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1 /* | 1 /* |
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 // This file contains classes that implement RtpSenderInterface. | 11 // This file contains classes that implement RtpSenderInterface. |
12 // An RtpSender associates a MediaStreamTrackInterface with an underlying | 12 // An RtpSender associates a MediaStreamTrackInterface with an underlying |
13 // transport (provided by AudioProviderInterface/VideoProviderInterface) | 13 // transport (provided by AudioProviderInterface/VideoProviderInterface) |
14 | 14 |
15 #ifndef WEBRTC_API_RTPSENDER_H_ | 15 #ifndef WEBRTC_API_RTPSENDER_H_ |
16 #define WEBRTC_API_RTPSENDER_H_ | 16 #define WEBRTC_API_RTPSENDER_H_ |
17 | 17 |
18 #include <memory> | 18 #include <memory> |
19 #include <string> | 19 #include <string> |
20 | 20 |
21 #include "webrtc/api/mediastreamprovider.h" | 21 #include "webrtc/api/mediastreaminterface.h" |
22 #include "webrtc/api/rtpsenderinterface.h" | 22 #include "webrtc/api/rtpsenderinterface.h" |
23 #include "webrtc/api/statscollector.h" | 23 #include "webrtc/api/statscollector.h" |
24 #include "webrtc/base/basictypes.h" | 24 #include "webrtc/base/basictypes.h" |
25 #include "webrtc/base/criticalsection.h" | 25 #include "webrtc/base/criticalsection.h" |
26 #include "webrtc/media/base/audiosource.h" | 26 #include "webrtc/media/base/audiosource.h" |
| 27 #include "webrtc/pc/channel.h" |
27 | 28 |
28 namespace webrtc { | 29 namespace webrtc { |
29 | 30 |
30 // Internal interface used by PeerConnection. | 31 // Internal interface used by PeerConnection. |
31 class RtpSenderInternal : public RtpSenderInterface { | 32 class RtpSenderInternal : public RtpSenderInterface { |
32 public: | 33 public: |
33 // Used to set the SSRC of the sender, once a local description has been set. | 34 // Used to set the SSRC of the sender, once a local description has been set. |
34 // If |ssrc| is 0, this indiates that the sender should disconnect from the | 35 // If |ssrc| is 0, this indiates that the sender should disconnect from the |
35 // underlying transport (this occurs if the sender isn't seen in a local | 36 // underlying transport (this occurs if the sender isn't seen in a local |
36 // description). | 37 // description). |
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65 cricket::AudioSource::Sink* sink_; | 66 cricket::AudioSource::Sink* sink_; |
66 // Critical section protecting |sink_|. | 67 // Critical section protecting |sink_|. |
67 rtc::CriticalSection lock_; | 68 rtc::CriticalSection lock_; |
68 }; | 69 }; |
69 | 70 |
70 class AudioRtpSender : public ObserverInterface, | 71 class AudioRtpSender : public ObserverInterface, |
71 public rtc::RefCountedObject<RtpSenderInternal> { | 72 public rtc::RefCountedObject<RtpSenderInternal> { |
72 public: | 73 public: |
73 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called | 74 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called |
74 // at the appropriate times. | 75 // at the appropriate times. |
| 76 // |channel| can be null if one does not exist yet. |
75 AudioRtpSender(AudioTrackInterface* track, | 77 AudioRtpSender(AudioTrackInterface* track, |
76 const std::string& stream_id, | 78 const std::string& stream_id, |
77 AudioProviderInterface* provider, | 79 cricket::VoiceChannel* channel, |
78 StatsCollector* stats); | 80 StatsCollector* stats); |
79 | 81 |
80 // Randomly generates stream_id. | 82 // Randomly generates stream_id. |
| 83 // |channel| can be null if one does not exist yet. |
81 AudioRtpSender(AudioTrackInterface* track, | 84 AudioRtpSender(AudioTrackInterface* track, |
82 AudioProviderInterface* provider, | 85 cricket::VoiceChannel* channel, |
83 StatsCollector* stats); | 86 StatsCollector* stats); |
84 | 87 |
85 // Randomly generates id and stream_id. | 88 // Randomly generates id and stream_id. |
86 AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); | 89 // |channel| can be null if one does not exist yet. |
| 90 AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats); |
87 | 91 |
88 virtual ~AudioRtpSender(); | 92 virtual ~AudioRtpSender(); |
89 | 93 |
90 // ObserverInterface implementation | 94 // ObserverInterface implementation |
91 void OnChanged() override; | 95 void OnChanged() override; |
92 | 96 |
93 // RtpSenderInterface implementation | 97 // RtpSenderInterface implementation |
94 bool SetTrack(MediaStreamTrackInterface* track) override; | 98 bool SetTrack(MediaStreamTrackInterface* track) override; |
95 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 99 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
96 return track_; | 100 return track_; |
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115 // RtpSenderInternal implementation. | 119 // RtpSenderInternal implementation. |
116 void SetSsrc(uint32_t ssrc) override; | 120 void SetSsrc(uint32_t ssrc) override; |
117 | 121 |
118 void set_stream_id(const std::string& stream_id) override { | 122 void set_stream_id(const std::string& stream_id) override { |
119 stream_id_ = stream_id; | 123 stream_id_ = stream_id; |
120 } | 124 } |
121 std::string stream_id() const override { return stream_id_; } | 125 std::string stream_id() const override { return stream_id_; } |
122 | 126 |
123 void Stop() override; | 127 void Stop() override; |
124 | 128 |
| 129 // Does not take ownership. |
| 130 // Should call SetChannel(nullptr) before |channel| is destroyed. |
| 131 void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; } |
| 132 |
125 private: | 133 private: |
126 // TODO(nisse): Since SSRC == 0 is technically valid, figure out | 134 // TODO(nisse): Since SSRC == 0 is technically valid, figure out |
127 // some other way to test if we have a valid SSRC. | 135 // some other way to test if we have a valid SSRC. |
128 bool can_send_track() const { return track_ && ssrc_; } | 136 bool can_send_track() const { return track_ && ssrc_; } |
129 // Helper function to construct options for | 137 // Helper function to construct options for |
130 // AudioProviderInterface::SetAudioSend. | 138 // AudioProviderInterface::SetAudioSend. |
131 void SetAudioSend(); | 139 void SetAudioSend(); |
| 140 // Helper function to call SetAudioSend with "stop sending" parameters. |
| 141 void ClearAudioSend(); |
132 | 142 |
133 std::string id_; | 143 std::string id_; |
134 std::string stream_id_; | 144 std::string stream_id_; |
135 AudioProviderInterface* provider_; | 145 cricket::VoiceChannel* channel_ = nullptr; |
136 StatsCollector* stats_; | 146 StatsCollector* stats_; |
137 rtc::scoped_refptr<AudioTrackInterface> track_; | 147 rtc::scoped_refptr<AudioTrackInterface> track_; |
138 uint32_t ssrc_ = 0; | 148 uint32_t ssrc_ = 0; |
139 bool cached_track_enabled_ = false; | 149 bool cached_track_enabled_ = false; |
140 bool stopped_ = false; | 150 bool stopped_ = false; |
141 | 151 |
142 // Used to pass the data callback from the |track_| to the other end of | 152 // Used to pass the data callback from the |track_| to the other end of |
143 // cricket::AudioSource. | 153 // cricket::AudioSource. |
144 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; | 154 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; |
145 }; | 155 }; |
146 | 156 |
147 class VideoRtpSender : public ObserverInterface, | 157 class VideoRtpSender : public ObserverInterface, |
148 public rtc::RefCountedObject<RtpSenderInternal> { | 158 public rtc::RefCountedObject<RtpSenderInternal> { |
149 public: | 159 public: |
| 160 // |channel| can be null if one does not exist yet. |
150 VideoRtpSender(VideoTrackInterface* track, | 161 VideoRtpSender(VideoTrackInterface* track, |
151 const std::string& stream_id, | 162 const std::string& stream_id, |
152 VideoProviderInterface* provider); | 163 cricket::VideoChannel* channel); |
153 | 164 |
154 // Randomly generates stream_id. | 165 // Randomly generates stream_id. |
155 VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider); | 166 // |channel| can be null if one does not exist yet. |
| 167 VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel); |
156 | 168 |
157 // Randomly generates id and stream_id. | 169 // Randomly generates id and stream_id. |
158 explicit VideoRtpSender(VideoProviderInterface* provider); | 170 // |channel| can be null if one does not exist yet. |
| 171 explicit VideoRtpSender(cricket::VideoChannel* channel); |
159 | 172 |
160 virtual ~VideoRtpSender(); | 173 virtual ~VideoRtpSender(); |
161 | 174 |
162 // ObserverInterface implementation | 175 // ObserverInterface implementation |
163 void OnChanged() override; | 176 void OnChanged() override; |
164 | 177 |
165 // RtpSenderInterface implementation | 178 // RtpSenderInterface implementation |
166 bool SetTrack(MediaStreamTrackInterface* track) override; | 179 bool SetTrack(MediaStreamTrackInterface* track) override; |
167 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 180 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
168 return track_; | 181 return track_; |
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187 // RtpSenderInternal implementation. | 200 // RtpSenderInternal implementation. |
188 void SetSsrc(uint32_t ssrc) override; | 201 void SetSsrc(uint32_t ssrc) override; |
189 | 202 |
190 void set_stream_id(const std::string& stream_id) override { | 203 void set_stream_id(const std::string& stream_id) override { |
191 stream_id_ = stream_id; | 204 stream_id_ = stream_id; |
192 } | 205 } |
193 std::string stream_id() const override { return stream_id_; } | 206 std::string stream_id() const override { return stream_id_; } |
194 | 207 |
195 void Stop() override; | 208 void Stop() override; |
196 | 209 |
| 210 // Does not take ownership. |
| 211 // Should call SetChannel(nullptr) before |channel| is destroyed. |
| 212 void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; } |
| 213 |
197 private: | 214 private: |
198 bool can_send_track() const { return track_ && ssrc_; } | 215 bool can_send_track() const { return track_ && ssrc_; } |
199 // Helper function to construct options for | 216 // Helper function to construct options for |
200 // VideoProviderInterface::SetVideoSend. | 217 // VideoProviderInterface::SetVideoSend. |
201 void SetVideoSend(); | 218 void SetVideoSend(); |
202 // Helper function to call SetVideoSend with "stop sending" parameters. | 219 // Helper function to call SetVideoSend with "stop sending" parameters. |
203 void ClearVideoSend(); | 220 void ClearVideoSend(); |
204 | 221 |
205 std::string id_; | 222 std::string id_; |
206 std::string stream_id_; | 223 std::string stream_id_; |
207 VideoProviderInterface* provider_; | 224 cricket::VideoChannel* channel_ = nullptr; |
208 rtc::scoped_refptr<VideoTrackInterface> track_; | 225 rtc::scoped_refptr<VideoTrackInterface> track_; |
209 uint32_t ssrc_ = 0; | 226 uint32_t ssrc_ = 0; |
210 bool cached_track_enabled_ = false; | 227 bool cached_track_enabled_ = false; |
211 bool stopped_ = false; | 228 bool stopped_ = false; |
212 }; | 229 }; |
213 | 230 |
214 } // namespace webrtc | 231 } // namespace webrtc |
215 | 232 |
216 #endif // WEBRTC_API_RTPSENDER_H_ | 233 #endif // WEBRTC_API_RTPSENDER_H_ |
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