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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: Rebase. Created 4 years, 8 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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22 #include "webrtc/base/logging.h" 22 #include "webrtc/base/logging.h"
23 #include "webrtc/base/networkroute.h" 23 #include "webrtc/base/networkroute.h"
24 #include "webrtc/base/optional.h" 24 #include "webrtc/base/optional.h"
25 #include "webrtc/base/sigslot.h" 25 #include "webrtc/base/sigslot.h"
26 #include "webrtc/base/socket.h" 26 #include "webrtc/base/socket.h"
27 #include "webrtc/base/window.h" 27 #include "webrtc/base/window.h"
28 #include "webrtc/media/base/codec.h" 28 #include "webrtc/media/base/codec.h"
29 #include "webrtc/media/base/mediaconstants.h" 29 #include "webrtc/media/base/mediaconstants.h"
30 #include "webrtc/media/base/streamparams.h" 30 #include "webrtc/media/base/streamparams.h"
31 #include "webrtc/media/base/videosinkinterface.h" 31 #include "webrtc/media/base/videosinkinterface.h"
32 #include "webrtc/media/base/videosourceinterface.h"
32 // TODO(juberti): re-evaluate this include 33 // TODO(juberti): re-evaluate this include
33 #include "webrtc/pc/audiomonitor.h" 34 #include "webrtc/pc/audiomonitor.h"
34 35
35 namespace rtc { 36 namespace rtc {
36 class Buffer; 37 class Buffer;
37 class RateLimiter; 38 class RateLimiter;
38 class Timing; 39 class Timing;
39 } 40 }
40 41
41 namespace webrtc { 42 namespace webrtc {
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988 // Starts or stops transmission (and potentially capture) of local video. 989 // Starts or stops transmission (and potentially capture) of local video.
989 virtual bool SetSend(bool send) = 0; 990 virtual bool SetSend(bool send) = 0;
990 // Configure stream for sending. 991 // Configure stream for sending.
991 virtual bool SetVideoSend(uint32_t ssrc, 992 virtual bool SetVideoSend(uint32_t ssrc,
992 bool enable, 993 bool enable,
993 const VideoOptions* options) = 0; 994 const VideoOptions* options) = 0;
994 // Sets the sink object to be used for the specified stream. 995 // Sets the sink object to be used for the specified stream.
995 // If SSRC is 0, the renderer is used for the 'default' stream. 996 // If SSRC is 0, the renderer is used for the 'default' stream.
996 virtual bool SetSink(uint32_t ssrc, 997 virtual bool SetSink(uint32_t ssrc,
997 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; 998 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
998 // If |ssrc| is 0, replace the default capturer (engine capturer) with 999 // Register a source. The |ssrc| must correspond to a registered send stream.
999 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. 1000 virtual void SetSource(
1000 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; 1001 uint32_t ssrc,
1002 rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0;
1001 // Gets quality stats for the channel. 1003 // Gets quality stats for the channel.
1002 virtual bool GetStats(VideoMediaInfo* info) = 0; 1004 virtual bool GetStats(VideoMediaInfo* info) = 0;
1003 }; 1005 };
1004 1006
1005 enum DataMessageType { 1007 enum DataMessageType {
1006 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID 1008 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID
1007 // values. 1009 // values.
1008 DMT_NONE = 0, 1010 DMT_NONE = 0,
1009 DMT_CONTROL = 1, 1011 DMT_CONTROL = 1,
1010 DMT_BINARY = 2, 1012 DMT_BINARY = 2,
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1120 // Signal when the media channel is ready to send the stream. Arguments are: 1122 // Signal when the media channel is ready to send the stream. Arguments are:
1121 // writable(bool) 1123 // writable(bool)
1122 sigslot::signal1<bool> SignalReadyToSend; 1124 sigslot::signal1<bool> SignalReadyToSend;
1123 // Signal for notifying that the remote side has closed the DataChannel. 1125 // Signal for notifying that the remote side has closed the DataChannel.
1124 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1126 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1125 }; 1127 };
1126 1128
1127 } // namespace cricket 1129 } // namespace cricket
1128 1130
1129 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1131 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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