| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| #include <string.h> |
| |
| #include <type_traits> |
| |
| #include "absl/algorithm/container.h" |
| #include "api/array_view.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| constexpr size_t kMidRsidMaxSize = 16; |
| |
| // Check if passed character is a "token-char" from RFC 4566. |
| // https://datatracker.ietf.org/doc/html/rfc4566#section-9 |
| // token-char = %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39 |
| // / %x41-5A / %x5E-7E |
| bool IsTokenChar(char ch) { |
| return ch == 0x21 || (ch >= 0x23 && ch <= 0x27) || ch == 0x2a || ch == 0x2b || |
| ch == 0x2d || ch == 0x2e || (ch >= 0x30 && ch <= 0x39) || |
| (ch >= 0x41 && ch <= 0x5a) || (ch >= 0x5e && ch <= 0x7e); |
| } |
| } // namespace |
| |
| bool IsLegalMidName(absl::string_view name) { |
| return (name.size() <= kMidRsidMaxSize && !name.empty() && |
| absl::c_all_of(name, IsTokenChar)); |
| } |
| |
| bool IsLegalRsidName(absl::string_view name) { |
| return (name.size() <= kMidRsidMaxSize && !name.empty() && |
| absl::c_all_of(name, isalnum)); |
| } |
| |
| StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {} |
| |
| RtpPacketCounter::RtpPacketCounter(const RtpPacket& packet) |
| : header_bytes(packet.headers_size()), |
| payload_bytes(packet.payload_size()), |
| padding_bytes(packet.padding_size()), |
| packets(1) {} |
| |
| RtpPacketCounter::RtpPacketCounter(const RtpPacketToSend& packet_to_send) |
| : RtpPacketCounter(static_cast<const RtpPacket&>(packet_to_send)) { |
| total_packet_delay = |
| packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero()); |
| } |
| |
| void RtpPacketCounter::AddPacket(const RtpPacket& packet) { |
| ++packets; |
| header_bytes += packet.headers_size(); |
| padding_bytes += packet.padding_size(); |
| payload_bytes += packet.payload_size(); |
| } |
| |
| void RtpPacketCounter::AddPacket(const RtpPacketToSend& packet_to_send) { |
| AddPacket(static_cast<const RtpPacket&>(packet_to_send)); |
| total_packet_delay += |
| packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero()); |
| } |
| |
| } // namespace webrtc |