| 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 ee6b2297c77e8e3a981615e52d3c0ccf0db7c767..1267e08be2cfb9696c083e95141c79ebb2917a9a 100644
|
| --- a/webrtc/modules/audio_device/audio_device_buffer.h
|
| +++ b/webrtc/modules/audio_device/audio_device_buffer.h
|
| @@ -8,12 +8,9 @@
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
|
|
| -#ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
|
| -#define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
|
| +#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
| +#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
|
|
| -#include "webrtc/base/criticalsection.h"
|
| -#include "webrtc/base/task_queue.h"
|
| -#include "webrtc/base/thread_checker.h"
|
| #include "webrtc/modules/audio_device/include/audio_device.h"
|
| #include "webrtc/system_wrappers/include/file_wrapper.h"
|
| #include "webrtc/typedefs.h"
|
| @@ -66,35 +63,10 @@
|
| int32_t SetTypingStatus(bool typingStatus);
|
|
|
| private:
|
| - // Posts the first delayed task in the task queue and starts the periodic
|
| - // timer.
|
| - void StartTimer();
|
| -
|
| - // Called periodically on the internal thread created by the TaskQueue.
|
| - void LogStats();
|
| -
|
| - // Updates counters in each play/record callback but does it on the task
|
| - // queue to ensure that they can be read by LogStats() without any locks since
|
| - // each task is serialized by the task queue.
|
| - void UpdateRecStats(size_t num_samples);
|
| - void UpdatePlayStats(size_t num_samples);
|
| -
|
| - // Ensures that methods are called on the same thread as the thread that
|
| - // creates this object.
|
| - rtc::ThreadChecker thread_checker_;
|
| -
|
| - rtc::CriticalSection _critSect;
|
| - rtc::CriticalSection _critSectCb;
|
| + CriticalSectionWrapper& _critSect;
|
| + CriticalSectionWrapper& _critSectCb;
|
|
|
| AudioTransport* _ptrCbAudioTransport;
|
| -
|
| - // 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
|
| - // each task.
|
| - rtc::TaskQueue task_queue_;
|
| -
|
| - // Ensures that the timer is only started once.
|
| - bool timer_has_started_;
|
|
|
| uint32_t _recSampleRate;
|
| uint32_t _playSampleRate;
|
| @@ -135,40 +107,8 @@
|
| int _recDelayMS;
|
| int _clockDrift;
|
| int high_delay_counter_;
|
| -
|
| - // Counts number of times LogStats() has been called.
|
| - size_t num_stat_reports_;
|
| -
|
| - // Total number of recording callbacks where the source provides 10ms audio
|
| - // data each time.
|
| - uint64_t rec_callbacks_;
|
| -
|
| - // Total number of recording callbacks stored at the last timer task.
|
| - uint64_t last_rec_callbacks_;
|
| -
|
| - // Total number of playback callbacks where the sink asks for 10ms audio
|
| - // data each time.
|
| - uint64_t play_callbacks_;
|
| -
|
| - // Total number of playout callbacks stored at the last timer task.
|
| - uint64_t last_play_callbacks_;
|
| -
|
| - // Total number of recorded audio samples.
|
| - uint64_t rec_samples_;
|
| -
|
| - // Total number of recorded samples stored at the previous timer task.
|
| - uint64_t last_rec_samples_;
|
| -
|
| - // Total number of played audio samples.
|
| - uint64_t play_samples_;
|
| -
|
| - // Total number of played samples stored at the previous timer task.
|
| - uint64_t last_play_samples_;
|
| -
|
| - // Time stamp of last stat report.
|
| - uint64_t last_log_stat_time_;
|
| };
|
|
|
| } // namespace webrtc
|
|
|
| -#endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
|
| +#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
|
|