Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/video_coding/codec_timer.h
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modules/video_coding/codec_timer.h
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
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#include <queue>
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/rtc_base/numerics/percentile_filter.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class VCMCodecTimer {
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public:
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VCMCodecTimer();
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// Add a new decode time to the filter.
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void AddTiming(int64_t new_decode_time_ms, int64_t now_ms);
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// Get the required decode time in ms. It is the 95th percentile observed
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// decode time within a time window.
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int64_t RequiredDecodeTimeMs() const;
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private:
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struct Sample {
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Sample(int64_t decode_time_ms, int64_t sample_time_ms);
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int64_t decode_time_ms;
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int64_t sample_time_ms;
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};
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// The number of samples ignored so far.
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int ignored_sample_count_;
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// Queue with history of latest decode time values.
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std::queue<Sample> history_;
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// |filter_| contains the same values as |history_|, but in a data structure
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// that allows efficient retrieval of the percentile value.
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PercentileFilter<int64_t> filter_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
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