From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../libwebrtc/modules/video_coding/packet_buffer.h | 134 +++++++++++++++++++++ 1 file changed, 134 insertions(+) create mode 100644 third_party/libwebrtc/modules/video_coding/packet_buffer.h (limited to 'third_party/libwebrtc/modules/video_coding/packet_buffer.h') diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer.h b/third_party/libwebrtc/modules/video_coding/packet_buffer.h new file mode 100644 index 0000000000..47b2ffe199 --- /dev/null +++ b/third_party/libwebrtc/modules/video_coding/packet_buffer.h @@ -0,0 +1,134 @@ +/* + * 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); + + // Clear all packets older than |seq_num|. Returns the number of packets + // cleared. + uint32_t ClearTo(uint16_t seq_num); + void Clear(); + + void ForceSpsPpsIdrIsH264Keyframe(); + void ResetSpsPpsIdrIsH264Keyframe(); + + 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_; + + std::set> received_padding_; + + // 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_ -- cgit v1.2.3