This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
93 lines
3.5 KiB
C++
93 lines
3.5 KiB
C++
/*
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* Copyright 2015 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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// This file contains classes that implement RtpReceiverInterface.
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// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
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// transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
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#ifndef PC_RTP_RECEIVER_H_
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#define PC_RTP_RECEIVER_H_
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#include <stdint.h>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/dtls_transport_interface.h"
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#include "api/media_stream_interface.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "media/base/media_channel.h"
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namespace webrtc {
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// Internal class used by PeerConnection.
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class RtpReceiverInternal : public RtpReceiverInterface {
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public:
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// Call on the signaling thread, to let the receiver know that the the
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// embedded source object should enter a stopped/ended state and the track's
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// state set to `kEnded`, a final state that cannot be reversed.
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virtual void Stop() = 0;
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// Sets the underlying MediaEngine channel associated with this RtpSender.
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// A VoiceMediaChannel should be used for audio RtpSenders and
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// a VideoMediaChannel should be used for video RtpSenders.
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// NOTE:
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// * SetMediaChannel(nullptr) must be called before the media channel is
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// destroyed.
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// * This method must be invoked on the worker thread.
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virtual void SetMediaChannel(
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cricket::MediaReceiveChannelInterface* media_channel) = 0;
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// Configures the RtpReceiver with the underlying media channel, with the
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// given SSRC as the stream identifier.
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virtual void SetupMediaChannel(uint32_t ssrc) = 0;
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// Configures the RtpReceiver with the underlying media channel to receive an
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// unsignaled receive stream.
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virtual void SetupUnsignaledMediaChannel() = 0;
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virtual void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
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// This SSRC is used as an identifier for the receiver between the API layer
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// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
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virtual absl::optional<uint32_t> ssrc() const = 0;
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// Call this to notify the RtpReceiver when the first packet has been received
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// on the corresponding channel.
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virtual void NotifyFirstPacketReceived() = 0;
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// Set the associated remote media streams for this receiver. The remote track
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// will be removed from any streams that are no longer present and added to
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// any new streams.
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virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
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// TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
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// set_stream_ids() as soon as downstream projects are no longer dependent on
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// stream objects.
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virtual void SetStreams(
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
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// Returns an ID that changes if the attached track changes, but
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// otherwise remains constant. Used to generate IDs for stats.
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// The special value zero means that no track is attached.
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virtual int AttachmentId() const = 0;
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protected:
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static int GenerateUniqueId();
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static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
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CreateStreamsFromIds(std::vector<std::string> stream_ids);
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};
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} // namespace webrtc
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#endif // PC_RTP_RECEIVER_H_
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