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1 /*
2  *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 // This file contains classes that implement RtpReceiverInterface.
12 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying
13 // transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
14 
15 #ifndef PC_RTP_RECEIVER_H_
16 #define PC_RTP_RECEIVER_H_
17 
18 #include <stdint.h>
19 
20 #include <string>
21 #include <vector>
22 
23 #include "absl/types/optional.h"
24 #include "api/crypto/frame_decryptor_interface.h"
25 #include "api/media_stream_interface.h"
26 #include "api/media_types.h"
27 #include "api/rtp_parameters.h"
28 #include "api/rtp_receiver_interface.h"
29 #include "api/scoped_refptr.h"
30 #include "api/video/video_frame.h"
31 #include "api/video/video_sink_interface.h"
32 #include "api/video/video_source_interface.h"
33 #include "media/base/media_channel.h"
34 #include "media/base/video_broadcaster.h"
35 #include "pc/video_track_source.h"
36 #include "rtc_base/ref_counted_object.h"
37 #include "rtc_base/thread.h"
38 
39 namespace webrtc {
40 
41 // Internal class used by PeerConnection.
42 class RtpReceiverInternal : public RtpReceiverInterface {
43  public:
44   virtual void Stop() = 0;
45 
46   // Sets the underlying MediaEngine channel associated with this RtpSender.
47   // A VoiceMediaChannel should be used for audio RtpSenders and
48   // a VideoMediaChannel should be used for video RtpSenders.
49   // Must call SetMediaChannel(nullptr) before the media channel is destroyed.
50   virtual void SetMediaChannel(cricket::MediaChannel* media_channel) = 0;
51 
52   // Configures the RtpReceiver with the underlying media channel, with the
53   // given SSRC as the stream identifier.
54   virtual void SetupMediaChannel(uint32_t ssrc) = 0;
55 
56   // Configures the RtpReceiver with the underlying media channel to receive an
57   // unsignaled receive stream.
58   virtual void SetupUnsignaledMediaChannel() = 0;
59 
60   virtual void set_transport(
61       rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
62   // This SSRC is used as an identifier for the receiver between the API layer
63   // and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
64   virtual uint32_t ssrc() const = 0;
65 
66   // Call this to notify the RtpReceiver when the first packet has been received
67   // on the corresponding channel.
68   virtual void NotifyFirstPacketReceived() = 0;
69 
70   // Set the associated remote media streams for this receiver. The remote track
71   // will be removed from any streams that are no longer present and added to
72   // any new streams.
73   virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
74   // TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
75   // set_stream_ids() as soon as downstream projects are no longer dependent on
76   // stream objects.
77   virtual void SetStreams(
78       const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
79 
80   // Returns an ID that changes if the attached track changes, but
81   // otherwise remains constant. Used to generate IDs for stats.
82   // The special value zero means that no track is attached.
83   virtual int AttachmentId() const = 0;
84 
85  protected:
86   static int GenerateUniqueId();
87 
88   static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
89   CreateStreamsFromIds(std::vector<std::string> stream_ids);
90 
91   static void MaybeAttachFrameDecryptorToMediaChannel(
92       const absl::optional<uint32_t>& ssrc,
93       rtc::Thread* worker_thread,
94       rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
95       cricket::MediaChannel* media_channel,
96       bool stopped);
97 };
98 
99 }  // namespace webrtc
100 
101 #endif  // PC_RTP_RECEIVER_H_
102