webrtc/call/rtp_stream_receiver_controller_interface.h
Niels Möller cb99ccd244 Update/delete old TODOs
Bug: webrtc:10198
Change-Id: I0341e068d792bc0b143db86e675988f4cd07ff2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267822
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37454}
2022-07-06 07:49:04 +00:00

46 lines
1.8 KiB
C++

/*
* 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_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
#include <memory>
#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
// An RtpStreamReceiver is responsible for the rtp-specific but
// media-independent state needed for receiving an RTP stream.
// TODO(bugs.webrtc.org/7135): Currently, only owns the association between ssrc
// and the stream's RtpPacketSinkInterface. Ownership of corresponding objects
// from modules/rtp_rtcp/ should move to this class (or rather, the
// corresponding implementation class). We should add methods for getting rtp
// receive stats, and for sending RTCP messages related to the receive stream.
class RtpStreamReceiverInterface {
public:
virtual ~RtpStreamReceiverInterface() {}
};
// This class acts as a factory for RtpStreamReceiver objects.
class RtpStreamReceiverControllerInterface {
public:
virtual ~RtpStreamReceiverControllerInterface() {}
virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
uint32_t ssrc,
RtpPacketSinkInterface* sink) = 0;
// For registering additional sinks, needed for FlexFEC.
virtual bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
};
} // namespace webrtc
#endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_