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Side by Side Diff: webrtc/media/sctp/sctpdataengine.h

Issue 1587193006: Move talk/media to webrtc/media (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Disable sign-compare warning on Win Clang Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * Copyright 2012 The WebRTC Project Authors. All rights reserved.
3 * Copyright 2012 Google Inc. and Robin Seggelmann
4 * 3 *
5 * Redistribution and use in source and binary forms, with or without 4 * Use of this source code is governed by a BSD-style license
6 * modification, are permitted provided that the following conditions are met: 5 * that can be found in the LICENSE file in the root of the source
7 * 6 * tree. An additional intellectual property rights grant can be found
8 * 1. Redistributions of source code must retain the above copyright notice, 7 * in the file PATENTS. All contributing project authors may
9 * this list of conditions and the following disclaimer. 8 * be found in the AUTHORS file in the root of the source tree.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 9 */
27 10
28 #ifndef TALK_MEDIA_SCTP_SCTPDATAENGINE_H_ 11 #ifndef WEBRTC_MEDIA_SCTP_SCTPDATAENGINE_H_
29 #define TALK_MEDIA_SCTP_SCTPDATAENGINE_H_ 12 #define WEBRTC_MEDIA_SCTP_SCTPDATAENGINE_H_
30 13
31 #include <errno.h> 14 #include <errno.h>
32 #include <string> 15 #include <string>
33 #include <vector> 16 #include <vector>
34 17
35 namespace cricket { 18 namespace cricket {
36 // Some ERRNO values get re-#defined to WSA* equivalents in some talk/ 19 // Some ERRNO values get re-#defined to WSA* equivalents in some talk/
37 // headers. We save the original ones in an enum. 20 // headers. We save the original ones in an enum.
38 enum PreservedErrno { 21 enum PreservedErrno {
39 SCTP_EINPROGRESS = EINPROGRESS, 22 SCTP_EINPROGRESS = EINPROGRESS,
40 SCTP_EWOULDBLOCK = EWOULDBLOCK 23 SCTP_EWOULDBLOCK = EWOULDBLOCK
41 }; 24 };
42 } // namespace cricket 25 } // namespace cricket
43 26
44 #include "talk/media/base/codec.h"
45 #include "talk/media/base/mediachannel.h"
46 #include "talk/media/base/mediaengine.h"
47 #include "webrtc/base/buffer.h" 27 #include "webrtc/base/buffer.h"
48 #include "webrtc/base/scoped_ptr.h" 28 #include "webrtc/base/scoped_ptr.h"
29 #include "webrtc/media/base/codec.h"
30 #include "webrtc/media/base/mediachannel.h"
31 #include "webrtc/media/base/mediaengine.h"
49 32
50 // Defined by "usrsctplib/usrsctp.h" 33 // Defined by "usrsctplib/usrsctp.h"
51 struct sockaddr_conn; 34 struct sockaddr_conn;
52 struct sctp_assoc_change; 35 struct sctp_assoc_change;
53 struct sctp_stream_reset_event; 36 struct sctp_stream_reset_event;
54 // Defined by <sys/socket.h> 37 // Defined by <sys/socket.h>
55 struct socket; 38 struct socket;
56 namespace cricket { 39 namespace cricket {
57 // The highest stream ID (Sid) that SCTP allows, and the number of streams we 40 // The highest stream ID (Sid) that SCTP allows, and the number of streams we
58 // tell SCTP we're going to use. 41 // tell SCTP we're going to use.
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246 StreamSet open_streams_; 229 StreamSet open_streams_;
247 StreamSet queued_reset_streams_; 230 StreamSet queued_reset_streams_;
248 StreamSet sent_reset_streams_; 231 StreamSet sent_reset_streams_;
249 232
250 // A human-readable name for debugging messages. 233 // A human-readable name for debugging messages.
251 std::string debug_name_; 234 std::string debug_name_;
252 }; 235 };
253 236
254 } // namespace cricket 237 } // namespace cricket
255 238
256 #endif // TALK_MEDIA_SCTP_SCTPDATAENGINE_H_ 239 #endif // WEBRTC_MEDIA_SCTP_SCTPDATAENGINE_H_
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