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https://github.com/mollyim/webrtc.git
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e66b83f8ad
Bug: None Change-Id: I451bb2c1f175a77aefbc8363009bf35a769fe941 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264442 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37037}
62 lines
2 KiB
C++
62 lines
2 KiB
C++
/*
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* Copyright (c) 2017 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 "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include <string.h>
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#include <type_traits>
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#include "absl/algorithm/container.h"
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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namespace webrtc {
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namespace {
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constexpr size_t kMidRsidMaxSize = 16;
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// Check if passed character is a "token-char" from RFC 4566.
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// https://datatracker.ietf.org/doc/html/rfc4566#section-9
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// token-char = %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
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// / %x41-5A / %x5E-7E
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bool IsTokenChar(char ch) {
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return ch == 0x21 || (ch >= 0x23 && ch <= 0x27) || ch == 0x2a || ch == 0x2b ||
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ch == 0x2d || ch == 0x2e || (ch >= 0x30 && ch <= 0x39) ||
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(ch >= 0x41 && ch <= 0x5a) || (ch >= 0x5e && ch <= 0x7e);
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}
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} // namespace
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bool IsLegalMidName(absl::string_view name) {
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return (name.size() <= kMidRsidMaxSize && !name.empty() &&
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absl::c_all_of(name, IsTokenChar));
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}
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bool IsLegalRsidName(absl::string_view name) {
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return (name.size() <= kMidRsidMaxSize && !name.empty() &&
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absl::c_all_of(name, isalnum));
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}
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StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
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RtpPacketCounter::RtpPacketCounter(const RtpPacket& packet)
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: header_bytes(packet.headers_size()),
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payload_bytes(packet.payload_size()),
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padding_bytes(packet.padding_size()),
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packets(1) {}
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void RtpPacketCounter::AddPacket(const RtpPacket& packet) {
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++packets;
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header_bytes += packet.headers_size();
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padding_bytes += packet.padding_size();
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payload_bytes += packet.payload_size();
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}
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} // namespace webrtc
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