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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 VideoCapturer; | 52 class VideoCapturer; |
55 struct RtpHeader; | 53 struct RtpHeader; |
56 struct VideoFormat; | 54 struct VideoFormat; |
57 | 55 |
58 const int kScreencastDefaultFps = 5; | 56 const int kScreencastDefaultFps = 5; |
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1024 // Gets the currently set codecs/payload types to be used for outgoing media. | 1022 // Gets the currently set codecs/payload types to be used for outgoing media. |
1025 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; | 1023 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; |
1026 // Starts or stops transmission (and potentially capture) of local video. | 1024 // Starts or stops transmission (and potentially capture) of local video. |
1027 virtual bool SetSend(bool send) = 0; | 1025 virtual bool SetSend(bool send) = 0; |
1028 // Configure stream for sending and register a source. | 1026 // Configure stream for sending and register a source. |
1029 // The |ssrc| must correspond to a registered send stream. | 1027 // The |ssrc| must correspond to a registered send stream. |
1030 virtual bool SetVideoSend( | 1028 virtual bool SetVideoSend( |
1031 uint32_t ssrc, | 1029 uint32_t ssrc, |
1032 bool enable, | 1030 bool enable, |
1033 const VideoOptions* options, | 1031 const VideoOptions* options, |
1034 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0; | 1032 rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0; |
1035 // Sets the sink object to be used for the specified stream. | 1033 // Sets the sink object to be used for the specified stream. |
1036 // If SSRC is 0, the renderer is used for the 'default' stream. | 1034 // If SSRC is 0, the renderer is used for the 'default' stream. |
1037 virtual bool SetSink(uint32_t ssrc, | 1035 virtual bool SetSink(uint32_t ssrc, |
1038 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | 1036 rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0; |
1039 // Gets quality stats for the channel. | 1037 // Gets quality stats for the channel. |
1040 virtual bool GetStats(VideoMediaInfo* info) = 0; | 1038 virtual bool GetStats(VideoMediaInfo* info) = 0; |
1041 }; | 1039 }; |
1042 | 1040 |
1043 enum DataMessageType { | 1041 enum DataMessageType { |
1044 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | 1042 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID |
1045 // values. | 1043 // values. |
1046 DMT_NONE = 0, | 1044 DMT_NONE = 0, |
1047 DMT_CONTROL = 1, | 1045 DMT_CONTROL = 1, |
1048 DMT_BINARY = 2, | 1046 DMT_BINARY = 2, |
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1158 // Signal when the media channel is ready to send the stream. Arguments are: | 1156 // Signal when the media channel is ready to send the stream. Arguments are: |
1159 // writable(bool) | 1157 // writable(bool) |
1160 sigslot::signal1<bool> SignalReadyToSend; | 1158 sigslot::signal1<bool> SignalReadyToSend; |
1161 // Signal for notifying that the remote side has closed the DataChannel. | 1159 // Signal for notifying that the remote side has closed the DataChannel. |
1162 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1160 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1163 }; | 1161 }; |
1164 | 1162 |
1165 } // namespace cricket | 1163 } // namespace cricket |
1166 | 1164 |
1167 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1165 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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