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1 /* | |
2 * Copyright 2017 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 #ifndef WEBRTC_ORTC_RTPTRANSPORTCONTROLLERSHIM_H_ | |
12 #define WEBRTC_ORTC_RTPTRANSPORTCONTROLLERSHIM_H_ | |
13 | |
14 #include <memory> | |
15 #include <string> | |
16 #include <vector> | |
17 | |
18 #include "webrtc/base/constructormagic.h" | |
19 #include "webrtc/base/thread.h" | |
20 #include "webrtc/call/call.h" | |
21 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | |
22 #include "webrtc/api/ortc/rtptransportcontrollerinterface.h" | |
23 #include "webrtc/pc/channelmanager.h" | |
24 #include "webrtc/pc/mediacontroller.h" | |
25 #include "webrtc/media/base/mediachannel.h" // For MediaConfig. | |
26 | |
27 namespace webrtc { | |
28 | |
29 // Implementation of RtpTransportControllerInterface. Wraps a MediaController, | |
30 // a VoiceChannel and VideoChannel, and maintains a list of dependent RTP | |
31 // transports. | |
32 // | |
33 // When used along with an RtpSenderShim or RtpReceiverShim, the | |
34 // sender/receiver passes its parameters along to this class, which turns them | |
35 // into cricket:: media descriptions (the interface used by BaseChannel). | |
36 // | |
37 // Due to the fact that BaseChannel has different subclasses for audio/video, | |
38 // the actual BaseChannel object is not created until an RtpSender/RtpReceiver | |
39 // needs them. | |
40 // | |
41 // All methods should be called on the signaling thread. | |
42 // | |
43 // TODO(deadbeef): When BaseChannel is split apart into separate | |
44 // "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this shim | |
45 // object can be replaced by a "real" one. | |
46 class RtpTransportControllerShim : public RtpTransportControllerInterface { | |
47 public: | |
48 // Creates a proxy that will call "public interface" methods on the correct | |
49 // thread. | |
50 // | |
51 // Doesn't take ownership of any objects passed in. | |
52 // | |
53 // |channel_manager| must not be null. | |
54 static std::unique_ptr<RtpTransportControllerInterface> CreateProxied( | |
55 const cricket::MediaConfig& config, | |
56 cricket::ChannelManager* channel_manager, | |
57 webrtc::RtcEventLog* event_log, | |
58 rtc::Thread* signaling_thread, | |
59 rtc::Thread* worker_thread); | |
60 | |
61 ~RtpTransportControllerShim() override; | |
62 | |
63 // RtpTransportControllerInterface implementation. | |
64 std::vector<RtpTransportInterface*> GetTransports() const override; | |
65 | |
66 // Methods used internally by RtpTransportShim. | |
67 MediaControllerInterface* media_controller() const { | |
68 return media_controller_.get(); | |
69 } | |
70 | |
71 rtc::Thread* signaling_thread() const { return signaling_thread_; } | |
72 rtc::Thread* worker_thread() const { return worker_thread_; } | |
73 | |
74 // Doesn't take ownership. | |
75 // | |
76 // NOTE: "AddTransport" takes a proxy class, such that "GetTransports()" can | |
77 // return proxies, but the other methods take a pointer to the inner object, | |
78 // since these methods are called by the inner object which is unaware of the | |
79 // proxy. | |
80 void AddTransport(RtpTransportInterface* transport_proxy); | |
81 void RemoveTransport(RtpTransportInterface* inner_transport); | |
pthatcher1
2017/02/10 22:36:53
If this has to be an "inner" and not a proxy, shou
Taylor Brandstetter
2017/02/14 06:55:05
Done.
| |
82 RTCError SetRtcpParameters(const RtcpParameters& parameters, | |
pthatcher1
2017/02/10 22:36:53
Why is it necessary to first AddTransport and then
Taylor Brandstetter
2017/02/14 06:55:05
AddTransport occurs when the transport is created.
| |
83 RtpTransportInterface* inner_transport); | |
84 | |
85 // Methods used by RtpSenderShim/RtpReceiverShim. | |
86 // | |
87 // AttachSender/AttachReceiver ensures only one sender/receiver shim per | |
88 // media type is trying to use this object simultaneously, and the | |
89 // sender/receiver for the same media type are using the same transport. | |
90 // That's all this class currently supports, due to limits of BaseChannel. | |
91 // | |
92 // The "Detach" methods will cause the corresponding parameters to be | |
93 // cleared, and will allow a different sender or receiver to be connected. | |
94 RTCError AttachAudioSender(RtpTransportInterface* inner_transport); | |
95 RTCError AttachVideoSender(RtpTransportInterface* inner_transport); | |
96 RTCError AttachAudioReceiver(RtpTransportInterface* inner_transport); | |
97 RTCError AttachVideoReceiver(RtpTransportInterface* inner_transport); | |
pthatcher1
2017/02/10 22:36:53
It's weird that it's "Attach$Type$Actioner" when t
Taylor Brandstetter
2017/02/14 06:55:05
I think you misunderstand how these are used; see
| |
98 | |
99 void DetachAudioSender(); | |
100 void DetachVideoSender(); | |
101 void DetachAudioReceiver(); | |
102 void DetachVideoReceiver(); | |
103 | |
104 cricket::VoiceChannel* voice_channel() { return voice_channel_; } | |
105 cricket::VideoChannel* video_channel() { return video_channel_; } | |
pthatcher1
2017/02/10 22:36:53
Why are these public?
Taylor Brandstetter
2017/02/14 06:55:06
Because they're needed by RtpSenderAdapter/RtpRece
| |
106 | |
107 // |primary_ssrc| out parameter is filled with either | |
108 // |parameters.encodings[0].ssrc|, or a generated SSRC if that's left unset. | |
109 RTCError ValidateAndApplyAudioSenderParameters( | |
pthatcher1
2017/02/10 22:36:53
Why not just call it ApplyXParameters? I think th
Taylor Brandstetter
2017/02/14 06:55:06
I disagree. These parameters are passed through so
| |
110 const RtpParameters& parameters, | |
111 uint32_t* primary_ssrc); | |
112 RTCError ValidateAndApplyVideoSenderParameters( | |
113 const RtpParameters& parameters, | |
114 uint32_t* primary_ssrc); | |
115 RTCError ValidateAndApplyAudioReceiverParameters( | |
116 const RtpParameters& parameters); | |
117 RTCError ValidateAndApplyVideoReceiverParameters( | |
118 const RtpParameters& parameters); | |
119 | |
120 protected: | |
121 RtpTransportControllerShim* GetInternal() override { return this; } | |
122 | |
123 private: | |
124 // Only expected to be called by RtpTransportControllerShim::CreateProxied. | |
125 RtpTransportControllerShim(const cricket::MediaConfig& config, | |
126 cricket::ChannelManager* channel_manager, | |
127 webrtc::RtcEventLog* event_log, | |
128 rtc::Thread* signaling_thread, | |
129 rtc::Thread* worker_thread); | |
130 | |
131 void CreateVoiceChannel(); | |
132 void CreateVideoChannel(); | |
133 void DestroyVoiceChannel(); | |
134 void DestroyVideoChannel(); | |
135 | |
136 void CopyRtcpParametersToDescriptions( | |
137 const RtcpParameters& params, | |
138 cricket::MediaContentDescription* local, | |
139 cricket::MediaContentDescription* remote); | |
140 | |
141 // Helper function to generate an SSRC that doesn't match one in any of the | |
142 // "content description" structs, or in |new_params| (which is needed since | |
143 // multiple SSRCs may be gneerated in one go). | |
144 uint32_t GenerateUnusedSsrc(const cricket::StreamParams& new_params) const; | |
145 | |
146 // |description| is the matching description where existing SSRCs can be | |
147 // found. | |
148 // This is a member function because it may need to generate SSRCs | |
149 // that don't match existing ones. | |
150 RTCError ValidateAndConvertSenderEncodings( | |
151 const std::vector<RtpEncodingParameters> encodings, | |
152 const std::string& cname, | |
153 const cricket::MediaContentDescription& description, | |
154 cricket::StreamParamsVec* cricket_streams, | |
155 bool* sending, | |
156 int* bandwidth) const; | |
157 | |
158 rtc::Thread* signaling_thread_; | |
159 rtc::Thread* worker_thread_; | |
160 // |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_| | |
161 // are somewhat redundant, but the latter are only set when | |
162 // RtpSenders/RtpReceivers are attached to the transport. | |
163 std::vector<RtpTransportInterface*> transport_proxies_; | |
164 RtpTransportInterface* inner_audio_transport_ = nullptr; | |
165 RtpTransportInterface* inner_video_transport_ = nullptr; | |
166 std::unique_ptr<MediaControllerInterface> media_controller_; | |
167 | |
168 // BaseChannel takes content descriptions as input, so we store them here | |
169 // such that they can be updated when a new RtpSenderShim/RtpReceiverShim | |
170 // attaches itself. | |
171 cricket::AudioContentDescription local_audio_description_; | |
172 cricket::AudioContentDescription remote_audio_description_; | |
173 cricket::VideoContentDescription local_video_description_; | |
174 cricket::VideoContentDescription remote_video_description_; | |
pthatcher1
2017/02/10 22:36:53
Aren't these store on the VoiceChannel and VideoCh
Taylor Brandstetter
2017/02/14 06:55:05
No and no.
| |
175 cricket::VoiceChannel* voice_channel_ = nullptr; | |
176 cricket::VideoChannel* video_channel_ = nullptr; | |
177 bool have_audio_sender_ = false; | |
178 bool have_video_sender_ = false; | |
179 bool have_audio_receiver_ = false; | |
180 bool have_video_receiver_ = false; | |
181 | |
182 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportControllerShim); | |
183 }; | |
184 | |
185 } // namespace webrtc | |
186 | |
187 #endif // WEBRTC_ORTC_RTPTRANSPORTCONTROLLERSHIM_H_ | |
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