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 {
+class AudioInputController::AudioCallback

[o1ka, 2017/01/16 15:04:52]

Could you add a comment explaining the reasons why functionality is split
between AIC and AIC::Callback the way it is?

[tommi (sloooow) - chröme, 2017/01/16 17:02:32]

Done.

+    : 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();

[o1ka, 2017/01/16 10:23:31]

maxmorin@: here is another dependency on browser process we missed.

+    controller_->sync_writer_->Write(source, volume, key_pressed,

[o1ka, 2017/01/16 10:23:31]

I feel like relationships between AIC and AIC::Callback are too complicated now.
What if AIC::Callback will own |sync_writer|?
And what if all the "log audio levels" logic will be an AIC method? 
And same with PerformOptionalDebugRecording(): what if it will be an AIC method?
And then we provide AIC::ReportError() and AIC::Callback() does not need
|weak_controller_|.

[tommi (sloooow) - chröme, 2017/01/16 12:34:30]

Before, they were all in the same class. With this change, I find that it's
clearer that the lifetime of the functionality that goes together with the
Callback class, is shorter than what goes for the AIC.
I considered that but didn't move ownership to here because of the variable in
the parent class.  However, considering that there's no "StopRecording" method,
only Close() and now that I look at the implementation here:

https://cs.chromium.org/chromium/src/content/browser/renderer_host/media/audi...

It looks like the sync writer ownership can in fact be moved to this class.  In
practice it doesn't make a huge difference though because sync_writer_ is a
const variable in the controller and Close() will be done at the same time and
on the same thread.
The PostTask call must be done in the Callback class because we're on the hw
callback thread and must use the weak pointer.  The weak pointer is also
constructed on the audio manager's thread, so creating a new weak pointer inside
of CheckAudioPower, is not possible.
The AIC would then just need its own weak_controller_ and that would be
confusing.  The key thing is that adding a reference count to the AIC from the
hw callback thread, is something we need to avoid.  Since the hw callback thread
belongs to the Callback thread, I prefer that all PostTask calls from there, be
done within that class and use the same weak_controller_ variable.

[o1ka, 2017/01/16 15:04:52]

Agree that moving ownership does not make sense. And I like that we use
aggregation instead of inheritance here. But I don't like how we split
controller logic, including private member access, between two classes - I find
it not very readable.
Why CheckForKeyboardInput() logic and |prev_key_down_count_| is in controller?
Same for |power_measurement_is_enabled_| - why can't it be a member of
AIC::Callback()? The edge between objects looks a bit random and too complicated
to me.
However, it may be a personal preference, and we can refactor it later if we
need ;) so - up to you.

[tommi (sloooow) - chröme, 2017/01/16 17:02:32]

I agree that it's not exactly elegant.  I added a comment that explains why we
do this here and that it can be improved as far as readability goes.

+                                     hardware_delay_bytes);
+    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();

[o1ka, 2017/01/16 15:04:52]

Shouldn't we init it right before starting the stream at l.434?
Otherwise the first callback will count all key presses since the controller was
created, and I believe we are interested in key presses happened while the input
data buffer was captured.

[tommi (sloooow) - chröme, 2017/01/16 17:02:32]

Good point, moved.  In general I think this isn't very clear in the class.  E.g.
we measure "duration" from create to close, and not capture to stop capture.

   }
+
+  // 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;
-  }
+  audio_callback_.reset(new AudioCallback(this));
 
-  if (handler_)
-    handler_->OnLog(this, "AIC::DoRecord");
+  handler_->OnLog(this, "AIC::DoRecord");
 
-  stream_->Start(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();

[o1ka, 2017/01/16 15:04:52]

We can do it first thing after l.442 to avoid posting to this thread while
DoClose() is being executed: anyways all the pointers will be invalidated by the
moment we exit it.

[o1ka, 2017/01/16 15:08:51]

Ah, disregard this, I forgot that post does not check the pointer.

[tommi (sloooow) - chröme, 2017/01/16 17:02:32]

Acknowledged.

 }
 
 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