ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
63 lines
1.7 KiB
C++
63 lines
1.7 KiB
C++
/*
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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 "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc
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{
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// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
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#define MAX_HISTORY_SIZE 10
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#define SHORT_FILTER_MS 1000
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class VCMShortMaxSample
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{
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public:
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VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
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int32_t shortMax;
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int64_t timeMs;
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};
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class VCMCodecTimer
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{
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public:
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VCMCodecTimer();
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// Updates and returns the max filtered decode time.
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int32_t StopTimer(int64_t startTimeMs, int64_t nowMs);
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// Empty the list of timers.
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void Reset();
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// Get the required decode time in ms.
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int32_t RequiredDecodeTimeMs(FrameType frameType) const;
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private:
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void UpdateMaxHistory(int32_t decodeTime, int64_t now);
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void MaxFilter(int32_t newTime, int64_t nowMs);
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void ProcessHistory(int64_t nowMs);
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int32_t _filteredMax;
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// The number of samples ignored so far.
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int32_t _ignoredSampleCount;
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int32_t _shortMax;
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VCMShortMaxSample _history[MAX_HISTORY_SIZE];
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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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