| OLD | NEW |
| 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 Loading... |
| 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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