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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
47 lines
1.8 KiB
C++
47 lines
1.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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#include <memory>
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#include "call/rtp_packet_sink_interface.h"
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namespace webrtc {
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// An RtpStreamReceiver is responsible for the rtp-specific but
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// media-independent state needed for receiving an RTP stream.
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// TODO(nisse): Currently, only owns the association between ssrc and
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// the stream's RtpPacketSinkInterface. Ownership of corresponding
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// objects from modules/rtp_rtcp/ should move to this class (or
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// rather, the corresponding implementation class). We should add
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// methods for getting rtp receive stats, and for sending RTCP
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// messages related to the receive stream.
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class RtpStreamReceiverInterface {
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public:
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virtual ~RtpStreamReceiverInterface() {}
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};
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// This class acts as a factory for RtpStreamReceiver objects.
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class RtpStreamReceiverControllerInterface {
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public:
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virtual ~RtpStreamReceiverControllerInterface() {}
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virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
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uint32_t ssrc,
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RtpPacketSinkInterface* sink) = 0;
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// For registering additional sinks, needed for FlexFEC.
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virtual bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
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virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
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};
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} // namespace webrtc
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#endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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