129 lines
4.0 KiB
C++
129 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 MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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#define MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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#include <memory>
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#include <queue>
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#include <set>
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#include <vector>
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#include "absl/base/attributes.h"
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#include "api/rtp_packet_info.h"
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#include "api/units/timestamp.h"
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#include "api/video/encoded_image.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/numerics/sequence_number_util.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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namespace video_coding {
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class PacketBuffer {
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public:
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struct Packet {
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Packet() = default;
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Packet(const RtpPacketReceived& rtp_packet,
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const RTPVideoHeader& video_header);
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Packet(const Packet&) = delete;
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Packet(Packet&&) = delete;
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Packet& operator=(const Packet&) = delete;
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Packet& operator=(Packet&&) = delete;
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~Packet() = default;
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VideoCodecType codec() const { return video_header.codec; }
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int width() const { return video_header.width; }
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int height() const { return video_header.height; }
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bool is_first_packet_in_frame() const {
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return video_header.is_first_packet_in_frame;
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}
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bool is_last_packet_in_frame() const {
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return video_header.is_last_packet_in_frame;
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}
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// If all its previous packets have been inserted into the packet buffer.
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// Set and used internally by the PacketBuffer.
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bool continuous = false;
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bool marker_bit = false;
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uint8_t payload_type = 0;
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uint16_t seq_num = 0;
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uint32_t timestamp = 0;
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int times_nacked = -1;
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rtc::CopyOnWriteBuffer video_payload;
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RTPVideoHeader video_header;
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};
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struct InsertResult {
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std::vector<std::unique_ptr<Packet>> packets;
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// Indicates if the packet buffer was cleared, which means that a key
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// frame request should be sent.
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bool buffer_cleared = false;
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};
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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, size_t max_buffer_size);
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~PacketBuffer();
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ABSL_MUST_USE_RESULT InsertResult
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InsertPacket(std::unique_ptr<Packet> packet);
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ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num);
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void ClearTo(uint16_t seq_num);
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void Clear();
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void ForceSpsPpsIdrIsH264Keyframe();
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private:
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void ClearInternal();
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// Tries to expand the buffer.
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bool ExpandBufferSize();
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// Test if all previous packets has arrived for the given sequence number.
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bool PotentialNewFrame(uint16_t seq_num) const;
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// Test if all packets of a frame has arrived, and if so, returns packets to
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// create frames.
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std::vector<std::unique_ptr<Packet>> FindFrames(uint16_t seq_num);
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void UpdateMissingPackets(uint16_t seq_num);
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// buffer_.size() and max_size_ must always be a power of two.
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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_;
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// If the packet buffer has received its first packet.
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bool first_packet_received_;
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// If the buffer is cleared to |first_seq_num_|.
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bool is_cleared_to_first_seq_num_;
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// Buffer that holds the the inserted packets and information needed to
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// determine continuity between them.
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std::vector<std::unique_ptr<Packet>> buffer_;
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absl::optional<uint16_t> newest_inserted_seq_num_;
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std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_;
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// Indicates if we should require SPS, PPS, and IDR for a particular
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// RTP timestamp to treat the corresponding frame as a keyframe.
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bool sps_pps_idr_is_h264_keyframe_;
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};
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} // namespace video_coding
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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