blob: b58042197ebea8ad0184a53ece0a5d61477527d4 [file] [log] [blame]
* 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.
namespace webrtc {
class PacedSender;
class PacketRouter;
class RtpPacketSender;
struct RtpKeepAliveConfig;
class SendSideCongestionController;
class TransportFeedbackObserver;
// 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 {
virtual ~RtpTransportControllerSendInterface() {}
virtual PacketRouter* packet_router() = 0;
virtual PacedSender* pacer() = 0;
// Currently returning the same pointer, but with different types.
virtual SendSideCongestionController* send_side_cc() = 0;
virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
virtual RtpPacketSender* packet_sender() = 0;
virtual const RtpKeepAliveConfig& keepalive_config() const = 0;
// SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
// settings.
// |min_send_bitrate_bps| is the total minimum send bitrate required by all
// sending streams. This is the minimum bitrate the PacedSender will use.
// Note that SendSideCongestionController::OnNetworkChanged can still be
// called with a lower bitrate estimate. |max_padding_bitrate_bps| is the max
// bitrate the send streams request for padding. This can be higher than the
// current network estimate and tells the PacedSender how much it should max
// pad unless there is real packets to send.
virtual void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
int max_padding_bitrate_bps) = 0;
} // namespace webrtc