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Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2013 Google Inc. 3 * Copyright 2013 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation 11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution. 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 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. 14 * derived from this software without specific prior written permission.
15 * 15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 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 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 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, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 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 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #include "talk/app/webrtc/sctputils.h" 28 #include "webrtc/api/sctputils.h"
29 29
30 #include "webrtc/base/buffer.h" 30 #include "webrtc/base/buffer.h"
31 #include "webrtc/base/bytebuffer.h" 31 #include "webrtc/base/bytebuffer.h"
32 #include "webrtc/base/logging.h" 32 #include "webrtc/base/logging.h"
33 33
34 namespace webrtc { 34 namespace webrtc {
35 35
36 // Format defined at 36 // Format defined at
37 // http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section 37 // http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section
38 38
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196 payload->SetData(buffer.Data(), buffer.Length()); 196 payload->SetData(buffer.Data(), buffer.Length());
197 return true; 197 return true;
198 } 198 }
199 199
200 void WriteDataChannelOpenAckMessage(rtc::Buffer* payload) { 200 void WriteDataChannelOpenAckMessage(rtc::Buffer* payload) {
201 rtc::ByteBuffer buffer(rtc::ByteBuffer::ORDER_NETWORK); 201 rtc::ByteBuffer buffer(rtc::ByteBuffer::ORDER_NETWORK);
202 buffer.WriteUInt8(DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE); 202 buffer.WriteUInt8(DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE);
203 payload->SetData(buffer.Data(), buffer.Length()); 203 payload->SetData(buffer.Data(), buffer.Length());
204 } 204 }
205 } // namespace webrtc 205 } // namespace webrtc
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