Adds thread safety annotations to the AudioDeviceBuffer class

webrtc/modules/audio_device/audio_device_buffer.h

Index: webrtc/modules/audio_device/audio_device_buffer.h
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index 7e9f3e3eec30105a5971727002fe9b8bb39d87eb..864fb95d0a44e8e58e3dc246ff7b900b0e9cd11d 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -13,7 +13,9 @@
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/race_checker.h"
 #include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
@@ -47,27 +49,27 @@ class AudioDeviceBuffer {
   void StopPlayout();
   void StopRecording();
 
-  int32_t SetRecordingSampleRate(uint32_t fsHz);
-  int32_t SetPlayoutSampleRate(uint32_t fsHz);
+  int32_t SetRecordingSampleRate(uint32_t fsHz) LOCKS_EXCLUDED(lock_);
+  int32_t SetPlayoutSampleRate(uint32_t fsHz) LOCKS_EXCLUDED(lock_);
   int32_t RecordingSampleRate() const;
   int32_t PlayoutSampleRate() const;
 
-  int32_t SetRecordingChannels(size_t channels);
-  int32_t SetPlayoutChannels(size_t channels);
+  int32_t SetRecordingChannels(size_t channels) LOCKS_EXCLUDED(lock_);
+  int32_t SetPlayoutChannels(size_t channels) LOCKS_EXCLUDED(lock_);
   size_t RecordingChannels() const;
   size_t PlayoutChannels() const;
   int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
   int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
 
   virtual int32_t SetRecordedBuffer(const void* audio_buffer,
-                                    size_t num_samples);
+                                    size_t num_samples) LOCKS_EXCLUDED(lock_);
   int32_t SetCurrentMicLevel(uint32_t level);
   virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
-  virtual int32_t DeliverRecordedData();
+  virtual int32_t DeliverRecordedData() LOCKS_EXCLUDED(lock_);
   uint32_t NewMicLevel() const;
 
-  virtual int32_t RequestPlayoutData(size_t num_samples);
-  virtual int32_t GetPlayoutData(void* audio_buffer);
+  virtual int32_t RequestPlayoutData(size_t num_samples) LOCKS_EXCLUDED(lock_);
+  virtual int32_t GetPlayoutData(void* audio_buffer) LOCKS_EXCLUDED(lock_);
 
   // TODO(henrika): these methods should not be used and does not contain any
   // valid implementation. Investigate the possibility to either remove them
@@ -107,14 +109,18 @@ class AudioDeviceBuffer {
   // creates this object.
   rtc::ThreadChecker thread_checker_;

[kwiberg-webrtc, 2016/11/01 15:53:50]

No member variable is annotated as being protected by thread_checker_. Is it not
used?

[henrika_webrtc, 2016/11/02 10:29:18]

As we discussed. Will try to fix that.

 
-  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
-  // and it must outlive this object.
-  AudioTransport* audio_transport_cb_;
+  // Verifies that access to some members are serialized.
+  rtc::RaceChecker race_checker_;
 
-  // TODO(henrika): given usage of thread checker, it should be possible to
-  // remove all locks in this class.
   rtc::CriticalSection lock_;
-  rtc::CriticalSection lock_cb_;
+
+  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+  // and it must outlive this object. It is not possible to change this member
+  // while any media is active. It is possible to start media without calling
+  // RegisterAudioCallback() but that will lead to ignored audio callbacks in
+  // both directions where native audio will be acive but no audio samples will
+  // be transported.
+  AudioTransport* audio_transport_cb_ GUARDED_BY(race_checker_);
 
   // Task queue used to invoke LogStats() periodically. Tasks are executed on a
   // worker thread but it does not necessarily have to be the same thread for
@@ -128,24 +134,20 @@ class AudioDeviceBuffer {
   bool recording_;

[kwiberg-webrtc, 2016/11/01 15:53:50]

These are not annotated. It's usually good to make sure that *every* member
variable (apart from the mutex(es)...) are annotated, or have a comment
explaining why they don't need to be protected.

[henrika_webrtc, 2016/11/02 10:29:18]

Acknowledged.

 
   // Sample rate in Hertz.
-  uint32_t rec_sample_rate_;
-  uint32_t play_sample_rate_;
+  uint32_t rec_sample_rate_ GUARDED_BY(lock_);
+  uint32_t play_sample_rate_ GUARDED_BY(lock_);
 
   // Number of audio channels.
-  size_t rec_channels_;
-  size_t play_channels_;
-
-  // Number of bytes per audio sample (2 or 4).
-  size_t rec_bytes_per_sample_;
-  size_t play_bytes_per_sample_;
+  size_t rec_channels_ GUARDED_BY(lock_);
+  size_t play_channels_ GUARDED_BY(lock_);
 
   // Byte buffer used for recorded audio samples. Size can be changed
   // dynamically.
-  rtc::Buffer rec_buffer_;
+  rtc::Buffer rec_buffer_ GUARDED_BY(race_checker_);
 
   // Buffer used for audio samples to be played out. Size can be changed
   // dynamically.
-  rtc::Buffer play_buffer_;
+  rtc::Buffer play_buffer_ GUARDED_BY(race_checker_);
 
   // AGC parameters.
   uint32_t current_mic_level_;