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..047315ec80fc08c6ee286da439302b60d6f1dd6a 100644
--- a/webrtc/test/fake_audio_device.cc
+++ b/webrtc/test/fake_audio_device.cc
@@ -12,49 +12,104 @@
 
 #include <algorithm>
 
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/media_file/media_file_utility.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/random.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 {
+
+constexpr int kFrameLengthMs = 10;
+constexpr int kFramesPerSecond = 1000 / kFrameLengthMs;
+
+}  // namespace
 namespace test {
 
-FakeAudioDevice::FakeAudioDevice(Clock* clock,
-                                 const std::string& filename,
-                                 float speed)
-    : audio_callback_(NULL),
+// Assuming 10ms audio packets..
+class FakeAudioDevice::PulsedNoiseCapturer {
+ public:
+  PulsedNoiseCapturer(int num_samples_per_frame, int16_t max_amplitude)
+      : fill_with_zero_(false),
+        random_generator_(1),
+        max_amplitude_(max_amplitude),
+        random_audio_(num_samples_per_frame),
+        silent_audio_(num_samples_per_frame, 0) {
+    RTC_DCHECK_GT(max_amplitude, 0);

[the sun, 2017/01/31 16:27:08]

nit: DHCECK num_samples_per_frame >= 0

[peah-webrtc, 2017/01/31 16:37:33]

Or specify it as a size_t.

[perkj_webrtc, 2017/01/31 19:11:07]

size_t since that is what the it is being used as. Despite our previous
discussion about style guide... 

+  }
+
+  rtc::ArrayView<const int16_t> Capture() {
+    fill_with_zero_ = !fill_with_zero_;
+    if (!fill_with_zero_) {
+      std::generate(random_audio_.begin(), random_audio_.end(), [&]() {
+        return random_generator_.Rand(-max_amplitude_, max_amplitude_);
+      });
+    }
+    return fill_with_zero_ ? silent_audio_ : random_audio_;
+  }
+
+ private:
+  bool fill_with_zero_;
+  Random random_generator_;
+  const int16_t max_amplitude_;
+  std::vector<int16_t> random_audio_;
+  std::vector<int16_t> silent_audio_;
+};
+
+FakeAudioDevice::FakeAudioDevice(float speed,
+                                 int sampling_frequency_in_hz,
+                                 int16_t max_amplitude)
+    : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+      num_samples_per_frame_(
+          rtc::CheckedDivExact(sampling_frequency_in_hz_, kFramesPerSecond)),
+      audio_callback_(nullptr),
+      rendering_(false),
       capturing_(false),
-      captured_audio_(),
-      playout_buffer_(),
+      capturer_(new FakeAudioDevice::PulsedNoiseCapturer(num_samples_per_frame_,
+                                                         max_amplitude)),
+      playout_buffer_(num_samples_per_frame_, 0),
       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_));
-  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;
+      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice") {
+  RTC_DCHECK(
+      sampling_frequency_in_hz == 8000 || sampling_frequency_in_hz == 16000 ||
+      sampling_frequency_in_hz == 32000 || sampling_frequency_in_hz == 44100 ||
+      sampling_frequency_in_hz == 48000);
 }
 
 FakeAudioDevice::~FakeAudioDevice() {
-  Stop();
-
+  StopPlayout();
+  StopRecording();
   thread_.Stop();
 }
 
-int32_t FakeAudioDevice::Init() {
+int32_t FakeAudioDevice::StartPlayout() {
+  rtc::CritScope cs(&lock_);
+  rendering_ = true;
+  return 0;
+}
+
+int32_t FakeAudioDevice::StopPlayout() {
+  rtc::CritScope cs(&lock_);
+  rendering_ = false;
+  return 0;
+}
+
+int32_t FakeAudioDevice::StartRecording() {
+  rtc::CritScope cs(&lock_);
+  capturing_ = true;
+  return 0;
+}
+
+int32_t FakeAudioDevice::StopRecording() {
   rtc::CritScope cs(&lock_);
-  if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
-    return -1;
+  capturing_ = false;
+  return 0;
+}
 
-  if (!tick_->StartTimer(true, 10 / speed_))
-    return -1;
+int32_t FakeAudioDevice::Init() {
+  RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);
   return 0;
@@ -62,18 +117,14 @@ int32_t FakeAudioDevice::Init() {
 
 int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
   rtc::CritScope cs(&lock_);
+  RTC_DCHECK(callback || audio_callback_ != nullptr);
   audio_callback_ = callback;
   return 0;
 }
 
 bool FakeAudioDevice::Playing() const {
   rtc::CritScope cs(&lock_);
-  return capturing_;
-}
-
-int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
-  *delay_ms = 0;
-  return 0;
+  return rendering_;
 }
 
 bool FakeAudioDevice::Recording() const {
@@ -82,65 +133,33 @@ bool FakeAudioDevice::Recording() const {
 }
 
 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));
-      size_t samples_needed = kFrequencyHz / 100;
-      int64_t now_ms = clock_.TimeInMilliseconds();
-      uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
-      if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
-        samples_needed = std::min(
-            static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
-            kBufferSizeBytes / 2);
-      }
+      // Capture 10ms of audio. 2 bytes per sample.
+      rtc::ArrayView<const int16_t> audio_data = capturer_->Capture();
+      uint32_t new_mic_level = 0;
+      audio_callback_->RecordedDataIsAvailable(
+          audio_data.data(), audio_data.size(), 2, 1, sampling_frequency_in_hz_,
+          0, 0, 0, false, new_mic_level);
+    }
+    if (rendering_) {
       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));
+      audio_callback_->NeedMorePlayData(
+          num_samples_per_frame_, 2, 1, sampling_frequency_in_hz_,
+          playout_buffer_.data(), samples_out, &elapsed_time_ms, &ntp_time_ms);
     }
   }
   tick_->Wait(WEBRTC_EVENT_INFINITE);
 }
 
-void FakeAudioDevice::Start() {
-  rtc::CritScope cs(&lock_);
-  capturing_ = true;
-}
 
-void FakeAudioDevice::Stop() {
-  rtc::CritScope cs(&lock_);
-  capturing_ = false;
-}
 }  // namespace test
 }  // namespace webrtc