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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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 |
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21 #include "webrtc/base/dscp.h" | 21 #include "webrtc/base/dscp.h" |
22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
23 #include "webrtc/base/optional.h" | 23 #include "webrtc/base/optional.h" |
24 #include "webrtc/base/sigslot.h" | 24 #include "webrtc/base/sigslot.h" |
25 #include "webrtc/base/socket.h" | 25 #include "webrtc/base/socket.h" |
26 #include "webrtc/base/window.h" | 26 #include "webrtc/base/window.h" |
27 #include "webrtc/media/base/codec.h" | 27 #include "webrtc/media/base/codec.h" |
28 #include "webrtc/media/base/mediaconstants.h" | 28 #include "webrtc/media/base/mediaconstants.h" |
29 #include "webrtc/media/base/streamparams.h" | 29 #include "webrtc/media/base/streamparams.h" |
30 #include "webrtc/media/base/videosinkinterface.h" | 30 #include "webrtc/media/base/videosinkinterface.h" |
31 #include "webrtc/media/base/videosourceinterface.h" | |
31 // TODO(juberti): re-evaluate this include | 32 // TODO(juberti): re-evaluate this include |
32 #include "webrtc/pc/audiomonitor.h" | 33 #include "webrtc/pc/audiomonitor.h" |
33 | 34 |
34 namespace rtc { | 35 namespace rtc { |
35 class Buffer; | 36 class Buffer; |
36 class RateLimiter; | 37 class RateLimiter; |
37 class Timing; | 38 class Timing; |
38 } | 39 } |
39 | 40 |
40 namespace webrtc { | 41 namespace webrtc { |
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986 // Starts or stops transmission (and potentially capture) of local video. | 987 // Starts or stops transmission (and potentially capture) of local video. |
987 virtual bool SetSend(bool send) = 0; | 988 virtual bool SetSend(bool send) = 0; |
988 // Configure stream for sending. | 989 // Configure stream for sending. |
989 virtual bool SetVideoSend(uint32_t ssrc, | 990 virtual bool SetVideoSend(uint32_t ssrc, |
990 bool enable, | 991 bool enable, |
991 const VideoOptions* options) = 0; | 992 const VideoOptions* options) = 0; |
992 // Sets the sink object to be used for the specified stream. | 993 // Sets the sink object to be used for the specified stream. |
993 // If SSRC is 0, the renderer is used for the 'default' stream. | 994 // If SSRC is 0, the renderer is used for the 'default' stream. |
994 virtual bool SetSink(uint32_t ssrc, | 995 virtual bool SetSink(uint32_t ssrc, |
995 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | 996 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; |
996 // If |ssrc| is 0, replace the default capturer (engine capturer) with | 997 // Register a source. The |ssrc| must correspond to a registered |
pbos-webrtc
2016/03/31 14:14:31
reformat comment (send fits on previous line)
nisse-webrtc
2016/04/01 09:35:39
Done. You use fill-column 80?
pbos-webrtc
2016/04/01 10:55:28
Not sure what that is in vim speak, we have 80 cha
| |
997 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. | 998 // send stream. |
998 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; | 999 virtual void SetSource( |
1000 uint32_t ssrc, | |
1001 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; | |
999 // Gets quality stats for the channel. | 1002 // Gets quality stats for the channel. |
1000 virtual bool GetStats(VideoMediaInfo* info) = 0; | 1003 virtual bool GetStats(VideoMediaInfo* info) = 0; |
1001 }; | 1004 }; |
1002 | 1005 |
1003 enum DataMessageType { | 1006 enum DataMessageType { |
1004 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 1007 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
1005 // values. | 1008 // values. |
1006 DMT_NONE = 0, | 1009 DMT_NONE = 0, |
1007 DMT_CONTROL = 1, | 1010 DMT_CONTROL = 1, |
1008 DMT_BINARY = 2, | 1011 DMT_BINARY = 2, |
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1115 // Signal when the media channel is ready to send the stream. Arguments are: | 1118 // Signal when the media channel is ready to send the stream. Arguments are: |
1116 // writable(bool) | 1119 // writable(bool) |
1117 sigslot::signal1<bool> SignalReadyToSend; | 1120 sigslot::signal1<bool> SignalReadyToSend; |
1118 // Signal for notifying that the remote side has closed the DataChannel. | 1121 // Signal for notifying that the remote side has closed the DataChannel. |
1119 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1122 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1120 }; | 1123 }; |
1121 | 1124 |
1122 } // namespace cricket | 1125 } // namespace cricket |
1123 | 1126 |
1124 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1127 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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