b4d31085b4
Added various timestamps to the FrameObject class which are needed to calculate the jitter delay. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/2124943002 Cr-Commit-Position: refs/heads/master@{#13434}
127 lines
4.0 KiB
C++
127 lines
4.0 KiB
C++
/*
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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/thread_annotations.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
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#include "webrtc/modules/video_coding/sequence_number_util.h"
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namespace webrtc {
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class Clock;
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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(Clock* clock,
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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 Clear();
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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 determine whether a sequence of packets is continuous or not.
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struct ContinuityInfo {
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// The sequence number of the packet.
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uint16_t seq_num = 0;
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// If this is the first packet of the frame.
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bool frame_begin = false;
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// If this is the last packet of the frame.
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bool frame_end = false;
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// If this slot is currently used.
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bool used = false;
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// If all its previous packets have been inserted into the packet buffer.
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bool continuous = false;
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// If this packet has been used to create a frame already.
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bool frame_created = false;
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};
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Clock* const clock_;
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// Tries to expand the buffer.
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bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Test if all previous packets has arrived for the given sequence number.
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bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Test if all packets of a frame has arrived, and if so, creates a frame.
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// May create multiple frames per invocation.
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void FindFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Copy the bitstream for |frame| to |destination|.
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bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
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// Get the packet with sequence number |seq_num|.
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VCMPacket* GetPacket(uint16_t seq_num);
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// Mark all slots used by |frame| as not used.
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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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// The fist sequence number currently in the buffer.
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uint16_t first_seq_num_ GUARDED_BY(crit_);
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// The last sequence number currently in the buffer.
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uint16_t last_seq_num_ GUARDED_BY(crit_);
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// If the packet buffer has received its first packet.
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bool first_packet_received_ GUARDED_BY(crit_);
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// Buffer that holds the inserted packets.
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std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);
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// Buffer that holds the information about which slot that is currently in use
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// and information needed to determine the continuity between packets.
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std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);
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// Frames that have received all their packets are handed off to the
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// |reference_finder_| which finds the dependencies between the frames.
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RtpFrameReferenceFinder reference_finder_;
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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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