call/video_rtp_sender_interface.h

/*
 *  Copyright (c) 2018 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 CALL_VIDEO_RTP_SENDER_INTERFACE_H_
#define CALL_VIDEO_RTP_SENDER_INTERFACE_H_

#include <map>
#include <vector>

#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"

namespace webrtc {
class VideoBitrateAllocation;
struct FecProtectionParams;

class VideoRtpSenderInterface : public EncodedImageCallback {
 public:
  virtual void RegisterProcessThread(ProcessThread* module_process_thread) = 0;
  virtual void DeRegisterProcessThread() = 0;

  // PayloadRouter will only route packets if being active, all packets will be
  // dropped otherwise.
  virtual void SetActive(bool active) = 0;
  // Sets the sending status of the rtp modules and appropriately sets the
  // payload router to active if any rtp modules are active.
  virtual void SetActiveModules(const std::vector<bool> active_modules) = 0;
  virtual bool IsActive() = 0;

  virtual void OnNetworkAvailability(bool network_available) = 0;
  virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
  virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;

  virtual bool FecEnabled() const = 0;

  virtual bool NackEnabled() const = 0;

  virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;

  virtual void ProtectionRequest(const FecProtectionParams* delta_params,
                                 const FecProtectionParams* key_params,
                                 uint32_t* sent_video_rate_bps,
                                 uint32_t* sent_nack_rate_bps,
                                 uint32_t* sent_fec_rate_bps) = 0;

  virtual void SetMaxRtpPacketSize(size_t max_rtp_packet_size) = 0;
  virtual void OnBitrateAllocationUpdated(
      const VideoBitrateAllocation& bitrate) = 0;
};
}  // namespace webrtc
#endif  // CALL_VIDEO_RTP_SENDER_INTERFACE_H_