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 |