chromium/third_party/webrtc/call/rtp_transport_controller_send_interface.h

/*
 *  Copyright (c) 2017 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_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
#include <stddef.h>
#include <stdint.h>

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

#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_packet_sender.h"
#include "api/transport/bandwidth_estimation_settings.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/units/timestamp.h"
#include "call/rtp_config.h"
#include "common_video/frame_counts.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"

namespace rtc {
struct SentPacket;
struct NetworkRoute;
}  // namespace rtc
namespace webrtc {

class FrameEncryptorInterface;
class TargetTransferRateObserver;
class Transport;
class PacketRouter;
class RtpVideoSenderInterface;
class RtpPacketSender;
class RtpRtcpInterface;

struct RtpSenderObservers { … };

struct RtpSenderFrameEncryptionConfig { … };

// An RtpTransportController should own everything related to the RTP
// transport to/from a remote endpoint. We should have separate
// interfaces for send and receive side, even if they are implemented
// by the same class. This is an ongoing refactoring project. At some
// point, this class should be promoted to a public api under
// webrtc/api/rtp/.
//
// For a start, this object is just a collection of the objects needed
// by the VideoSendStream constructor. The plan is to move ownership
// of all RTP-related objects here, and add methods to create per-ssrc
// objects which would then be passed to VideoSendStream. Eventually,
// direct accessors like packet_router() should be removed.
//
// This should also have a reference to the underlying
// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
// WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
// WebrtcSession. Video and audio always uses different transport
// objects, even in the common case where they are bundled over the
// same underlying transport.
//
// Extracting the logic of the webrtc::Transport from BaseChannel and
// subclasses into a separate class seems to be a prerequesite for
// moving the transport here.
class RtpTransportControllerSendInterface { … };

}  // namespace webrtc

#endif  // CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_