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Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )

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Reason for revert:
Seems to break things upstream.

Original issue's description:
> Adds data logging in native AudioDeviceBuffer class.
>
> Goal is to provide periodic logging of most essential audio parameters
> for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
>
> BUG=NONE
>
> Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> Cr-Commit-Position: refs/heads/master@{#13440}

TBR=stefan@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE

Review-Url: https://codereview.webrtc.org/2139233002
Cr-Commit-Position: refs/heads/master@{#13441}
This commit is contained in:
sprang
2016-07-12 03:08:45 -07:00
committed by Commit bot
parent 348e411dd2
commit 025aa94ccb
2 changed files with 38 additions and 193 deletions
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70
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -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"
@ -66,36 +63,11 @@ class AudioDeviceBuffer {
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 @@ class AudioDeviceBuffer {
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