/* * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef MODULES_VIDEO_CODING_PACKET_BUFFER_H_ #define MODULES_VIDEO_CODING_PACKET_BUFFER_H_ #include #include #include #include #include "absl/base/attributes.h" #include "api/rtp_packet_info.h" #include "api/units/timestamp.h" #include "api/video/encoded_image.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_video_header.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/numerics/sequence_number_util.h" #include "rtc_base/thread_annotations.h" namespace webrtc { namespace video_coding { class PacketBuffer { public: struct Packet { Packet() = default; Packet(const RtpPacketReceived& rtp_packet, const RTPVideoHeader& video_header); Packet(const Packet&) = delete; Packet(Packet&&) = delete; Packet& operator=(const Packet&) = delete; Packet& operator=(Packet&&) = delete; ~Packet() = default; VideoCodecType codec() const { return video_header.codec; } int width() const { return video_header.width; } int height() const { return video_header.height; } bool is_first_packet_in_frame() const { return video_header.is_first_packet_in_frame; } bool is_last_packet_in_frame() const { return video_header.is_last_packet_in_frame; } // If all its previous packets have been inserted into the packet buffer. // Set and used internally by the PacketBuffer. bool continuous = false; bool marker_bit = false; uint8_t payload_type = 0; uint16_t seq_num = 0; uint32_t timestamp = 0; int times_nacked = -1; rtc::CopyOnWriteBuffer video_payload; RTPVideoHeader video_header; }; struct InsertResult { std::vector> packets; // Indicates if the packet buffer was cleared, which means that a key // frame request should be sent. bool buffer_cleared = false; }; // Both |start_buffer_size| and |max_buffer_size| must be a power of 2. PacketBuffer(size_t start_buffer_size, size_t max_buffer_size); ~PacketBuffer(); ABSL_MUST_USE_RESULT InsertResult InsertPacket(std::unique_ptr packet); ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num); void ClearTo(uint16_t seq_num); void Clear(); void ForceSpsPpsIdrIsH264Keyframe(); private: void ClearInternal(); // Tries to expand the buffer. bool ExpandBufferSize(); // Test if all previous packets has arrived for the given sequence number. bool PotentialNewFrame(uint16_t seq_num) const; // Test if all packets of a frame has arrived, and if so, returns packets to // create frames. std::vector> FindFrames(uint16_t seq_num); void UpdateMissingPackets(uint16_t seq_num); // buffer_.size() and max_size_ must always be a power of two. const size_t max_size_; // The fist sequence number currently in the buffer. uint16_t first_seq_num_; // If the packet buffer has received its first packet. bool first_packet_received_; // If the buffer is cleared to |first_seq_num_|. bool is_cleared_to_first_seq_num_; // Buffer that holds the the inserted packets and information needed to // determine continuity between them. std::vector> buffer_; absl::optional newest_inserted_seq_num_; std::set> missing_packets_; // Indicates if we should require SPS, PPS, and IDR for a particular // RTP timestamp to treat the corresponding frame as a keyframe. bool sps_pps_idr_is_h264_keyframe_; }; } // namespace video_coding } // namespace webrtc #endif // MODULES_VIDEO_CODING_PACKET_BUFFER_H_