modules/remote_bitrate_estimator/remote_bitrate_estimator_multi_stream.h

/*
 *  Copyright (c) 2012 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.
 */

// This class estimates the incoming available bandwidth.

#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_IMPL_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_IMPL_H_

#include <map>

#include "modules/remote_bitrate_estimator/bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
#include "system_wrappers/interface/constructor_magic.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "typedefs.h"

namespace webrtc {

class RemoteBitrateEstimatorMultiStream : public RemoteBitrateEstimator {
 public:
  RemoteBitrateEstimatorMultiStream(RemoteBitrateObserver* observer,
                                    const OverUseDetectorOptions& options);

  ~RemoteBitrateEstimatorMultiStream() {}

  // Stores an RTCP SR (NTP, RTP timestamp) tuple for a specific SSRC to be used
  // in future RTP timestamp to NTP time conversions. As soon as any SSRC has
  // two tuples the RemoteBitrateEstimator will switch to multi-stream mode.
  void IncomingRtcp(unsigned int ssrc, uint32_t ntp_secs, uint32_t ntp_frac,
                    uint32_t rtp_timestamp);

  // Called for each incoming packet. The first SSRC will immediately be used
  // for overuse detection. Subsequent SSRCs will only be used when at least
  // two RTCP SR reports with the same SSRC have been received.
  void IncomingPacket(unsigned int ssrc,
                      int packet_size,
                      int64_t arrival_time,
                      uint32_t rtp_timestamp);

  // Triggers a new estimate calculation.
  void UpdateEstimate(unsigned int ssrc, int64_t time_now);

  // Set the current round-trip time experienced by the streams going into this
  // estimator.
  void SetRtt(unsigned int rtt);

  // Removes all data for |ssrc|.
  void RemoveStream(unsigned int ssrc);

  // Returns true if a valid estimate exists and sets |bitrate_bps| to the
  // estimated bitrate in bits per second.
  bool LatestEstimate(unsigned int ssrc, unsigned int* bitrate_bps) const;

 private:
  typedef std::map<unsigned int, synchronization::RtcpList> StreamMap;

  RemoteRateControl remote_rate_;
  OveruseDetector overuse_detector_;
  BitRateStats incoming_bitrate_;
  RemoteBitrateObserver* observer_;
  StreamMap streams_;
  scoped_ptr<CriticalSectionWrapper> crit_sect_;
  unsigned int initial_ssrc_;
  bool multi_stream_;

  DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorMultiStream);
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_IMPL_H_