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|    1 /* |    1 /* | 
|    2  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |    2  *  Copyright (c) 2016 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 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ |   10 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ | 
|   11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ |   11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ | 
|   12  |   12  | 
|   13 #include <vector> |   13 #include <vector> | 
|   14  |   14  | 
 |   15 #include "webrtc/api/array_view.h" | 
|   15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |   16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 
|   16 #include "webrtc/rtc_base/array_view.h" |  | 
|   17 #include "webrtc/rtc_base/basictypes.h" |   17 #include "webrtc/rtc_base/basictypes.h" | 
|   18 #include "webrtc/rtc_base/copyonwritebuffer.h" |   18 #include "webrtc/rtc_base/copyonwritebuffer.h" | 
|   19  |   19  | 
|   20 namespace webrtc { |   20 namespace webrtc { | 
|   21 struct RTPHeader; |   21 struct RTPHeader; | 
|   22 class RtpHeaderExtensionMap; |   22 class RtpHeaderExtensionMap; | 
|   23 class Random; |   23 class Random; | 
|   24  |   24  | 
|   25 namespace rtp { |   25 namespace rtp { | 
|   26 class Packet { |   26 class Packet { | 
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|  197   auto buffer = AllocateExtension(Extension::kId, Extension::kValueSizeBytes); |  197   auto buffer = AllocateExtension(Extension::kId, Extension::kValueSizeBytes); | 
|  198   if (buffer.empty()) |  198   if (buffer.empty()) | 
|  199     return false; |  199     return false; | 
|  200   memset(buffer.data(), 0, Extension::kValueSizeBytes); |  200   memset(buffer.data(), 0, Extension::kValueSizeBytes); | 
|  201   return true; |  201   return true; | 
|  202 } |  202 } | 
|  203 }  // namespace rtp |  203 }  // namespace rtp | 
|  204 }  // namespace webrtc |  204 }  // namespace webrtc | 
|  205  |  205  | 
|  206 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ |  206 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_ | 
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