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-rw-r--r--third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc75
1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
new file mode 100644
index 0000000000..e4aec93696
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
@@ -0,0 +1,75 @@
+/*
+ * 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