refactor AudioInputDevice to remove race condition.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright (c) 2011 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <vector>
 
 #include "base/basictypes.h"
 #include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
+#include "base/threading/thread.h"
 #include "base/time.h"
 #include "content/renderer/media/audio_device.h"
 #include "content/renderer/media/audio_input_device.h"
+#include "content/renderer/media/audio_input_device_event_handler.h"
 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h"
 
 // A WebRtcAudioDeviceImpl instance implements the abstract interface
 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc::
 // VoiceEngine) to register this class as an external AudioDeviceModule.
 // The user can then call WebRtcAudioDeviceImpl::StartPlayout() and
 // WebRtcAudioDeviceImpl::StartRecording() from the render process
 // to initiate and start audio rendering and capturing in the browser process.
 // IPC is utilized to set up the media streams.
 //
@@ skipping 44 matching lines @@
 //  |   .-------------------------------.             .------------------.
 //  |
 //  |                             .---------------------.
 //  .-> RecordedDataIsAvailable ->| webrtc::VoiceEngine | => (encode+transmit)
 //                                .---------------------.
 //
 //  (*) Using SyncSocket for inter-process synchronization with low latency.
 //      The actual data is transferred via SharedMemory. IPC is not involved
 //      in the actual media transfer.
 //
 //  This class must be created on the main render thread since it creates

[henrika_dont_use, 2011/08/07 16:52:27]

Would be great with some extended comments about the new ADM thread (only used
on input side so far).

[wjia(left Chromium), 2011/08/09 01:40:36]

Done.

 //  AudioDevice and AudioInputDevice objects and they both require a valid
 //  RenderThread::current() pointer.
 //
 class WebRtcAudioDeviceImpl
     : public webrtc::AudioDeviceModule,
       public AudioDevice::RenderCallback,
-      public AudioInputDevice::CaptureCallback {
+      public AudioInputDevice::CaptureCallback,
+      public AudioInputDeviceEventHandler {

[henrika_dont_use, 2011/08/07 16:52:27]

Could have been defines using same style as AudioInputDevice::CaptureCallback,
i.e, not in a separate file. You choose not to because...?

[wjia(left Chromium), 2011/08/09 01:40:36]

see Nested Classes in http://www.chromium.org/developers/coding-style

  public:
   WebRtcAudioDeviceImpl(size_t input_buffer_size,
                         size_t output_buffer_size,
                         int input_channels,
                         int output_channels,
                         double input_sample_rate,
                         double output_sample_rate);
   virtual ~WebRtcAudioDeviceImpl();
 
   // AudioDevice::RenderCallback implementation.
   virtual void Render(const std::vector<float*>& audio_data,
                       size_t number_of_frames,
-                      size_t audio_delay_milliseconds);
+                      size_t audio_delay_milliseconds) OVERRIDE;

[henrika_dont_use, 2011/08/07 16:52:27]

General question; what is the benefit of using this macro?

[wjia(left Chromium), 2011/08/09 01:40:36]

The compiler can detect inconsistent change when the function is changed in base
class.

 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
                        size_t number_of_frames,
-                       size_t audio_delay_milliseconds);
+                       size_t audio_delay_milliseconds) OVERRIDE;
+
+  // AudioInputDeviceEventHandler implementation.
+  virtual void OnRecordingStarted() OVERRIDE;
+  virtual void OnRecordingStopped() OVERRIDE;
 
   // webrtc::Module implementation.
   virtual int32_t Version(char* version,
                           uint32_t& remaining_buffer_in_bytes,
-                          uint32_t& position) const;
-  virtual int32_t ChangeUniqueId(const int32_t id);
-  virtual int32_t TimeUntilNextProcess();
-  virtual int32_t Process();
+                          uint32_t& position) const OVERRIDE;
+  virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
+  virtual int32_t TimeUntilNextProcess() OVERRIDE;
+  virtual int32_t Process() OVERRIDE;
 
   // webrtc::AudioDeviceModule implementation.
-  virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const;
-  virtual ErrorCode LastError() const;
+  virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE;
+  virtual ErrorCode LastError() const OVERRIDE;
 
   virtual int32_t RegisterEventObserver(
-      webrtc::AudioDeviceObserver* event_callback);
-  virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback);
+      webrtc::AudioDeviceObserver* event_callback) OVERRIDE;
+  virtual int32_t RegisterAudioCallback(
+      webrtc::AudioTransport* audio_callback) OVERRIDE;
 
-  virtual int32_t Init();
-  virtual int32_t Terminate();
-  virtual bool Initialized() const;
+  virtual int32_t Init() OVERRIDE;
+  virtual int32_t Terminate() OVERRIDE;
+  virtual bool Initialized() const OVERRIDE;
 
-  virtual int16_t PlayoutDevices();
-  virtual int16_t RecordingDevices();
-  virtual int32_t PlayoutDeviceName(uint16_t index,
-                                    char name[webrtc::kAdmMaxDeviceNameSize],
-                                    char guid[webrtc::kAdmMaxGuidSize]);
-  virtual int32_t RecordingDeviceName(uint16_t index,
-                                      char name[webrtc::kAdmMaxDeviceNameSize],
-                                      char guid[webrtc::kAdmMaxGuidSize]);
+  virtual int16_t PlayoutDevices() OVERRIDE;
+  virtual int16_t RecordingDevices() OVERRIDE;
+  virtual int32_t PlayoutDeviceName(
+      uint16_t index,
+      char name[webrtc::kAdmMaxDeviceNameSize],
+      char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
+  virtual int32_t RecordingDeviceName(
+      uint16_t index,
+      char name[webrtc::kAdmMaxDeviceNameSize],
+      char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
 
-  virtual int32_t SetPlayoutDevice(uint16_t index);
-  virtual int32_t SetPlayoutDevice(WindowsDeviceType device);
-  virtual int32_t SetRecordingDevice(uint16_t index);
-  virtual int32_t SetRecordingDevice(WindowsDeviceType device);
+  virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
+  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) OVERRIDE;
+  virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
+  virtual int32_t SetRecordingDevice(WindowsDeviceType device) OVERRIDE;
 
-  virtual int32_t PlayoutIsAvailable(bool* available);
-  virtual int32_t InitPlayout();
-  virtual bool PlayoutIsInitialized() const;
-  virtual int32_t RecordingIsAvailable(bool* available);
-  virtual int32_t InitRecording();
-  virtual bool RecordingIsInitialized() const;
+  virtual int32_t PlayoutIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t InitPlayout() OVERRIDE;
+  virtual bool PlayoutIsInitialized() const OVERRIDE;
+  virtual int32_t RecordingIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t InitRecording() OVERRIDE;
+  virtual bool RecordingIsInitialized() const OVERRIDE;
 
-  virtual int32_t StartPlayout();
-  virtual int32_t StopPlayout();
-  virtual bool Playing() const;
-  virtual int32_t StartRecording();
-  virtual int32_t StopRecording();
-  virtual bool Recording() const;
+  virtual int32_t StartPlayout() OVERRIDE;
+  virtual int32_t StopPlayout() OVERRIDE;
+  virtual bool Playing() const OVERRIDE;
+  virtual int32_t StartRecording() OVERRIDE;
+  virtual int32_t StopRecording() OVERRIDE;
+  virtual bool Recording() const OVERRIDE;
 
-  virtual int32_t SetAGC(bool enable);
-  virtual bool AGC() const;
+  virtual int32_t SetAGC(bool enable) OVERRIDE;
+  virtual bool AGC() const OVERRIDE;
 
   virtual int32_t SetWaveOutVolume(uint16_t volume_left,
-                                   uint16_t volume_right);
+                                   uint16_t volume_right) OVERRIDE;
   virtual int32_t WaveOutVolume(uint16_t* volume_left,
-                                uint16_t* volume_right) const;
+                                uint16_t* volume_right) const OVERRIDE;
 
-  virtual int32_t SpeakerIsAvailable(bool* available);
-  virtual int32_t InitSpeaker();
-  virtual bool SpeakerIsInitialized() const;
-  virtual int32_t MicrophoneIsAvailable(bool* available);
-  virtual int32_t InitMicrophone();
-  virtual bool MicrophoneIsInitialized() const;
+  virtual int32_t SpeakerIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t InitSpeaker() OVERRIDE;
+  virtual bool SpeakerIsInitialized() const OVERRIDE;
+  virtual int32_t MicrophoneIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t InitMicrophone() OVERRIDE;
+  virtual bool MicrophoneIsInitialized() const OVERRIDE;
 
-  virtual int32_t SpeakerVolumeIsAvailable(bool* available);
-  virtual int32_t SetSpeakerVolume(uint32_t volume);
-  virtual int32_t SpeakerVolume(uint32_t* volume) const;
-  virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const;
-  virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const;
-  virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const;
+  virtual int32_t SpeakerVolumeIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
+  virtual int32_t SpeakerVolume(uint32_t* volume) const OVERRIDE;
+  virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const OVERRIDE;
+  virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const OVERRIDE;
+  virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const OVERRIDE;
 
-  virtual int32_t MicrophoneVolumeIsAvailable(bool* available);
-  virtual int32_t SetMicrophoneVolume(uint32_t volume);
-  virtual int32_t MicrophoneVolume(uint32_t* volume) const;
-  virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const;
-  virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const;
-  virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const;
+  virtual int32_t MicrophoneVolumeIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
+  virtual int32_t MicrophoneVolume(uint32_t* volume) const OVERRIDE;
+  virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const OVERRIDE;
+  virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE;
+  virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const OVERRIDE;
 
-  virtual int32_t SpeakerMuteIsAvailable(bool* available);
-  virtual int32_t SetSpeakerMute(bool enable);
-  virtual int32_t SpeakerMute(bool* enabled) const;
+  virtual int32_t SpeakerMuteIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
+  virtual int32_t SpeakerMute(bool* enabled) const OVERRIDE;
 
-  virtual int32_t MicrophoneMuteIsAvailable(bool* available);
-  virtual int32_t SetMicrophoneMute(bool enable);
-  virtual int32_t MicrophoneMute(bool* enabled) const;
+  virtual int32_t MicrophoneMuteIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
+  virtual int32_t MicrophoneMute(bool* enabled) const OVERRIDE;
 
-  virtual int32_t MicrophoneBoostIsAvailable(bool* available);
-  virtual int32_t SetMicrophoneBoost(bool enable);
-  virtual int32_t MicrophoneBoost(bool* enabled) const;
+  virtual int32_t MicrophoneBoostIsAvailable(bool* available) OVERRIDE;
+  virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
+  virtual int32_t MicrophoneBoost(bool* enabled) const OVERRIDE;
 
-  virtual int32_t StereoPlayoutIsAvailable(bool* available) const;
-  virtual int32_t SetStereoPlayout(bool enable);
-  virtual int32_t StereoPlayout(bool* enabled) const;
-  virtual int32_t StereoRecordingIsAvailable(bool* available) const;
-  virtual int32_t SetStereoRecording(bool enable);
-  virtual int32_t StereoRecording(bool* enabled) const;
-  virtual int32_t SetRecordingChannel(const ChannelType channel);
-  virtual int32_t RecordingChannel(ChannelType* channel) const;
+  virtual int32_t StereoPlayoutIsAvailable(bool* available) const OVERRIDE;
+  virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
+  virtual int32_t StereoPlayout(bool* enabled) const OVERRIDE;
+  virtual int32_t StereoRecordingIsAvailable(bool* available) const OVERRIDE;
+  virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
+  virtual int32_t StereoRecording(bool* enabled) const OVERRIDE;
+  virtual int32_t SetRecordingChannel(const ChannelType channel) OVERRIDE;
+  virtual int32_t RecordingChannel(ChannelType* channel) const OVERRIDE;
 
-  virtual int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms);
-  virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const;
-  virtual int32_t PlayoutDelay(uint16_t* delay_ms) const;
-  virtual int32_t RecordingDelay(uint16_t* delay_ms) const;
+  virtual int32_t SetPlayoutBuffer(const BufferType type,
+                                   uint16_t size_ms) OVERRIDE;
+  virtual int32_t PlayoutBuffer(BufferType* type,
+                                uint16_t* size_ms) const OVERRIDE;
+  virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
+  virtual int32_t RecordingDelay(uint16_t* delay_ms) const OVERRIDE;
 
-  virtual int32_t CPULoad(uint16_t* load) const;
+  virtual int32_t CPULoad(uint16_t* load) const OVERRIDE;
 
   virtual int32_t StartRawOutputFileRecording(
-      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
-  virtual int32_t StopRawOutputFileRecording();
+      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
+  virtual int32_t StopRawOutputFileRecording() OVERRIDE;
   virtual int32_t StartRawInputFileRecording(
-      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
-  virtual int32_t StopRawInputFileRecording();
+      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
+  virtual int32_t StopRawInputFileRecording() OVERRIDE;
 
-  virtual int32_t SetRecordingSampleRate(const uint32_t samples_per_sec);
-  virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const;
-  virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec);
-  virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const;
+  virtual int32_t SetRecordingSampleRate(
+      const uint32_t samples_per_sec) OVERRIDE;
+  virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
+  virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
+  virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
-  virtual int32_t ResetAudioDevice();
-  virtual int32_t SetLoudspeakerStatus(bool enable);
-  virtual int32_t GetLoudspeakerStatus(bool* enabled) const;
-
-  // Helpers.
-  bool BufferSizeIsValid(size_t buffer_size, float sample_rate) const;
+  virtual int32_t ResetAudioDevice() OVERRIDE;
+  virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
+  virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
   // Accessors.
   size_t input_buffer_size() const { return input_buffer_size_; }
   size_t output_buffer_size() const { return output_buffer_size_; }
   int input_channels() const { return input_channels_; }
   int output_channels() const { return output_channels_; }
 
  private:
+  enum State {
+    kStarting,
+    kStarted,
+    kPaused,
+    kStopped,
+    kStopping,
+    kError,
+  };
+
+  // Helpers.
+  bool BufferSizeIsValid(size_t buffer_size, float sample_rate) const;
+
+  // Helpers running on Adm thread, corresponding to API functions.

[henrika_dont_use, 2011/08/07 16:52:27]

ADM=Audio Device Module (should be ADM, right?) But hey, I did not call this
class WebRTCAudioDeviceImpl either...
Compromise proposal, use ADM in comment, Adm in method signatures.

[wjia(left Chromium), 2011/08/09 01:40:36]

Done.

+  void StartRecordingOnAdmThread();
+  void StopRecordingOnAdmThread();
+  void OnRecordingStartedOnAdmThread();
+  void OnRecordingStoppedOnAdmThread();
+  void RegisterAudioCallbackOnAdmThread(webrtc::AudioTransport* audio_callback,
+                                        int32_t* error,
+                                        base::WaitableEvent* event);
+  void GetRecordingOnAdmThread(bool* recording,
+                               base::WaitableEvent* event) const;
+  void GetRecordingDelayOnAdmThread(uint16_t* delay_ms,
+                                    base::WaitableEvent* event) const;
+
   // Provides access to the native audio input layer in the browser process.
   scoped_refptr<AudioInputDevice> audio_input_device_;
 
   // Provides access to the native audio output layer in the browser process.
   scoped_refptr<AudioDevice> audio_output_device_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
   webrtc::AudioDeviceModule::ErrorCode last_error_;
 
   size_t input_buffer_size_;
   size_t output_buffer_size_;
   int input_channels_;
   int output_channels_;
   double input_sample_rate_;
   double output_sample_rate_;
 
   int bytes_per_sample_;
 
+  base::Thread adm_thread_;
+  scoped_refptr<base::MessageLoopProxy> adm_message_loop_;
+  base::Lock lock_;

[henrika_dont_use, 2011/08/07 16:52:27]

Comment on what it protects.

[wjia(left Chromium), 2011/08/09 01:40:36]

Done.

+  base::WaitableEvent recording_stop_event_;
+
   bool initialized_;
   bool playing_;
-  bool recording_;
+  State recording_state_;
 
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   base::TimeTicks last_process_time_;
 
   // Buffers used for temporary storage during capture/render callbacks.
   // Allocated during construction to save stack.
   scoped_array<int16> input_buffer_;
   scoped_array<int16> output_buffer_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
+DISABLE_RUNNABLE_METHOD_REFCOUNT(WebRtcAudioDeviceImpl);

[henrika_dont_use, 2011/08/07 16:52:27]

Why are these macros required now?

[wjia(left Chromium), 2011/08/09 01:40:36]

These are needed for NewRunnableMethod.

[wjia(left Chromium), 2011/08/09 01:40:36]

These are needed for NewRunnableMethod.

+DISABLE_RUNNABLE_METHOD_REFCOUNT(const WebRtcAudioDeviceImpl);
+
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_