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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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| 24 #include "webrtc/base/logging.h" | 24 #include "webrtc/base/logging.h" |
| 25 #include "webrtc/base/networkroute.h" | 25 #include "webrtc/base/networkroute.h" |
| 26 #include "webrtc/base/optional.h" | 26 #include "webrtc/base/optional.h" |
| 27 #include "webrtc/base/sigslot.h" | 27 #include "webrtc/base/sigslot.h" |
| 28 #include "webrtc/base/socket.h" | 28 #include "webrtc/base/socket.h" |
| 29 #include "webrtc/base/window.h" | 29 #include "webrtc/base/window.h" |
| 30 #include "webrtc/config.h" | 30 #include "webrtc/config.h" |
| 31 #include "webrtc/media/base/codec.h" | 31 #include "webrtc/media/base/codec.h" |
| 32 #include "webrtc/media/base/mediaconstants.h" | 32 #include "webrtc/media/base/mediaconstants.h" |
| 33 #include "webrtc/media/base/streamparams.h" | 33 #include "webrtc/media/base/streamparams.h" |
| 34 // TODO(nisse): Temporarily; to be replaced with a forward declaration | |
| 35 // of webrtc::VideoFrame when dependency on cricket::VideoFrame is deleted. | |
| 36 #include "webrtc/media/base/videoframe.h" | |
| 37 #include "webrtc/media/base/videosinkinterface.h" | 34 #include "webrtc/media/base/videosinkinterface.h" |
| 38 #include "webrtc/media/base/videosourceinterface.h" | 35 #include "webrtc/media/base/videosourceinterface.h" |
| 39 // TODO(juberti): re-evaluate this include | 36 // TODO(juberti): re-evaluate this include |
| 40 #include "webrtc/pc/audiomonitor.h" | 37 #include "webrtc/pc/audiomonitor.h" |
| 41 | 38 |
| 42 namespace rtc { | 39 namespace rtc { |
| 43 class RateLimiter; | 40 class RateLimiter; |
| 44 class Timing; | 41 class Timing; |
| 45 } | 42 } |
| 46 | 43 |
| 47 namespace webrtc { | 44 namespace webrtc { |
| 48 class AudioSinkInterface; | 45 class AudioSinkInterface; |
| 46 class VideoFrame; |
| 49 } | 47 } |
| 50 | 48 |
| 51 namespace cricket { | 49 namespace cricket { |
| 52 | 50 |
| 53 class AudioSource; | 51 class AudioSource; |
| 54 class ScreencastId; | 52 class ScreencastId; |
| 55 class VideoCapturer; | 53 class VideoCapturer; |
| 56 struct RtpHeader; | 54 struct RtpHeader; |
| 57 struct VideoFormat; | 55 struct VideoFormat; |
| 58 | 56 |
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| 1000 // Gets the currently set codecs/payload types to be used for outgoing media. | 998 // Gets the currently set codecs/payload types to be used for outgoing media. |
| 1001 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; | 999 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; |
| 1002 // Starts or stops transmission (and potentially capture) of local video. | 1000 // Starts or stops transmission (and potentially capture) of local video. |
| 1003 virtual bool SetSend(bool send) = 0; | 1001 virtual bool SetSend(bool send) = 0; |
| 1004 // Configure stream for sending and register a source. | 1002 // Configure stream for sending and register a source. |
| 1005 // The |ssrc| must correspond to a registered send stream. | 1003 // The |ssrc| must correspond to a registered send stream. |
| 1006 virtual bool SetVideoSend( | 1004 virtual bool SetVideoSend( |
| 1007 uint32_t ssrc, | 1005 uint32_t ssrc, |
| 1008 bool enable, | 1006 bool enable, |
| 1009 const VideoOptions* options, | 1007 const VideoOptions* options, |
| 1010 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; | 1008 rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0; |
| 1011 // Sets the sink object to be used for the specified stream. | 1009 // Sets the sink object to be used for the specified stream. |
| 1012 // If SSRC is 0, the renderer is used for the 'default' stream. | 1010 // If SSRC is 0, the renderer is used for the 'default' stream. |
| 1013 virtual bool SetSink(uint32_t ssrc, | 1011 virtual bool SetSink(uint32_t ssrc, |
| 1014 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | 1012 rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0; |
| 1015 // Gets quality stats for the channel. | 1013 // Gets quality stats for the channel. |
| 1016 virtual bool GetStats(VideoMediaInfo* info) = 0; | 1014 virtual bool GetStats(VideoMediaInfo* info) = 0; |
| 1017 }; | 1015 }; |
| 1018 | 1016 |
| 1019 enum DataMessageType { | 1017 enum DataMessageType { |
| 1020 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 1018 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
| 1021 // values. | 1019 // values. |
| 1022 DMT_NONE = 0, | 1020 DMT_NONE = 0, |
| 1023 DMT_CONTROL = 1, | 1021 DMT_CONTROL = 1, |
| 1024 DMT_BINARY = 2, | 1022 DMT_BINARY = 2, |
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| 1134 // Signal when the media channel is ready to send the stream. Arguments are: | 1132 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1135 // writable(bool) | 1133 // writable(bool) |
| 1136 sigslot::signal1<bool> SignalReadyToSend; | 1134 sigslot::signal1<bool> SignalReadyToSend; |
| 1137 // Signal for notifying that the remote side has closed the DataChannel. | 1135 // Signal for notifying that the remote side has closed the DataChannel. |
| 1138 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1136 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
| 1139 }; | 1137 }; |
| 1140 | 1138 |
| 1141 } // namespace cricket | 1139 } // namespace cricket |
| 1142 | 1140 |
| 1143 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1141 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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