Refactor FakeAudioDevice to have separate methods for starting recording and playout.

webrtc/test/fake_audio_device.cc

Index: webrtc/test/fake_audio_device.cc
diff --git a/webrtc/test/fake_audio_device.cc b/webrtc/test/fake_audio_device.cc
index 9e5e95fb0dbdc229b61dceb7dd182b404c4edc91..57ba15852f548ce6ef7d00e1377bd8410273ff9c 100644
--- a/webrtc/test/fake_audio_device.cc
+++ b/webrtc/test/fake_audio_device.cc
@@ -11,50 +11,83 @@
 #include "webrtc/test/fake_audio_device.h"
 
 #include <algorithm>
+#include <cmath>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/media_file/media_file_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace test {
 
-FakeAudioDevice::FakeAudioDevice(Clock* clock,
-                                 const std::string& filename,
-                                 float speed)
+namespace {
+
+const double kPi = std::acos(-1);
+
+}  // namespace
+
+class FakeAudioDevice::SinusCapturer {
+ public:
+  SinusCapturer(AudioTransport* audio_callback,
+                int frequency_in_hz,
+                int sampling_frequency,
+                uint16_t peak_to_peak)

[peah-webrtc, 2017/01/25 10:45:27]

Do you really need to specify frequency_in_hz and peak_to_peak? Is the ability
to choose those a feature that will be used? Would it be possible to hardcode
the signal features in this class?

[perkj_webrtc, 2017/01/25 13:43:25]

yes, now use pulsed noise as you suggested. 

+      : audio_callback_(audio_callback),
+        normalized_frequency_((2 * kPi * frequency_in_hz) / sampling_frequency),
+        sampling_frequency_(sampling_frequency),
+        peak_to_peak_(peak_to_peak),
+        sine_step_(0) {}
+
+  // Capture |num_samples| of audio data, 2 bytes per sample.
+  void Capture(size_t num_samples) {
+    captured_audio_.SetSize(num_samples);
+    // Get 10ms of audio. 2 bytes per sample.
+    for (size_t i = 0; i < num_samples; ++i) {
+      sine_step_ += normalized_frequency_;

[peah-webrtc, 2017/01/25 10:45:27]

As we discussed offline you can use a pulsed noise sequence here.

Generating int16_t noise values can be done by code similar to

void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v) {
+  for (auto& v_k : v) {
+    v_k = 2 * 32767.f * random_generator->Rand<float>() - 32767.f;
+  }
+}

[perkj_webrtc, 2017/01/25 13:43:25]

Done.

+      if (sine_step_ >= 2 * kPi)
+        sine_step_ -= 2 * kPi;
+      uint16_t sample = static_cast<uint16_t>((1.0 + std::sin(sine_step_)) / 2 *
+                                              peak_to_peak_);
+      captured_audio_[i] = sample;
+    }
+
+    uint32_t new_mic_level;
+    RTC_CHECK_EQ(0, audio_callback_->RecordedDataIsAvailable(
+                        captured_audio_.data(), num_samples, 2, 1,
+                        sampling_frequency_, 0, 0, 0, false, new_mic_level));
+  }
+
+ private:
+  AudioTransport* const audio_callback_;
+  const float normalized_frequency_;
+  const uint32_t sampling_frequency_;
+  const float peak_to_peak_;
+  float sine_step_;
+  rtc::BufferT<uint16_t> captured_audio_;
+};
+
+FakeAudioDevice::FakeAudioDevice(Clock* clock, float speed)
     : audio_callback_(NULL),
-      capturing_(false),
-      captured_audio_(),
+      rendering_(false),
       playout_buffer_(),
       speed_(speed),
       last_playout_ms_(-1),
       clock_(clock, speed),
       tick_(EventTimerWrapper::Create()),
-      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
-      file_utility_(new ModuleFileUtility(0)),
-      input_stream_(FileWrapper::Create()) {
-  memset(captured_audio_, 0, sizeof(captured_audio_));
+      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice") {
   memset(playout_buffer_, 0, sizeof(playout_buffer_));
-  // Open audio input file as read-only and looping.
-  EXPECT_TRUE(input_stream_->OpenFile(filename.c_str(), true)) << filename;
 }
 
 FakeAudioDevice::~FakeAudioDevice() {
-  Stop();
-
+  StopPlayout();
+  StopRecording();
   thread_.Stop();
 }
 
 int32_t FakeAudioDevice::Init() {
-  rtc::CritScope cs(&lock_);
-  if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
-    return -1;
-
-  if (!tick_->StartTimer(true, 10 / speed_))
-    return -1;
+  RTC_CHECK(tick_->StartTimer(true, 10 / speed_));
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);
   return 0;
@@ -68,7 +101,7 @@ int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
 
 bool FakeAudioDevice::Playing() const {
   rtc::CritScope cs(&lock_);
-  return capturing_;
+  return rendering_;
 }
 
 int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
@@ -78,36 +111,22 @@ int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
 
 bool FakeAudioDevice::Recording() const {
   rtc::CritScope cs(&lock_);
-  return capturing_;
+  return capturer_ != nullptr;

[peah-webrtc, 2017/01/25 10:45:27]

Since capturer_ is a unique_ptr, you can return whether it is set without
checking for it to be null,
eg.
"return capturer_;"

(or maybe "return !!capturer_;"

[perkj_webrtc, 2017/01/25 13:43:25]

Done.

 }
 
 bool FakeAudioDevice::Run(void* obj) {
-  static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
+  static_cast<FakeAudioDevice*>(obj)->ProcessAudio();
   return true;
 }
 
-void FakeAudioDevice::CaptureAudio() {
+void FakeAudioDevice::ProcessAudio() {
   {
     rtc::CritScope cs(&lock_);
-    if (capturing_) {
-      int bytes_read = file_utility_->ReadPCMData(
-          *input_stream_.get(), captured_audio_, kBufferSizeBytes);
-      if (bytes_read <= 0)
-        return;
-      // 2 bytes per sample.
-      size_t num_samples = static_cast<size_t>(bytes_read / 2);
-      uint32_t new_mic_level;
-      EXPECT_EQ(0,
-                audio_callback_->RecordedDataIsAvailable(captured_audio_,
-                                                         num_samples,
-                                                         2,
-                                                         1,
-                                                         kFrequencyHz,
-                                                         0,
-                                                         0,
-                                                         0,
-                                                         false,
-                                                         new_mic_level));
+    if (capturer_) {
+      // Capture 10ms of audio. 2 bytes per sample.
+      capturer_->Capture(kFrequencyHz / 100);
+    }
+    if (rendering_) {
       size_t samples_needed = kFrequencyHz / 100;
       int64_t now_ms = clock_.TimeInMilliseconds();
       uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
@@ -119,28 +138,38 @@ void FakeAudioDevice::CaptureAudio() {
       size_t samples_out = 0;
       int64_t elapsed_time_ms = -1;
       int64_t ntp_time_ms = -1;
-      EXPECT_EQ(0,
-                audio_callback_->NeedMorePlayData(samples_needed,
-                                                  2,
-                                                  1,
-                                                  kFrequencyHz,
-                                                  playout_buffer_,
-                                                  samples_out,
-                                                  &elapsed_time_ms,
-                                                  &ntp_time_ms));
+      RTC_CHECK_EQ(0, audio_callback_->NeedMorePlayData(
+                          samples_needed, 2, 1, kFrequencyHz, playout_buffer_,
+                          samples_out, &elapsed_time_ms, &ntp_time_ms));
     }
   }
   tick_->Wait(WEBRTC_EVENT_INFINITE);
 }
 
-void FakeAudioDevice::Start() {
+int32_t FakeAudioDevice::StartPlayout() {

[peah-webrtc, 2017/01/25 10:45:28]

Please order the methods in the same order as they are in the class definition.

[perkj_webrtc, 2017/01/25 13:43:25]

Done.

   rtc::CritScope cs(&lock_);
-  capturing_ = true;
+  rendering_ = true;
+  return 0;
 }
 
-void FakeAudioDevice::Stop() {
+int32_t FakeAudioDevice::StopPlayout() {
   rtc::CritScope cs(&lock_);
-  capturing_ = false;
+  rendering_ = false;
+  return 0;
 }
+
+void FakeAudioDevice::StartRecordingSine(int frequency_in_hz,
+                                         uint16_t peak_to_peak) {
+  rtc::CritScope cs(&lock_);
+  capturer_.reset(new FakeAudioDevice::SinusCapturer(
+      audio_callback_, frequency_in_hz, kFrequencyHz, peak_to_peak));
+}
+
+int32_t FakeAudioDevice::StopRecording() {
+  rtc::CritScope cs(&lock_);
+  capturer_.reset();
+  return 0;
+}
+
 }  // namespace test
 }  // namespace webrtc