packet_buffer.h - mozsearch

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/*

 *  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 <memory>

#include <queue>

#include <set>

#include <vector>

#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,

           int64_t sequence_number,

           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;

    uint16_t seq_num() const { return static_cast<uint16_t>(sequence_number); }

    // 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;

    int64_t sequence_number = 0;

    uint32_t timestamp = 0;

    int times_nacked = -1;

    rtc::CopyOnWriteBuffer video_payload;

    RTPVideoHeader video_header;

};

  struct InsertResult {

    std::vector<std::unique_ptr<Packet>> 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> 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<std::unique_ptr<Packet>> 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<std::unique_ptr<Packet>> buffer_;

  absl::optional<uint16_t> newest_inserted_seq_num_;

  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_;

  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> 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_