/* * 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" namespace webrtc { StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {} constexpr size_t StreamId::kMaxSize; // Check if passed character is a "token-char" from RFC 4566. static 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); } bool StreamId::IsLegalMidName(rtc::ArrayView name) { return (name.size() <= kMaxSize && name.size() > 0 && std::all_of(name.data(), name.data() + name.size(), IsTokenChar)); } bool StreamId::IsLegalRsidName(rtc::ArrayView name) { return (name.size() <= kMaxSize && name.size() > 0 && std::all_of(name.data(), name.data() + name.size(), isalnum)); } void StreamId::Set(const char* data, size_t size) { // If |data| contains \0, the stream id size might become less than |size|. RTC_CHECK_LE(size, kMaxSize); memcpy(value_, data, size); if (size < kMaxSize) value_[size] = 0; } // StreamId is used as member of RTPHeader that is sometimes copied with memcpy // and thus assume trivial destructibility. static_assert(std::is_trivially_destructible::value, ""); PayloadUnion::PayloadUnion(const AudioPayload& payload) : payload_(payload) {} PayloadUnion::PayloadUnion(const VideoPayload& payload) : payload_(payload) {} PayloadUnion::PayloadUnion(const PayloadUnion&) = default; PayloadUnion::PayloadUnion(PayloadUnion&&) = default; PayloadUnion::~PayloadUnion() = default; PayloadUnion& PayloadUnion::operator=(const PayloadUnion&) = default; PayloadUnion& PayloadUnion::operator=(PayloadUnion&&) = default; } // namespace webrtc