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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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| 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 send stream. |
| 997 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. | 998 virtual void SetSource( |
| 998 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; | 999 uint32_t ssrc, |
| 1000 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; |
| 999 // Gets quality stats for the channel. | 1001 // Gets quality stats for the channel. |
| 1000 virtual bool GetStats(VideoMediaInfo* info) = 0; | 1002 virtual bool GetStats(VideoMediaInfo* info) = 0; |
| 1001 }; | 1003 }; |
| 1002 | 1004 |
| 1003 enum DataMessageType { | 1005 enum DataMessageType { |
| 1004 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 1006 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
| 1005 // values. | 1007 // values. |
| 1006 DMT_NONE = 0, | 1008 DMT_NONE = 0, |
| 1007 DMT_CONTROL = 1, | 1009 DMT_CONTROL = 1, |
| 1008 DMT_BINARY = 2, | 1010 DMT_BINARY = 2, |
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| 1118 // Signal when the media channel is ready to send the stream. Arguments are: | 1120 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1119 // writable(bool) | 1121 // writable(bool) |
| 1120 sigslot::signal1<bool> SignalReadyToSend; | 1122 sigslot::signal1<bool> SignalReadyToSend; |
| 1121 // Signal for notifying that the remote side has closed the DataChannel. | 1123 // Signal for notifying that the remote side has closed the DataChannel. |
| 1122 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1124 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
| 1123 }; | 1125 }; |
| 1124 | 1126 |
| 1125 } // namespace cricket | 1127 } // namespace cricket |
| 1126 | 1128 |
| 1127 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1129 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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