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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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| 978 // Starts or stops transmission (and potentially capture) of local video. | 979 // Starts or stops transmission (and potentially capture) of local video. |
| 979 virtual bool SetSend(bool send) = 0; | 980 virtual bool SetSend(bool send) = 0; |
| 980 // Configure stream for sending. | 981 // Configure stream for sending. |
| 981 virtual bool SetVideoSend(uint32_t ssrc, | 982 virtual bool SetVideoSend(uint32_t ssrc, |
| 982 bool enable, | 983 bool enable, |
| 983 const VideoOptions* options) = 0; | 984 const VideoOptions* options) = 0; |
| 984 // Sets the sink object to be used for the specified stream. | 985 // Sets the sink object to be used for the specified stream. |
| 985 // If SSRC is 0, the renderer is used for the 'default' stream. | 986 // If SSRC is 0, the renderer is used for the 'default' stream. |
| 986 virtual bool SetSink(uint32_t ssrc, | 987 virtual bool SetSink(uint32_t ssrc, |
| 987 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | 988 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; |
| 988 // If |ssrc| is 0, replace the default capturer (engine capturer) with | 989 // Register a source. The |ssrc| must correspond to a registered |
| 989 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. | 990 // send stream. |
| 990 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; | 991 virtual void SetSource( |
| 992 uint32_t ssrc, |
| 993 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; |
| 991 // Gets quality stats for the channel. | 994 // Gets quality stats for the channel. |
| 992 virtual bool GetStats(VideoMediaInfo* info) = 0; | 995 virtual bool GetStats(VideoMediaInfo* info) = 0; |
| 993 }; | 996 }; |
| 994 | 997 |
| 995 enum DataMessageType { | 998 enum DataMessageType { |
| 996 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 999 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
| 997 // values. | 1000 // values. |
| 998 DMT_NONE = 0, | 1001 DMT_NONE = 0, |
| 999 DMT_CONTROL = 1, | 1002 DMT_CONTROL = 1, |
| 1000 DMT_BINARY = 2, | 1003 DMT_BINARY = 2, |
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| 1107 // Signal when the media channel is ready to send the stream. Arguments are: | 1110 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1108 // writable(bool) | 1111 // writable(bool) |
| 1109 sigslot::signal1<bool> SignalReadyToSend; | 1112 sigslot::signal1<bool> SignalReadyToSend; |
| 1110 // Signal for notifying that the remote side has closed the DataChannel. | 1113 // Signal for notifying that the remote side has closed the DataChannel. |
| 1111 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1114 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
| 1112 }; | 1115 }; |
| 1113 | 1116 |
| 1114 } // namespace cricket | 1117 } // namespace cricket |
| 1115 | 1118 |
| 1116 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1119 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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