/*
 *  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 WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
#define WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_

#include <vector>

#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/packet.h"
#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
#include "webrtc/modules/video_coding/sequence_number_util.h"

namespace webrtc {
namespace video_coding {

class FrameObject;
class RtpFrameObject;

class OnCompleteFrameCallback {
 public:
  virtual ~OnCompleteFrameCallback() {}
  virtual void OnCompleteFrame(std::unique_ptr<FrameObject> frame) = 0;
};

class PacketBuffer {
 public:
  // 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,
               OnCompleteFrameCallback* frame_callback);

  bool InsertPacket(const VCMPacket& packet);
  void ClearTo(uint16_t seq_num);
  void Clear();

 private:
  friend RtpFrameObject;
  // Since we want the packet buffer to be as packet type agnostic
  // as possible we extract only the information needed in order
  // to determine whether a sequence of packets is continuous or not.
  struct ContinuityInfo {
    // The sequence number of the packet.
    uint16_t seq_num = 0;

    // If this is the first packet of the frame.
    bool frame_begin = false;

    // If this is the last packet of the frame.
    bool frame_end = false;

    // If this slot is currently used.
    bool used = false;

    // If all its previous packets have been inserted into the packet buffer.
    bool continuous = false;

    // If this packet has been used to create a frame already.
    bool frame_created = false;
  };

  // Tries to expand the buffer.
  bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);

  // Test if all previous packets has arrived for the given sequence number.
  bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);

  // Test if all packets of a frame has arrived, and if so, creates a frame.
  // May create multiple frames per invocation.
  void FindFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);

  // Copy the bitstream for |frame| to |destination|.
  bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);

  // Get the packet with sequence number |seq_num|.
  VCMPacket* GetPacket(uint16_t seq_num);

  // Mark all slots used by |frame| as not used.
  void ReturnFrame(RtpFrameObject* frame);

  rtc::CriticalSection crit_;

  // Buffer size_ and max_size_ must always be a power of two.
  size_t size_ GUARDED_BY(crit_);
  const size_t max_size_;

  // The fist sequence number currently in the buffer.
  uint16_t first_seq_num_ GUARDED_BY(crit_);

  // The last sequence number currently in the buffer.
  uint16_t last_seq_num_ GUARDED_BY(crit_);

  // If the packet buffer has received its first packet.
  bool first_packet_received_ GUARDED_BY(crit_);

  // Buffer that holds the inserted packets.
  std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);

  // Buffer that holds the information about which slot that is currently in use
  // and information needed to determine the continuity between packets.
  std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);

  // Frames that have received all their packets are handed off to the
  // |reference_finder_| which finds the dependencies between the frames.
  RtpFrameReferenceFinder reference_finder_;
};

}  // namespace video_coding
}  // namespace webrtc

#endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_