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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_PC_RTPSENDER_H_ |
16 #define WEBRTC_API_RTPSENDER_H_ | 16 #define WEBRTC_PC_RTPSENDER_H_ |
17 | 17 |
18 #include <memory> | 18 #include <memory> |
19 #include <string> | 19 #include <string> |
20 | 20 |
21 #include "webrtc/api/mediastreaminterface.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" | |
24 #include "webrtc/base/basictypes.h" | 23 #include "webrtc/base/basictypes.h" |
25 #include "webrtc/base/criticalsection.h" | 24 #include "webrtc/base/criticalsection.h" |
26 #include "webrtc/media/base/audiosource.h" | 25 #include "webrtc/media/base/audiosource.h" |
27 #include "webrtc/pc/channel.h" | 26 #include "webrtc/pc/channel.h" |
| 27 #include "webrtc/pc/statscollector.h" |
28 | 28 |
29 namespace webrtc { | 29 namespace webrtc { |
30 | 30 |
31 // Internal interface used by PeerConnection. | 31 // Internal interface used by PeerConnection. |
32 class RtpSenderInternal : public RtpSenderInterface { | 32 class RtpSenderInternal : public RtpSenderInterface { |
33 public: | 33 public: |
34 // 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. |
35 // 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 |
36 // 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 |
37 // description). | 37 // description). |
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225 rtc::scoped_refptr<VideoTrackInterface> track_; | 225 rtc::scoped_refptr<VideoTrackInterface> track_; |
226 uint32_t ssrc_ = 0; | 226 uint32_t ssrc_ = 0; |
227 bool cached_track_enabled_ = false; | 227 bool cached_track_enabled_ = false; |
228 VideoTrackInterface::ContentHint cached_track_content_hint_ = | 228 VideoTrackInterface::ContentHint cached_track_content_hint_ = |
229 VideoTrackInterface::ContentHint::kNone; | 229 VideoTrackInterface::ContentHint::kNone; |
230 bool stopped_ = false; | 230 bool stopped_ = false; |
231 }; | 231 }; |
232 | 232 |
233 } // namespace webrtc | 233 } // namespace webrtc |
234 | 234 |
235 #endif // WEBRTC_API_RTPSENDER_H_ | 235 #endif // WEBRTC_PC_RTPSENDER_H_ |
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