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/utility/frame_dropper.h
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modules/video_coding/utility/frame_dropper.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_UTILITY_FRAME_DROPPER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
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#include <cstddef>
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#include "webrtc/rtc_base/numerics/exp_filter.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// The Frame Dropper implements a variant of the leaky bucket algorithm
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// for keeping track of when to drop frames to avoid bit rate
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// over use when the encoder can't keep its bit rate.
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class FrameDropper {
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public:
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FrameDropper();
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explicit FrameDropper(float max_time_drops);
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virtual ~FrameDropper() {}
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// Resets the FrameDropper to its initial state.
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// This means that the frameRateWeight is set to its
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// default value as well.
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virtual void Reset();
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virtual void Enable(bool enable);
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// Answers the question if it's time to drop a frame
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// if we want to reach a given frame rate. Must be
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// called for every frame.
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//
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// Return value : True if we should drop the current frame
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virtual bool DropFrame();
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// Updates the FrameDropper with the size of the latest encoded
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// frame. The FrameDropper calculates a new drop ratio (can be
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// seen as the probability to drop a frame) and updates its
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// internal statistics.
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//
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// Input:
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// - frameSizeBytes : The size of the latest frame
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// returned from the encoder.
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// - deltaFrame : True if the encoder returned
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// a key frame.
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virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
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virtual void Leak(uint32_t inputFrameRate);
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// Sets the target bit rate and the frame rate produced by
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// the camera.
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//
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// Input:
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// - bitRate : The target bit rate
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virtual void SetRates(float bitRate, float incoming_frame_rate);
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private:
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void UpdateRatio();
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void CapAccumulator();
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rtc::ExpFilter key_frame_ratio_;
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rtc::ExpFilter delta_frame_size_avg_kbits_;
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// Key frames and large delta frames are not immediately accumulated in the
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// bucket since they can immediately overflow the bucket leading to large
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// drops on the following packets that may be much smaller. Instead these
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// large frames are accumulated over several frames when the bucket leaks.
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// |large_frame_accumulation_spread_| represents the number of frames over
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// which a large frame is accumulated.
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float large_frame_accumulation_spread_;
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// |large_frame_accumulation_count_| represents the number of frames left
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// to finish accumulating a large frame.
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int large_frame_accumulation_count_;
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// |large_frame_accumulation_chunk_size_| represents the size of a single
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// chunk for large frame accumulation.
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float large_frame_accumulation_chunk_size_;
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float accumulator_;
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float accumulator_max_;
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float target_bitrate_;
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bool drop_next_;
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rtc::ExpFilter drop_ratio_;
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int drop_count_;
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float incoming_frame_rate_;
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bool was_below_max_;
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bool enabled_;
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const float max_drop_duration_secs_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
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