src/modules/rtp_rtcp/source/transmission_bucket.h

/*
 *  Copyright (c) 2011 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_RTP_RTCP_TRANSMISSION_BUCKET_H_
#define WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_

#include <vector>

#include "typedefs.h"

namespace webrtc
{
class CriticalSectionWrapper;

class TransmissionBucket {
 public:
  TransmissionBucket();
  ~TransmissionBucket();

  // Resets members to initial state.
  void Reset();

  // Adds packet to be sent.
  void Fill(const uint16_t seq_num, const uint32_t num_bytes);

  // Returns true if there is no packet to be sent.
  bool Empty();

  // Updates the number of bytes that can be sent for the next time interval.
  void UpdateBytesPerInterval(const uint32_t delta_time_in_ms,
                              const uint16_t target_bitrate_kbps);

  // Checks if next packet in line can be transmitted. Returns the sequence
  // number of the packet on success, -1 otherwise. The packet is removed from
  // the vector on success.
  int32_t GetNextPacket();

 private:
   struct Packet {
     Packet(uint16_t sequence_number, uint16_t length_in_bytes)
       : sequence_number_(sequence_number),
         length_(length_in_bytes) {
     }
     uint16_t sequence_number_;
     uint16_t length_;
   };

   CriticalSectionWrapper* critsect_;
   uint32_t accumulator_;
   int32_t bytes_rem_total_;
   int32_t bytes_rem_interval_;
   std::vector<Packet> packets_;
   bool first_;
};
}  // namespace webrtc
#endif  // WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_