a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
127 lines
4.4 KiB
C++
127 lines
4.4 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 <ctype.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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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.size() > 0 &&
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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.size() > 0 &&
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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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PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
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uint16_t sequence_number)
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: PacketFeedback(-1,
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arrival_time_ms,
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kNoSendTime,
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sequence_number,
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0,
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0,
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0,
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PacedPacketInfo()) {}
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PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
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int64_t send_time_ms,
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uint16_t sequence_number,
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size_t payload_size,
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const PacedPacketInfo& pacing_info)
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: PacketFeedback(-1,
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arrival_time_ms,
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send_time_ms,
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sequence_number,
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payload_size,
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0,
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0,
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pacing_info) {}
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PacketFeedback::PacketFeedback(int64_t creation_time_ms,
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uint16_t sequence_number,
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size_t payload_size,
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uint16_t local_net_id,
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uint16_t remote_net_id,
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const PacedPacketInfo& pacing_info)
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: PacketFeedback(creation_time_ms,
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kNotReceived,
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kNoSendTime,
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sequence_number,
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payload_size,
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local_net_id,
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remote_net_id,
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pacing_info) {}
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PacketFeedback::PacketFeedback(int64_t creation_time_ms,
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int64_t arrival_time_ms,
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int64_t send_time_ms,
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uint16_t sequence_number,
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size_t payload_size,
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uint16_t local_net_id,
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uint16_t remote_net_id,
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const PacedPacketInfo& pacing_info)
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: creation_time_ms(creation_time_ms),
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arrival_time_ms(arrival_time_ms),
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send_time_ms(send_time_ms),
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sequence_number(sequence_number),
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long_sequence_number(0),
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payload_size(payload_size),
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unacknowledged_data(0),
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local_net_id(local_net_id),
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remote_net_id(remote_net_id),
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pacing_info(pacing_info),
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ssrc(0),
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rtp_sequence_number(0) {}
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PacketFeedback::PacketFeedback(const PacketFeedback&) = default;
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PacketFeedback& PacketFeedback::operator=(const PacketFeedback&) = default;
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PacketFeedback::~PacketFeedback() = default;
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bool PacketFeedback::operator==(const PacketFeedback& rhs) const {
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return arrival_time_ms == rhs.arrival_time_ms &&
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send_time_ms == rhs.send_time_ms &&
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sequence_number == rhs.sequence_number &&
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payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
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}
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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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