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Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 1766653002: Replace SetCapturer and SetCaptureDevice by SetSource. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased. Created 4 years, 9 months ago
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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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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
pbos-webrtc 2016/03/31 14:14:31 reformat comment (send fits on previous line)
nisse-webrtc 2016/04/01 09:35:39 Done. You use fill-column 80?
pbos-webrtc 2016/04/01 10:55:28 Not sure what that is in vim speak, we have 80 cha
997 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. 998 // send stream.
998 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; 999 virtual void SetSource(
1000 uint32_t ssrc,
1001 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0;
999 // Gets quality stats for the channel. 1002 // Gets quality stats for the channel.
1000 virtual bool GetStats(VideoMediaInfo* info) = 0; 1003 virtual bool GetStats(VideoMediaInfo* info) = 0;
1001 }; 1004 };
1002 1005
1003 enum DataMessageType { 1006 enum DataMessageType {
1004 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID 1007 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID
1005 // values. 1008 // values.
1006 DMT_NONE = 0, 1009 DMT_NONE = 0,
1007 DMT_CONTROL = 1, 1010 DMT_CONTROL = 1,
1008 DMT_BINARY = 2, 1011 DMT_BINARY = 2,
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1115 // Signal when the media channel is ready to send the stream. Arguments are: 1118 // Signal when the media channel is ready to send the stream. Arguments are:
1116 // writable(bool) 1119 // writable(bool)
1117 sigslot::signal1<bool> SignalReadyToSend; 1120 sigslot::signal1<bool> SignalReadyToSend;
1118 // Signal for notifying that the remote side has closed the DataChannel. 1121 // Signal for notifying that the remote side has closed the DataChannel.
1119 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1122 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1120 }; 1123 };
1121 1124
1122 } // namespace cricket 1125 } // namespace cricket
1123 1126
1124 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1127 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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