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Unified Diff: webrtc/modules/audio_device/audio_device_buffer.h

Issue 2141413002: Revert of Adds data logging in native AudioDeviceBuffer class (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 months ago
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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
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