Refactor AudioInputController and related interfaces.

media/audio/audio_input_controller.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/audio_input_controller.h"
 
+#include <inttypes.h>
+
 #include <algorithm>
 #include <limits>
 #include <utility>
 
 #include "base/bind.h"
 #include "base/metrics/histogram_macros.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/strings/stringprintf.h"
 #include "base/threading/thread_restrictions.h"
 #include "base/threading/thread_task_runner_handle.h"
 #include "base/time/time.h"
 #include "base/trace_event/trace_event.h"
 #include "media/audio/audio_file_writer.h"
 #include "media/base/user_input_monitor.h"
 
+namespace media {
 namespace {
 
 const int kMaxInputChannels = 3;
 
 #if defined(AUDIO_POWER_MONITORING)
 // Time in seconds between two successive measurements of audio power levels.
 const int kPowerMonitorLogIntervalSeconds = 15;
 
 // A warning will be logged when the microphone audio volume is below this
 // threshold.
 const int kLowLevelMicrophoneLevelPercent = 10;
 
 // Logs if the user has enabled the microphone mute or not. This is normally
 // done by marking a checkbox in an audio-settings UI which is unique for each
 // platform. Elements in this enum should not be added, deleted or rearranged.
 enum MicrophoneMuteResult {
   MICROPHONE_IS_MUTED = 0,
   MICROPHONE_IS_NOT_MUTED = 1,
   MICROPHONE_MUTE_MAX = MICROPHONE_IS_NOT_MUTED
 };
 
 void LogMicrophoneMuteResult(MicrophoneMuteResult result) {
   UMA_HISTOGRAM_ENUMERATION("Media.MicrophoneMuted",
                             result,
                             MICROPHONE_MUTE_MAX + 1);
 }
 
 // Helper method which calculates the average power of an audio bus. Unit is in
 // dBFS, where 0 dBFS corresponds to all channels and samples equal to 1.0.
-float AveragePower(const media::AudioBus& buffer) {
+float AveragePower(const AudioBus& buffer) {
   const int frames = buffer.frames();
   const int channels = buffer.channels();
   if (frames <= 0 || channels <= 0)
     return 0.0f;
 
   // Scan all channels and accumulate the sum of squares for all samples.
   float sum_power = 0.0f;
   for (int ch = 0; ch < channels; ++ch) {
     const float* channel_data = buffer.channel(ch);
     for (int i = 0; i < frames; i++) {
@@ skipping 11 matching lines @@
   const float kInsignificantPower = 1.0e-10f;  // -100 dBFS
   const float power_dbfs = average_power < kInsignificantPower ?
       -std::numeric_limits<float>::infinity() : 10.0f * log10f(average_power);
 
   return power_dbfs;
 }
 #endif  // AUDIO_POWER_MONITORING
 
 }  // namespace
 
-namespace media {
+// Private subclass of AIC that covers the state while capturing audio.
+// This class implements the callback interface from the lower level audio
+// layer and gets called back on the audio hw thread.
+// We implement this in a sub class instead of directly in the AIC so that
+// - The AIC itself is not an AudioInputCallback.
+// - The lifetime of the AudioCallback is shorter than the AIC
+// - How tasks are posted to the AIC from the hw callback thread, is different
+//   than how tasks are posted from the AIC to itself from the main thread.
+//   So, this difference is isolated to the subclass (see below).
+// - The callback class can gather information on what happened during capture
+//   and store it in a state that can be fetched after stopping capture
+//   (received_callback, error_during_callback).
+// The AIC itself must not be AddRef-ed on the hw callback thread so that we
+// can be guaranteed to not receive callbacks generated by the hw callback
+// thread after Close() has been called on the audio manager thread and
+// the callback object deleted. To avoid AddRef-ing the AIC and to cancel
+// potentially pending tasks, we use a weak pointer to the AIC instance
+// when posting.
+class AudioInputController::AudioCallback
+    : public AudioInputStream::AudioInputCallback {
+ public:
+  explicit AudioCallback(AudioInputController* controller)
+      : controller_(controller),
+        weak_controller_(controller->weak_ptr_factory_.GetWeakPtr()) {}
+  ~AudioCallback() override {}
+
+  bool received_callback() const { return received_callback_; }
+  bool error_during_callback() const { return error_during_callback_; }
+
+ private:
+  void OnData(AudioInputStream* stream,
+              const AudioBus* source,
+              uint32_t hardware_delay_bytes,
+              double volume) override {
+    TRACE_EVENT0("audio", "AC::OnData");
+
+    received_callback_ = true;
+
+    DeliverDataToSyncWriter(source, hardware_delay_bytes, volume);
+    PerformOptionalDebugRecording(source);
+  }
+
+  void OnError(AudioInputStream* stream) override {
+    error_during_callback_ = true;
+    controller_->task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(&AudioInputController::DoReportError, weak_controller_));
+  }
+
+  void PerformOptionalDebugRecording(const AudioBus* source) {
+    // Called on the hw callback thread while recording is enabled.
+    if (!controller_->debug_writer_ || !controller_->debug_writer_->WillWrite())
+      return;
+
+    // TODO(tommi): This is costly. AudioBus heap allocs and we create a new
+    // one for every callback. We could instead have a pool of bus objects
+    // that get returned to us somehow.
+    // We should also avoid calling PostTask here since the implementation
+    // of the debug writer will basically do a PostTask straight away anyway.
+    // Might require some modifications to AudioInputDebugWriter though since
+    // there are some threading concerns there and AudioInputDebugWriter's
+    // lifetime guarantees need to be longer than that of associated active
+    // audio streams.
+    std::unique_ptr<AudioBus> source_copy =
+        AudioBus::Create(source->channels(), source->frames());
+    source->CopyTo(source_copy.get());
+    controller_->task_runner_->PostTask(
+        FROM_HERE, base::Bind(&AudioInputController::WriteInputDataForDebugging,
+                              weak_controller_, base::Passed(&source_copy)));
+  }
+
+  void DeliverDataToSyncWriter(const AudioBus* source,
+                               uint32_t hardware_delay_bytes,
+                               double volume) {
+    bool key_pressed = controller_->CheckForKeyboardInput();
+    controller_->sync_writer_->Write(source, volume, key_pressed,
+                                     hardware_delay_bytes);
+
+    // The way the two classes interact here, could be done in a nicer way.
+    // As is, we call the AIC here to check the audio power, return  and then
+    // post a task to the AIC based on what the AIC said.
+    // The reason for this is to keep all PostTask calls from the hw callback
+    // thread to the AIC, that use a weak pointer, in the same class.
+    float average_power_dbfs;
+    int mic_volume_percent;
+    if (controller_->CheckAudioPower(source, volume, &average_power_dbfs,
+                                     &mic_volume_percent)) {
+      // Use event handler on the audio thread to relay a message to the ARIH
+      // in content which does the actual logging on the IO thread.
+      controller_->task_runner_->PostTask(
+          FROM_HERE,
+          base::Bind(&AudioInputController::DoLogAudioLevels, weak_controller_,
+                     average_power_dbfs, mic_volume_percent));
+    }
+  }
+
+  AudioInputController* const controller_;
+  // We do not want any pending posted tasks generated from the callback class
+  // to keep the controller object alive longer than it should. So we use
+  // a weak ptr whenever we post, we use this weak pointer.
+  base::WeakPtr<AudioInputController> weak_controller_;
+  bool received_callback_ = false;
+  bool error_during_callback_ = false;
+};
 
 // static
 AudioInputController::Factory* AudioInputController::factory_ = nullptr;
 
 AudioInputController::AudioInputController(
+    scoped_refptr<base::SingleThreadTaskRunner> task_runner,
     EventHandler* handler,
     SyncWriter* sync_writer,
     std::unique_ptr<AudioFileWriter> debug_writer,
     UserInputMonitor* user_input_monitor,
     const bool agc_is_enabled)
     : creator_task_runner_(base::ThreadTaskRunnerHandle::Get()),
+      task_runner_(std::move(task_runner)),
       handler_(handler),
       stream_(nullptr),
-      should_report_stats(0),
-      state_(CLOSED),
       sync_writer_(sync_writer),
       max_volume_(0.0),
       user_input_monitor_(user_input_monitor),
       agc_is_enabled_(agc_is_enabled),
 #if defined(AUDIO_POWER_MONITORING)
       power_measurement_is_enabled_(false),
       log_silence_state_(false),
       silence_state_(SILENCE_STATE_NO_MEASUREMENT),
 #endif
       prev_key_down_count_(0),
-      debug_writer_(std::move(debug_writer)) {
+      debug_writer_(std::move(debug_writer)),
+      weak_ptr_factory_(this) {
   DCHECK(creator_task_runner_.get());
+  DCHECK(handler_);
+  DCHECK(sync_writer_);
 }
 
 AudioInputController::~AudioInputController() {
-  DCHECK_EQ(state_, CLOSED);
+  DCHECK(!audio_callback_);
+  DCHECK(!stream_);
 }
 
 // static
 scoped_refptr<AudioInputController> AudioInputController::Create(
     AudioManager* audio_manager,
     EventHandler* event_handler,
+    SyncWriter* sync_writer,
     const AudioParameters& params,
     const std::string& device_id,
     UserInputMonitor* user_input_monitor) {
   DCHECK(audio_manager);
+  DCHECK(event_handler);
+  DCHECK(sync_writer);
 
   if (!params.IsValid() || (params.channels() > kMaxInputChannels))
     return nullptr;
 
   if (factory_) {
-    return factory_->Create(audio_manager->GetTaskRunner(),
-                            /*sync_writer*/ nullptr, audio_manager,
-                            event_handler, params, user_input_monitor);
+    return factory_->Create(audio_manager->GetTaskRunner(), sync_writer,
+                            audio_manager, event_handler, params,
+                            user_input_monitor);
   }
 
   scoped_refptr<AudioInputController> controller(new AudioInputController(
-      event_handler, nullptr, nullptr, user_input_monitor, false));
-
-  controller->task_runner_ = audio_manager->GetTaskRunner();
+      audio_manager->GetTaskRunner(), event_handler, sync_writer, nullptr,
+      user_input_monitor, false));
 
   // Create and open a new audio input stream from the existing
   // audio-device thread.
   if (!controller->task_runner_->PostTask(
           FROM_HERE,
           base::Bind(&AudioInputController::DoCreate,
                      controller,
                      base::Unretained(audio_manager),
                      params,
                      device_id))) {
     controller = nullptr;
   }
 
   return controller;
 }
 
 // static
 scoped_refptr<AudioInputController> AudioInputController::CreateLowLatency(
     AudioManager* audio_manager,
     EventHandler* event_handler,
     const AudioParameters& params,
     const std::string& device_id,
     SyncWriter* sync_writer,
     std::unique_ptr<AudioFileWriter> debug_writer,
     UserInputMonitor* user_input_monitor,
     const bool agc_is_enabled) {
   DCHECK(audio_manager);
   DCHECK(sync_writer);
+  DCHECK(event_handler);
 
   if (!params.IsValid() || (params.channels() > kMaxInputChannels))
     return nullptr;
 
   if (factory_) {
     return factory_->Create(audio_manager->GetTaskRunner(), sync_writer,
                             audio_manager, event_handler, params,
                             user_input_monitor);
   }
 
   // Create the AudioInputController object and ensure that it runs on
   // the audio-manager thread.
   scoped_refptr<AudioInputController> controller(new AudioInputController(
-      event_handler, sync_writer, std::move(debug_writer), user_input_monitor,
-      agc_is_enabled));
-  controller->task_runner_ = audio_manager->GetTaskRunner();
+      audio_manager->GetTaskRunner(), event_handler, sync_writer,
+      std::move(debug_writer), user_input_monitor, agc_is_enabled));
 
   // Create and open a new audio input stream from the existing
   // audio-device thread. Use the provided audio-input device.
   if (!controller->task_runner_->PostTask(
           FROM_HERE,
           base::Bind(&AudioInputController::DoCreateForLowLatency,
                      controller,
                      base::Unretained(audio_manager),
                      params,
                      device_id))) {
     controller = nullptr;
   }
 
   return controller;
 }
 
 // static
 scoped_refptr<AudioInputController> AudioInputController::CreateForStream(
     const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
     EventHandler* event_handler,
     AudioInputStream* stream,
     SyncWriter* sync_writer,
     std::unique_ptr<AudioFileWriter> debug_writer,
     UserInputMonitor* user_input_monitor) {
   DCHECK(sync_writer);
   DCHECK(stream);
+  DCHECK(event_handler);
 
   if (factory_) {
     return factory_->Create(
         task_runner, sync_writer, AudioManager::Get(), event_handler,
-        media::AudioParameters::UnavailableDeviceParams(), user_input_monitor);
+        AudioParameters::UnavailableDeviceParams(), user_input_monitor);
   }
 
   // Create the AudioInputController object and ensure that it runs on
   // the audio-manager thread.
   scoped_refptr<AudioInputController> controller(new AudioInputController(
-      event_handler, sync_writer, std::move(debug_writer), user_input_monitor,
-      false));
-  controller->task_runner_ = task_runner;
+      task_runner, event_handler, sync_writer, std::move(debug_writer),
+      user_input_monitor, false));
 
   if (!controller->task_runner_->PostTask(
           FROM_HERE,
           base::Bind(&AudioInputController::DoCreateForStream,
                      controller,
                      stream))) {
     controller = nullptr;
   }
 
   return controller;
 }
 
 void AudioInputController::Record() {
+  DCHECK(creator_task_runner_->BelongsToCurrentThread());
   task_runner_->PostTask(FROM_HERE, base::Bind(
       &AudioInputController::DoRecord, this));
 }
 
 void AudioInputController::Close(const base::Closure& closed_task) {
   DCHECK(!closed_task.is_null());
   DCHECK(creator_task_runner_->BelongsToCurrentThread());
 
   task_runner_->PostTaskAndReply(
       FROM_HERE, base::Bind(&AudioInputController::DoClose, this), closed_task);
 }
 
 void AudioInputController::SetVolume(double volume) {
+  DCHECK(creator_task_runner_->BelongsToCurrentThread());
   task_runner_->PostTask(FROM_HERE, base::Bind(
       &AudioInputController::DoSetVolume, this, volume));
 }
 
 void AudioInputController::DoCreate(AudioManager* audio_manager,
                                     const AudioParameters& params,
                                     const std::string& device_id) {
   DCHECK(task_runner_->BelongsToCurrentThread());
+  DCHECK(!stream_);
   SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.CreateTime");
-  if (handler_)
-    handler_->OnLog(this, "AIC::DoCreate");
+  handler_->OnLog(this, "AIC::DoCreate");
 
 #if defined(AUDIO_POWER_MONITORING)
   // Disable power monitoring for streams that run without AGC enabled to
   // avoid adding logs and UMA for non-WebRTC clients.
   power_measurement_is_enabled_ = agc_is_enabled_;
   last_audio_level_log_time_ = base::TimeTicks::Now();
   silence_state_ = SILENCE_STATE_NO_MEASUREMENT;
 #endif
 
+  // MakeAudioInputStream might fail and return nullptr. If so,
+  // DoCreateForStream will handle and report it.
   DoCreateForStream(audio_manager->MakeAudioInputStream(
       params, device_id, base::Bind(&AudioInputController::LogMessage, this)));
 }
 
 void AudioInputController::DoCreateForLowLatency(AudioManager* audio_manager,
                                                  const AudioParameters& params,
                                                  const std::string& device_id) {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
 #if defined(AUDIO_POWER_MONITORING)
   // We only log silence state UMA stats for low latency mode and if we use a
   // real device.
   if (params.format() != AudioParameters::AUDIO_FAKE)
     log_silence_state_ = true;
 #endif
 
   low_latency_create_time_ = base::TimeTicks::Now();
   DoCreate(audio_manager, params, device_id);
 }
 
 void AudioInputController::DoCreateForStream(
     AudioInputStream* stream_to_control) {
   DCHECK(task_runner_->BelongsToCurrentThread());
+  DCHECK(!stream_);
 
-  DCHECK(!stream_);
-  stream_ = stream_to_control;
-  should_report_stats = 1;
-
-  if (!stream_) {
-    if (handler_)
-      handler_->OnError(this, STREAM_CREATE_ERROR);
+  if (!stream_to_control) {
     LogCaptureStartupResult(CAPTURE_STARTUP_CREATE_STREAM_FAILED);
+    handler_->OnError(this, STREAM_CREATE_ERROR);
     return;
   }
 
-  if (stream_ && !stream_->Open()) {
-    stream_->Close();
-    stream_ = nullptr;
-    if (handler_)
-      handler_->OnError(this, STREAM_OPEN_ERROR);
+  if (!stream_to_control->Open()) {
+    stream_to_control->Close();
     LogCaptureStartupResult(CAPTURE_STARTUP_OPEN_STREAM_FAILED);
+    handler_->OnError(this, STREAM_OPEN_ERROR);
     return;
   }
 
   // Set AGC state using mode in |agc_is_enabled_| which can only be enabled in
   // CreateLowLatency().
 #if defined(AUDIO_POWER_MONITORING)
-  bool agc_is_supported = false;
-  agc_is_supported = stream_->SetAutomaticGainControl(agc_is_enabled_);
+  bool agc_is_supported =
+      stream_to_control->SetAutomaticGainControl(agc_is_enabled_);
   // Disable power measurements on platforms that does not support AGC at a
   // lower level. AGC can fail on platforms where we don't support the
   // functionality to modify the input volume slider. One such example is
   // Windows XP.
   power_measurement_is_enabled_ &= agc_is_supported;
 #else
-  stream_->SetAutomaticGainControl(agc_is_enabled_);
+  stream_to_control->SetAutomaticGainControl(agc_is_enabled_);
 #endif
 
-
-  state_ = CREATED;
-  if (handler_)
-    handler_->OnCreated(this);
-
-  if (user_input_monitor_) {
-    user_input_monitor_->EnableKeyPressMonitoring();
-    prev_key_down_count_ = user_input_monitor_->GetKeyPressCount();
-  }
+  // Finally, keep the stream pointer around, update the state and notify.
+  stream_ = stream_to_control;
+  handler_->OnCreated(this);
 }
 
 void AudioInputController::DoRecord() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.RecordTime");
 
-  if (state_ != CREATED)
+  if (!stream_ || audio_callback_)
     return;
 
-  {
-    base::AutoLock auto_lock(lock_);
-    state_ = RECORDING;
+  handler_->OnLog(this, "AIC::DoRecord");
+
+  if (user_input_monitor_) {
+    user_input_monitor_->EnableKeyPressMonitoring();
+    prev_key_down_count_ = user_input_monitor_->GetKeyPressCount();
   }
 
-  if (handler_)
-    handler_->OnLog(this, "AIC::DoRecord");
-
-  stream_->Start(this);
+  audio_callback_.reset(new AudioCallback(this));
+  stream_->Start(audio_callback_.get());
 }
 
 void AudioInputController::DoClose() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.CloseTime");
-  // If we have already logged something, this does nothing.
-  // Otherwise, we haven't recieved data.
-  LogCaptureStartupResult(CAPTURE_STARTUP_NEVER_GOT_DATA);
 
-  if (state_ == CLOSED)
+  if (!stream_)
     return;
 
-  // If this is a low-latency stream, log the total duration (since DoCreate)
-  // and add it to a UMA histogram.
-  if (!low_latency_create_time_.is_null()) {
-    base::TimeDelta duration =
-        base::TimeTicks::Now() - low_latency_create_time_;
-    UMA_HISTOGRAM_LONG_TIMES("Media.InputStreamDuration", duration);
-    if (handler_) {
-      std::string log_string =
-          base::StringPrintf("AIC::DoClose: stream duration=");
-      log_string += base::Int64ToString(duration.InSeconds());
-      log_string += " seconds";
-      handler_->OnLog(this, log_string);
+  // Allow calling unconditionally and bail if we don't have a stream to close.
+  if (audio_callback_) {
+    stream_->Stop();
+
+    if (!low_latency_create_time_.is_null()) {
+      LogCaptureStartupResult(audio_callback_->received_callback()
+                                  ? CAPTURE_STARTUP_OK
+                                  : CAPTURE_STARTUP_NEVER_GOT_DATA);
+      UMA_HISTOGRAM_BOOLEAN("Media.Audio.Capture.CallbackError",
+                            audio_callback_->error_during_callback());
+      if (audio_callback_->received_callback()) {
+        // Log the total duration (since DoCreate) and update the histogram.
+        const base::TimeDelta duration =
+            base::TimeTicks::Now() - low_latency_create_time_;
+        UMA_HISTOGRAM_LONG_TIMES("Media.InputStreamDuration", duration);
+        const std::string log_string = base::StringPrintf(
+            "AIC::DoClose: stream duration=%" PRId64 " seconds",
+            duration.InSeconds());
+        handler_->OnLog(this, log_string);
+      }
     }
+
+    audio_callback_.reset();
   }
 
-  DoStopCloseAndClearStream();
+  stream_->Close();
+  stream_ = nullptr;
 
-  if (SharedMemoryAndSyncSocketMode())
-    sync_writer_->Close();
+  sync_writer_->Close();
 
   if (user_input_monitor_)
     user_input_monitor_->DisableKeyPressMonitoring();
 
 #if defined(AUDIO_POWER_MONITORING)
   // Send UMA stats if enabled.
   if (log_silence_state_)
     LogSilenceState(silence_state_);
   log_silence_state_ = false;
 #endif
 
   if (debug_writer_)
     debug_writer_->Stop();
 
-  state_ = CLOSED;
+  max_volume_ = 0.0;
+  low_latency_create_time_ = base::TimeTicks();  // Reset to null.
+  weak_ptr_factory_.InvalidateWeakPtrs();
 }
 
 void AudioInputController::DoReportError() {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  if (handler_)
-    handler_->OnError(this, STREAM_ERROR);
+  handler_->OnError(this, STREAM_ERROR);
 }
 
 void AudioInputController::DoSetVolume(double volume) {
   DCHECK(task_runner_->BelongsToCurrentThread());
   DCHECK_GE(volume, 0);
   DCHECK_LE(volume, 1.0);
 
-  if (state_ != CREATED && state_ != RECORDING)
+  if (!stream_)
     return;
 
   // Only ask for the maximum volume at first call and use cached value
   // for remaining function calls.
   if (!max_volume_) {
     max_volume_ = stream_->GetMaxVolume();
   }
 
   if (max_volume_ == 0.0) {
     DLOG(WARNING) << "Failed to access input volume control";
     return;
   }
 
   // Set the stream volume and scale to a range matched to the platform.
   stream_->SetVolume(max_volume_ * volume);
 }
 
-void AudioInputController::OnData(AudioInputStream* stream,
-                                  const AudioBus* source,
-                                  uint32_t hardware_delay_bytes,
-                                  double volume) {
-  TRACE_EVENT0("audio", "AudioInputController::OnData");
-  if (debug_writer_ && debug_writer_->WillWrite()) {
-    std::unique_ptr<AudioBus> source_copy =
-        AudioBus::Create(source->channels(), source->frames());
-    source->CopyTo(source_copy.get());
-    task_runner_->PostTask(
-        FROM_HERE,
-        base::Bind(
-            &AudioInputController::WriteInputDataForDebugging,
-            this,
-            base::Passed(&source_copy)));
-  }
-  // Now we have data, so we know for sure that startup was ok.
-  LogCaptureStartupResult(CAPTURE_STARTUP_OK);
-
-  {
-    base::AutoLock auto_lock(lock_);
-    if (state_ != RECORDING)
-      return;
-  }
-
-  bool key_pressed = false;
-  if (user_input_monitor_) {
-    size_t current_count = user_input_monitor_->GetKeyPressCount();
-    key_pressed = current_count != prev_key_down_count_;
-    prev_key_down_count_ = current_count;
-    DVLOG_IF(6, key_pressed) << "Detected keypress.";
-  }
-
-  // Use SharedMemory and SyncSocket if the client has created a SyncWriter.
-  // Used by all low-latency clients except WebSpeech.
-  if (SharedMemoryAndSyncSocketMode()) {
-    sync_writer_->Write(source, volume, key_pressed, hardware_delay_bytes);
-
-#if defined(AUDIO_POWER_MONITORING)
-    // Only do power-level measurements if DoCreate() has been called. It will
-    // ensure that logging will mainly be done for WebRTC and WebSpeech
-    // clients.
-    if (!power_measurement_is_enabled_)
-      return;
-
-    // Perform periodic audio (power) level measurements.
-    if ((base::TimeTicks::Now() - last_audio_level_log_time_).InSeconds() >
-        kPowerMonitorLogIntervalSeconds) {
-      // Calculate the average power of the signal, or the energy per sample.
-      const float average_power_dbfs = AveragePower(*source);
-
-      // Add current microphone volume to log and UMA histogram.
-      const int mic_volume_percent = static_cast<int>(100.0 * volume);
-
-      // Use event handler on the audio thread to relay a message to the ARIH
-      // in content which does the actual logging on the IO thread.
-      task_runner_->PostTask(FROM_HERE,
-                             base::Bind(&AudioInputController::DoLogAudioLevels,
-                                        this,
-                                        average_power_dbfs,
-                                        mic_volume_percent));
-
-      last_audio_level_log_time_ = base::TimeTicks::Now();
-    }
-#endif
-    return;
-  }
-
-  // TODO(henrika): Investigate if we can avoid the extra copy here.
-  // (see http://crbug.com/249316 for details). AFAIK, this scope is only
-  // active for WebSpeech clients.
-  std::unique_ptr<AudioBus> audio_data =
-      AudioBus::Create(source->channels(), source->frames());
-  source->CopyTo(audio_data.get());
-
-  // Ownership of the audio buffer will be with the callback until it is run,
-  // when ownership is passed to the callback function.
-  task_runner_->PostTask(
-      FROM_HERE,
-      base::Bind(
-          &AudioInputController::DoOnData, this, base::Passed(&audio_data)));
-}
-
-void AudioInputController::DoOnData(std::unique_ptr<AudioBus> data) {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  if (handler_)
-    handler_->OnData(this, data.get());
-}
-
 void AudioInputController::DoLogAudioLevels(float level_dbfs,
                                             int microphone_volume_percent) {
 #if defined(AUDIO_POWER_MONITORING)
   DCHECK(task_runner_->BelongsToCurrentThread());
-  if (!handler_)
+  if (!stream_)
     return;
 
   // Detect if the user has enabled hardware mute by pressing the mute
   // button in audio settings for the selected microphone.
   const bool microphone_is_muted = stream_->IsMuted();
   if (microphone_is_muted) {
     LogMicrophoneMuteResult(MICROPHONE_IS_MUTED);
     handler_->OnLog(this, "AIC::OnData: microphone is muted!");
     // Return early if microphone is muted. No need to adding logs and UMA stats
     // of audio levels if we know that the micropone is muted.
@@ skipping 13 matching lines @@
 
   UMA_HISTOGRAM_PERCENTAGE("Media.MicrophoneVolume", microphone_volume_percent);
   log_string = base::StringPrintf(
       "AIC::OnData: microphone volume=%d%%", microphone_volume_percent);
   if (microphone_volume_percent < kLowLevelMicrophoneLevelPercent)
     log_string += " <=> low microphone level!";
   handler_->OnLog(this, log_string);
 #endif
 }
 
-void AudioInputController::OnError(AudioInputStream* stream) {
-  // Handle error on the audio-manager thread.
-  task_runner_->PostTask(FROM_HERE, base::Bind(
-      &AudioInputController::DoReportError, this));
-}
-
 void AudioInputController::EnableDebugRecording(
     const base::FilePath& file_name) {
+  DCHECK(creator_task_runner_->BelongsToCurrentThread());
   task_runner_->PostTask(
       FROM_HERE, base::Bind(&AudioInputController::DoEnableDebugRecording, this,
                             file_name));
 }
 
 void AudioInputController::DisableDebugRecording() {
   DCHECK(creator_task_runner_->BelongsToCurrentThread());
   task_runner_->PostTask(
       FROM_HERE,
       base::Bind(&AudioInputController::DoDisableDebugRecording, this));
 }
 
-void AudioInputController::DoStopCloseAndClearStream() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  // Allow calling unconditionally and bail if we don't have a stream to close.
-  if (stream_ != nullptr) {
-    stream_->Stop();
-    stream_->Close();
-    stream_ = nullptr;
-  }
-
-  // The event handler should not be touched after the stream has been closed.
-  handler_ = nullptr;
-}
-
 #if defined(AUDIO_POWER_MONITORING)
 void AudioInputController::UpdateSilenceState(bool silence) {
   if (silence) {
     if (silence_state_ == SILENCE_STATE_NO_MEASUREMENT) {
       silence_state_ = SILENCE_STATE_ONLY_SILENCE;
     } else if (silence_state_ == SILENCE_STATE_ONLY_AUDIO) {
       silence_state_ = SILENCE_STATE_AUDIO_AND_SILENCE;
     } else {
       DCHECK(silence_state_ == SILENCE_STATE_ONLY_SILENCE ||
              silence_state_ == SILENCE_STATE_AUDIO_AND_SILENCE);
@@ skipping 12 matching lines @@
 
 void AudioInputController::LogSilenceState(SilenceState value) {
   UMA_HISTOGRAM_ENUMERATION("Media.AudioInputControllerSessionSilenceReport",
                             value,
                             SILENCE_STATE_MAX + 1);
 }
 #endif
 
 void AudioInputController::LogCaptureStartupResult(
     CaptureStartupResult result) {
-  // Decrement shall_report_stats and check if it was 1 before decrement,
-  // which would imply that this is the first time this method is called
-  // after initialization. To avoid underflow, we
-  // also check if should_report_stats is one before decrementing.
-  if (base::AtomicRefCountIsOne(&should_report_stats) &&
-      !base::AtomicRefCountDec(&should_report_stats)) {
-    UMA_HISTOGRAM_ENUMERATION("Media.AudioInputControllerCaptureStartupSuccess",
-                              result, CAPTURE_STARTUP_RESULT_MAX + 1);
-  }
+  UMA_HISTOGRAM_ENUMERATION("Media.AudioInputControllerCaptureStartupSuccess",
+                            result, CAPTURE_STARTUP_RESULT_MAX + 1);
 }
 
 void AudioInputController::DoEnableDebugRecording(
     const base::FilePath& file_name) {
   DCHECK(task_runner_->BelongsToCurrentThread());
   if (debug_writer_)
     debug_writer_->Start(file_name);
 }
 
 void AudioInputController::DoDisableDebugRecording() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   if (debug_writer_)
     debug_writer_->Stop();
 }
 
 void AudioInputController::WriteInputDataForDebugging(
     std::unique_ptr<AudioBus> data) {
   DCHECK(task_runner_->BelongsToCurrentThread());
   if (debug_writer_)
     debug_writer_->Write(std::move(data));
 }
 
 void AudioInputController::LogMessage(const std::string& message) {
   DCHECK(task_runner_->BelongsToCurrentThread());
   handler_->OnLog(this, message);
 }
 
+bool AudioInputController::CheckForKeyboardInput() {
+  if (!user_input_monitor_)
+    return false;
+
+  const size_t current_count = user_input_monitor_->GetKeyPressCount();
+  const bool key_pressed = current_count != prev_key_down_count_;
+  prev_key_down_count_ = current_count;
+  DVLOG_IF(6, key_pressed) << "Detected keypress.";
+
+  return key_pressed;
+}
+
+bool AudioInputController::CheckAudioPower(const AudioBus* source,
+                                           double volume,
+                                           float* average_power_dbfs,
+                                           int* mic_volume_percent) {
+#if defined(AUDIO_POWER_MONITORING)
+  // Only do power-level measurements if DoCreate() has been called. It will
+  // ensure that logging will mainly be done for WebRTC and WebSpeech
+  // clients.
+  if (!power_measurement_is_enabled_)
+    return false;
+
+  // Perform periodic audio (power) level measurements.
+  const auto now = base::TimeTicks::Now();
+  if ((now - last_audio_level_log_time_).InSeconds() <=
+      kPowerMonitorLogIntervalSeconds) {
+    return false;
+  }
+
+  *average_power_dbfs = AveragePower(*source);
+  *mic_volume_percent = static_cast<int>(100.0 * volume);
+
+  last_audio_level_log_time_ = now;
+
+  return true;
+#else
+  return false;
+#endif
+}
+
 }  // namespace media