+int32_t FileReaderAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
+  rtc::CritScope cs(&lock_);
+  RTC_DCHECK(callback || audio_callback_ != nullptr);

[kwiberg-webrtc, 2017/03/07 10:11:57]

Test code, so CHECK instead of DCHECK everywhere?

Also, for consistency, either

  callback || audio_callback_

or

  callback != nullptr || audio_callback_ != nullptr

[oprypin_webrtc, 2017/03/09 08:23:24]

Done.

+  audio_callback_ = callback;
+  return 0;
+}
+
+bool FileReaderAudioDevice::Recording() const {
+  rtc::CritScope cs(&lock_);
+  return static_cast<bool>(wav_reader_);
+}
+
+bool FileReaderAudioDevice::WaitForFileEnd(int timeout_ms) {
+  RTC_DCHECK(Recording());
+  return done_reading_.Wait(timeout_ms);
+}
+
+bool FileReaderAudioDevice::Run(void* obj) {
+  static_cast<FileReaderAudioDevice*>(obj)->ProcessAudio();
+  return true;
+}
+
+void FileReaderAudioDevice::ProcessAudio() {
+  {
+    rtc::CritScope cs(&lock_);
+    // Capture 10ms of audio. 2 bytes per sample.
+    const size_t samples_out = wav_reader_->ReadSamples(
+        playout_buffer_.size(), playout_buffer_.data());
+    if (samples_out) {

[kwiberg-webrtc, 2017/03/07 10:11:57]

if (samples_out > 0)

to emphasize that samples_out is a number, not some sort of boolean.

[oprypin_webrtc, 2017/03/09 08:23:24]

Done.

+      RTC_CHECK(samples_out <= playout_buffer_.size());

[kwiberg-webrtc, 2017/03/07 10:11:57]

RTC_CHECK_LE

Also, move this check before line 102? It seems to apply to both if branches.

[oprypin_webrtc, 2017/03/09 08:23:24]

Done.

+      uint32_t new_mic_level;
+      audio_callback_->RecordedDataIsAvailable(
+          playout_buffer_.data(), samples_out, 2, 1,
+          sample_rate_hz_, 0, 0, 0, false, new_mic_level);
+    } else {
+      done_reading_.Set();
+    }
+  }
+  tick_->Wait(WEBRTC_EVENT_INFINITE);
+}
+
+
+FileWriterAudioDevice::FileWriterAudioDevice(std::string filename, float speed,
+                                             int sample_rate_hz)
+    : filename_(filename),
+      sample_rate_hz_(sample_rate_hz),
+      speed_(speed),
+      audio_callback_(nullptr),
+      wav_writer_(nullptr),
+      playout_buffer_(NumSamplesPerFrame(sample_rate_hz), 0),
+      tick_(EventTimerWrapper::Create()),
+      thread_(FileWriterAudioDevice::Run, this, "FileWriterAudioDevice") {
+  RTC_CHECK(
+      sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+      sample_rate_hz == 32000 || sample_rate_hz == 44100 ||
+      sample_rate_hz == 48000);
+}
+
+FileWriterAudioDevice::~FileWriterAudioDevice() {
+  StopPlayout();
+}
+
+int32_t FileWriterAudioDevice::StartPlayout() {
+  rtc::CritScope cs(&lock_);
+  RTC_CHECK(!wav_writer_);
+  wav_writer_.reset(new WavWriter(filename_, sample_rate_hz_, 1));
+
+  RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
+  return 0;
+}
+
+int32_t FileWriterAudioDevice::StopPlayout() {
+  rtc::CritScope cs(&lock_);
+  wav_writer_.reset();  // This also finalizes and closes the file
+  thread_.Stop();
+  return 0;
+}
+
+int32_t FileWriterAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
+  rtc::CritScope cs(&lock_);
+  RTC_DCHECK(callback || audio_callback_ != nullptr);
+  audio_callback_ = callback;
+  return 0;
+}
+
+bool FileWriterAudioDevice::Playing() const {
+  rtc::CritScope cs(&lock_);
+  return static_cast<bool>(wav_writer_);
+}
+
+bool FileWriterAudioDevice::Run(void* obj) {
+  static_cast<FileWriterAudioDevice*>(obj)->ProcessAudio();
+  return true;
+}
+
+void FileWriterAudioDevice::ProcessAudio() {
+  {
+    rtc::CritScope cs(&lock_);
+    size_t samples_out;
+    int64_t elapsed_time_ms;
+    int64_t ntp_time_ms;
+    audio_callback_->NeedMorePlayData(
+        playout_buffer_.size(), 2, 1, sample_rate_hz_,
+        playout_buffer_.data(), samples_out, &elapsed_time_ms, &ntp_time_ms);
+    wav_writer_->WriteSamples(playout_buffer_.data(), samples_out);
+  }
+  tick_->Wait(WEBRTC_EVENT_INFINITE);
+}
+
+}  // namespace test
+}  // namespace webrtc