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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 "webrtc/engine_configurations.h" | |
24 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" | 23 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" |
25 // TODO(tlegrand): Consider removing usage of gtest. | 24 // TODO(tlegrand): Consider removing usage of gtest. |
26 #include "webrtc/test/gtest.h" | 25 #include "webrtc/test/gtest.h" |
| 26 #include "webrtc/voice_engine_configurations.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]; |
35 rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) | | 35 rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) | |
36 (static_cast<uint32_t>(rtpHeader[5]) << 16) | | 36 (static_cast<uint32_t>(rtpHeader[5]) << 16) | |
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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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