
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
47 lines
1.7 KiB
C++
47 lines
1.7 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
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#include "webrtc/rtc_base/basictypes.h"
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namespace webrtc {
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namespace rtcp {
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class CommonHeader {
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public:
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CommonHeader() {}
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CommonHeader(const CommonHeader&) = default;
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CommonHeader& operator =(const CommonHeader&) = default;
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bool Parse(const uint8_t* buffer, size_t size_bytes);
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uint8_t type() const { return packet_type_; }
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// Depending on packet type same header field can be used either as count or
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// as feedback message type (fmt). Caller expected to know how it is used.
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uint8_t fmt() const { return count_or_format_; }
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uint8_t count() const { return count_or_format_; }
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size_t payload_size_bytes() const { return payload_size_; }
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const uint8_t* payload() const { return payload_; }
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// Returns pointer to the next RTCP packet in compound packet.
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const uint8_t* NextPacket() const {
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return payload_ + payload_size_ + padding_size_;
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}
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private:
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uint8_t packet_type_ = 0;
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uint8_t count_or_format_ = 0;
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uint8_t padding_size_ = 0;
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uint32_t payload_size_ = 0;
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const uint8_t* payload_ = nullptr;
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};
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} // namespace rtcp
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
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