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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}
86 lines
3.1 KiB
C++
86 lines
3.1 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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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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#include "webrtc/rtc_base/logging.h"
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namespace webrtc {
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namespace rtcp {
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// 0 1 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 0 |V=2|P| C/F |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 1 | Packet Type |
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// ----------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 2 | length |
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// --------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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// Common header for all RTCP packets, 4 octets.
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bool CommonHeader::Parse(const uint8_t* buffer, size_t size_bytes) {
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const size_t kHeaderSizeBytes = 4;
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const uint8_t kVersion = 2;
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if (size_bytes < kHeaderSizeBytes) {
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LOG(LS_WARNING) << "Too little data (" << size_bytes << " byte"
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<< (size_bytes != 1 ? "s" : "")
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<< ") remaining in buffer to parse RTCP header (4 bytes).";
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return false;
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}
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uint8_t version = buffer[0] >> 6;
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if (version != kVersion) {
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LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
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<< static_cast<int>(kVersion) << " but was "
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<< static_cast<int>(version);
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return false;
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}
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bool has_padding = (buffer[0] & 0x20) != 0;
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count_or_format_ = buffer[0] & 0x1F;
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packet_type_ = buffer[1];
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payload_size_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) * 4;
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payload_ = buffer + kHeaderSizeBytes;
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padding_size_ = 0;
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if (size_bytes < kHeaderSizeBytes + payload_size_) {
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LOG(LS_WARNING) << "Buffer too small (" << size_bytes
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<< " bytes) to fit an RtcpPacket with a header and "
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<< payload_size_ << " bytes.";
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return false;
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}
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if (has_padding) {
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if (payload_size_ == 0) {
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LOG(LS_WARNING) << "Invalid RTCP header: Padding bit set but 0 payload "
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"size specified.";
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return false;
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}
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padding_size_ = payload_[payload_size_ - 1];
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if (padding_size_ == 0) {
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LOG(LS_WARNING) << "Invalid RTCP header: Padding bit set but 0 padding "
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"size specified.";
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return false;
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}
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if (padding_size_ > payload_size_) {
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LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
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<< padding_size_ << ") for a packet payload size of "
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<< payload_size_ << " bytes.";
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return false;
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}
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payload_size_ -= padding_size_;
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}
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return true;
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}
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} // namespace rtcp
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} // namespace webrtc
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