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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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22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
23 #include "webrtc/base/networkroute.h" | 23 #include "webrtc/base/networkroute.h" |
24 #include "webrtc/base/optional.h" | 24 #include "webrtc/base/optional.h" |
25 #include "webrtc/base/sigslot.h" | 25 #include "webrtc/base/sigslot.h" |
26 #include "webrtc/base/socket.h" | 26 #include "webrtc/base/socket.h" |
27 #include "webrtc/base/window.h" | 27 #include "webrtc/base/window.h" |
28 #include "webrtc/media/base/codec.h" | 28 #include "webrtc/media/base/codec.h" |
29 #include "webrtc/media/base/mediaconstants.h" | 29 #include "webrtc/media/base/mediaconstants.h" |
30 #include "webrtc/media/base/streamparams.h" | 30 #include "webrtc/media/base/streamparams.h" |
31 #include "webrtc/media/base/videosinkinterface.h" | 31 #include "webrtc/media/base/videosinkinterface.h" |
| 32 #include "webrtc/media/base/videosourceinterface.h" |
32 // TODO(juberti): re-evaluate this include | 33 // TODO(juberti): re-evaluate this include |
33 #include "webrtc/pc/audiomonitor.h" | 34 #include "webrtc/pc/audiomonitor.h" |
34 | 35 |
35 namespace rtc { | 36 namespace rtc { |
36 class Buffer; | 37 class Buffer; |
37 class RateLimiter; | 38 class RateLimiter; |
38 class Timing; | 39 class Timing; |
39 } | 40 } |
40 | 41 |
41 namespace webrtc { | 42 namespace webrtc { |
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991 // Starts or stops transmission (and potentially capture) of local video. | 992 // Starts or stops transmission (and potentially capture) of local video. |
992 virtual bool SetSend(bool send) = 0; | 993 virtual bool SetSend(bool send) = 0; |
993 // Configure stream for sending. | 994 // Configure stream for sending. |
994 virtual bool SetVideoSend(uint32_t ssrc, | 995 virtual bool SetVideoSend(uint32_t ssrc, |
995 bool enable, | 996 bool enable, |
996 const VideoOptions* options) = 0; | 997 const VideoOptions* options) = 0; |
997 // Sets the sink object to be used for the specified stream. | 998 // Sets the sink object to be used for the specified stream. |
998 // If SSRC is 0, the renderer is used for the 'default' stream. | 999 // If SSRC is 0, the renderer is used for the 'default' stream. |
999 virtual bool SetSink(uint32_t ssrc, | 1000 virtual bool SetSink(uint32_t ssrc, |
1000 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | 1001 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; |
1001 // If |ssrc| is 0, replace the default capturer (engine capturer) with | 1002 // Register a source. The |ssrc| must correspond to a registered send stream. |
1002 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. | 1003 virtual void SetSource( |
1003 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; | 1004 uint32_t ssrc, |
| 1005 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; |
1004 // Gets quality stats for the channel. | 1006 // Gets quality stats for the channel. |
1005 virtual bool GetStats(VideoMediaInfo* info) = 0; | 1007 virtual bool GetStats(VideoMediaInfo* info) = 0; |
1006 }; | 1008 }; |
1007 | 1009 |
1008 enum DataMessageType { | 1010 enum DataMessageType { |
1009 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 1011 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
1010 // values. | 1012 // values. |
1011 DMT_NONE = 0, | 1013 DMT_NONE = 0, |
1012 DMT_CONTROL = 1, | 1014 DMT_CONTROL = 1, |
1013 DMT_BINARY = 2, | 1015 DMT_BINARY = 2, |
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1123 // Signal when the media channel is ready to send the stream. Arguments are: | 1125 // Signal when the media channel is ready to send the stream. Arguments are: |
1124 // writable(bool) | 1126 // writable(bool) |
1125 sigslot::signal1<bool> SignalReadyToSend; | 1127 sigslot::signal1<bool> SignalReadyToSend; |
1126 // Signal for notifying that the remote side has closed the DataChannel. | 1128 // Signal for notifying that the remote side has closed the DataChannel. |
1127 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1129 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1128 }; | 1130 }; |
1129 | 1131 |
1130 } // namespace cricket | 1132 } // namespace cricket |
1131 | 1133 |
1132 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1134 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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