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Side by Side Diff: webrtc/modules/audio_coding/main/test/RTPFile.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
11 #include "RTPFile.h" 11 #include "RTPFile.h"
12 12
13 #include <stdlib.h> 13 #include <stdlib.h>
14 #include <limits> 14 #include <limits>
15 15
16 #ifdef WIN32 16 #ifdef WIN32
17 # include <Winsock2.h> 17 # include <Winsock2.h>
18 #else 18 #else
19 # include <arpa/inet.h> 19 # include <arpa/inet.h>
20 #endif 20 #endif
21 21
22 #include "audio_coding_module.h" 22 #include "audio_coding_module.h"
23 #include "engine_configurations.h" 23 #include "engine_configurations.h"
24 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" 24 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
25 // TODO(tlegrand): Consider removing usage of gtest. 25 // TODO(tlegrand): Consider removing usage of gtest.
26 #include "testing/gtest/include/gtest/gtest.h" 26 #include "testing/gtest/include/gtest/gtest.h"
27 27
28 namespace webrtc { 28 namespace webrtc {
29 29
30 void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, 30 void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
31 const uint8_t* rtpHeader) { 31 const uint8_t* rtpHeader) {
32 rtpInfo->header.payloadType = rtpHeader[1]; 32 rtpInfo->header.payloadType = rtpHeader[1];
33 rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) | 33 rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
34 rtpHeader[3]; 34 rtpHeader[3];
(...skipping 183 matching lines...) Expand 10 before | Expand all | Expand 10 after
218 } 218 }
219 if (payloadSize < static_cast<size_t>((lengthBytes - 20))) { 219 if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
220 return 0; 220 return 0;
221 } 221 }
222 lengthBytes -= 20; 222 lengthBytes -= 20;
223 EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile)); 223 EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
224 return lengthBytes; 224 return lengthBytes;
225 } 225 }
226 226
227 } // namespace webrtc 227 } // namespace webrtc
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