kjellander c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00

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1.5 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
#include <queue>
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/numerics/percentile_filter.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class VCMCodecTimer {
public:
VCMCodecTimer();
// Add a new decode time to the filter.
void AddTiming(int64_t new_decode_time_ms, int64_t now_ms);
// Get the required decode time in ms. It is the 95th percentile observed
// decode time within a time window.
int64_t RequiredDecodeTimeMs() const;
private:
struct Sample {
Sample(int64_t decode_time_ms, int64_t sample_time_ms);
int64_t decode_time_ms;
int64_t sample_time_ms;
};
// The number of samples ignored so far.
int ignored_sample_count_;
// Queue with history of latest decode time values.
std::queue<Sample> history_;
// |filter_| contains the same values as |history_|, but in a data structure
// that allows efficient retrieval of the percentile value.
PercentileFilter<int64_t> filter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_