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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: Work-in-progress, after applying 1790633002. 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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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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