| 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 a9163ea0a611570792a3863c40220fe63ee68ec4..ed8b1f77228de3110dee5d74c9368a18b1778e13 100644
|
| --- a/webrtc/modules/audio_device/audio_device_buffer.h
|
| +++ b/webrtc/modules/audio_device/audio_device_buffer.h
|
| @@ -188,58 +188,58 @@ class AudioDeviceBuffer {
|
| // Keeps track of if playout/recording are active or not. A combination
|
| // of these states are used to determine when to start and stop the timer.
|
| // Only used on the creating thread and not used to control any media flow.
|
| - bool playing_ ACCESS_ON(main_thread_checker_);
|
| - bool recording_ ACCESS_ON(main_thread_checker_);
|
| + bool playing_ RTC_ACCESS_ON(main_thread_checker_);
|
| + bool recording_ RTC_ACCESS_ON(main_thread_checker_);
|
|
|
| // Buffer used for audio samples to be played out. Size can be changed
|
| // dynamically. The 16-bit samples are interleaved, hence the size is
|
| // proportional to the number of channels.
|
| - rtc::BufferT<int16_t> play_buffer_ ACCESS_ON(playout_thread_checker_);
|
| + rtc::BufferT<int16_t> play_buffer_ RTC_ACCESS_ON(playout_thread_checker_);
|
|
|
| // Byte buffer used for recorded audio samples. Size can be changed
|
| // dynamically.
|
| - rtc::BufferT<int16_t> rec_buffer_ ACCESS_ON(recording_thread_checker_);
|
| + rtc::BufferT<int16_t> rec_buffer_ RTC_ACCESS_ON(recording_thread_checker_);
|
|
|
| // AGC parameters.
|
| #if !defined(WEBRTC_WIN)
|
| - uint32_t current_mic_level_ ACCESS_ON(recording_thread_checker_);
|
| + uint32_t current_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
|
| #else
|
| // Windows uses a dedicated thread for volume APIs.
|
| uint32_t current_mic_level_;
|
| #endif
|
| - uint32_t new_mic_level_ ACCESS_ON(recording_thread_checker_);
|
| + uint32_t new_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
|
|
|
| // Contains true of a key-press has been detected.
|
| - bool typing_status_ ACCESS_ON(recording_thread_checker_);
|
| + bool typing_status_ RTC_ACCESS_ON(recording_thread_checker_);
|
|
|
| // Delay values used by the AEC.
|
| - int play_delay_ms_ ACCESS_ON(recording_thread_checker_);
|
| - int rec_delay_ms_ ACCESS_ON(recording_thread_checker_);
|
| + int play_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
|
| + int rec_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
|
|
|
| // Contains a clock-drift measurement.
|
| - int clock_drift_ ACCESS_ON(recording_thread_checker_);
|
| + int clock_drift_ RTC_ACCESS_ON(recording_thread_checker_);
|
|
|
| // Counts number of times LogStats() has been called.
|
| - size_t num_stat_reports_ ACCESS_ON(task_queue_);
|
| + size_t num_stat_reports_ RTC_ACCESS_ON(task_queue_);
|
|
|
| // Time stamp of last timer task (drives logging).
|
| - int64_t last_timer_task_time_ ACCESS_ON(task_queue_);
|
| + int64_t last_timer_task_time_ RTC_ACCESS_ON(task_queue_);
|
|
|
| // Counts number of audio callbacks modulo 50 to create a signal when
|
| // a new storage of audio stats shall be done.
|
| - int16_t rec_stat_count_ ACCESS_ON(recording_thread_checker_);
|
| - int16_t play_stat_count_ ACCESS_ON(playout_thread_checker_);
|
| + int16_t rec_stat_count_ RTC_ACCESS_ON(recording_thread_checker_);
|
| + int16_t play_stat_count_ RTC_ACCESS_ON(playout_thread_checker_);
|
|
|
| // Time stamps of when playout and recording starts.
|
| - int64_t play_start_time_ ACCESS_ON(main_thread_checker_);
|
| - int64_t rec_start_time_ ACCESS_ON(main_thread_checker_);
|
| + int64_t play_start_time_ RTC_ACCESS_ON(main_thread_checker_);
|
| + int64_t rec_start_time_ RTC_ACCESS_ON(main_thread_checker_);
|
|
|
| // Contains counters for playout and recording statistics.
|
| - Stats stats_ GUARDED_BY(lock_);
|
| + Stats stats_ RTC_GUARDED_BY(lock_);
|
|
|
| // Stores current stats at each timer task. Used to calculate differences
|
| // between two successive timer events.
|
| - Stats last_stats_ ACCESS_ON(task_queue_);
|
| + Stats last_stats_ RTC_ACCESS_ON(task_queue_);
|
|
|
| // Set to true at construction and modified to false as soon as one audio-
|
| // level estimate larger than zero is detected.
|
| @@ -249,12 +249,12 @@ class AudioDeviceBuffer {
|
| // StartPeriodicLogging() and set to false by StopPeriodicLogging().
|
| // Setting this member to false prevents (possiby invalid) log messages from
|
| // being printed in the LogStats() task.
|
| - bool log_stats_ ACCESS_ON(task_queue_);
|
| + bool log_stats_ RTC_ACCESS_ON(task_queue_);
|
|
|
| // Should *never* be defined in production builds. Only used for testing.
|
| // When defined, the output signal will be replaced by a sinus tone at 440Hz.
|
| #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
|
| - double phase_ ACCESS_ON(playout_thread_checker_);
|
| + double phase_ RTC_ACCESS_ON(playout_thread_checker_);
|
| #endif
|
| };
|
|
|
|
|