webrtc_m130/webrtc/media/base/rtpdataengine.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.
 */

#ifndef WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
#define WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_

#include <memory>
#include <string>
#include <vector>

#include "webrtc/base/timing.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/mediaengine.h"

namespace cricket {

struct DataCodec;

class RtpDataEngine : public DataEngineInterface {
 public:
  RtpDataEngine();

  virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);

  virtual const std::vector<DataCodec>& data_codecs() {
    return data_codecs_;
  }

  // Mostly for testing with a fake clock.  Ownership is passed in.
  void SetTiming(rtc::Timing* timing) {
    timing_.reset(timing);
  }

 private:
  std::vector<DataCodec> data_codecs_;
  std::unique_ptr<rtc::Timing> timing_;
};

// Keep track of sequence number and timestamp of an RTP stream.  The
// sequence number starts with a "random" value and increments.  The
// timestamp starts with a "random" value and increases monotonically
// according to the clockrate.
class RtpClock {
 public:
  RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
      : clockrate_(clockrate),
        last_seq_num_(first_seq_num),
        timestamp_offset_(timestamp_offset) {}

  // Given the current time (in number of seconds which must be
  // monotonically increasing), Return the next sequence number and
  // timestamp.
  void Tick(double now, int* seq_num, uint32_t* timestamp);

 private:
  int clockrate_;
  uint16_t last_seq_num_;
  uint32_t timestamp_offset_;
};

class RtpDataMediaChannel : public DataMediaChannel {
 public:
  // Timing* Used for the RtpClock
  explicit RtpDataMediaChannel(rtc::Timing* timing);
  // Sets Timing == NULL, so you'll need to call set_timer() before
  // using it.  This is needed by FakeMediaEngine.
  RtpDataMediaChannel();
  virtual ~RtpDataMediaChannel();

  void set_timing(rtc::Timing* timing) {
    timing_ = timing;
  }

  virtual bool SetSendParameters(const DataSendParameters& params);
  virtual bool SetRecvParameters(const DataRecvParameters& params);
  virtual bool AddSendStream(const StreamParams& sp);
  virtual bool RemoveSendStream(uint32_t ssrc);
  virtual bool AddRecvStream(const StreamParams& sp);
  virtual bool RemoveRecvStream(uint32_t ssrc);
  virtual bool SetSend(bool send) {
    sending_ = send;
    return true;
  }
  virtual bool SetReceive(bool receive) {
    receiving_ = receive;
    return true;
  }
  virtual void OnPacketReceived(rtc::Buffer* packet,
                                const rtc::PacketTime& packet_time);
  virtual void OnRtcpReceived(rtc::Buffer* packet,
                              const rtc::PacketTime& packet_time) {}
  virtual void OnReadyToSend(bool ready) {}
  virtual bool SendData(
    const SendDataParams& params,
    const rtc::Buffer& payload,
    SendDataResult* result);

 private:
  void Construct(rtc::Timing* timing);
  bool SetMaxSendBandwidth(int bps);
  bool SetSendCodecs(const std::vector<DataCodec>& codecs);
  bool SetRecvCodecs(const std::vector<DataCodec>& codecs);

  bool sending_;
  bool receiving_;
  rtc::Timing* timing_;
  std::vector<DataCodec> send_codecs_;
  std::vector<DataCodec> recv_codecs_;
  std::vector<StreamParams> send_streams_;
  std::vector<StreamParams> recv_streams_;
  std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
  std::unique_ptr<rtc::RateLimiter> send_limiter_;
};

}  // namespace cricket

#endif  // WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_