Packet buffer for the new jitter buffer.
BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
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webrtc/modules/video_coding/packet_buffer.h
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81
webrtc/modules/video_coding/packet_buffer.h
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/*
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* Copyright (c) 2016 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_PACKET_BUFFER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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#include <vector>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/modules/video_coding/packet.h"
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namespace webrtc {
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namespace video_coding {
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class FrameObject;
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class RtpFrameObject;
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class OnCompleteFrameCallback {
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public:
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virtual ~OnCompleteFrameCallback() {}
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virtual void OnCompleteFrame(std::unique_ptr<FrameObject> frame) = 0;
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};
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class PacketBuffer {
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public:
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// Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
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PacketBuffer(size_t start_buffer_size,
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size_t max_buffer_size,
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OnCompleteFrameCallback* frame_callback);
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bool InsertPacket(const VCMPacket& packet);
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void ClearTo(uint16_t seq_num);
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void Flush();
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private:
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friend RtpFrameObject;
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// Since we want the packet buffer to be as packet type agnostic
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// as possible we extract only the information needed in order
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// to determin whether a sequence of packets is continuous or not.
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struct ContinuityInfo {
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uint16_t seq_num = 0;
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bool frame_begin = false;
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bool frame_end = false;
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bool used = false;
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bool continuous = false;
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};
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bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void FindCompleteFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
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bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
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void ReturnFrame(RtpFrameObject* frame);
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rtc::CriticalSection crit_;
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// Buffer size_ and max_size_ must always be a power of two.
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size_t size_ GUARDED_BY(crit_);
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const size_t max_size_;
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uint16_t last_seq_num_ GUARDED_BY(crit_);
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uint16_t first_seq_num_ GUARDED_BY(crit_);
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bool initialized_ GUARDED_BY(crit_);
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std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);
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std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);
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OnCompleteFrameCallback* const frame_callback_;
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};
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} // namespace video_coding
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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