blob: d743f52f7e6df3ea268321200c33758516f29eeb [file] [log] [blame]
/*
* 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 "modules/rtp_rtcp/source/rtp_packet.h"
#include <ctype.h>
#include <string.h>
#include <algorithm>
#include <type_traits>
#include "api/array_view.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<const char> name) {
return (name.size() <= kMaxSize && name.size() > 0 &&
std::all_of(name.data(), name.data() + name.size(), IsTokenChar));
}
bool StreamId::IsLegalRsidName(rtc::ArrayView<const char> 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<StreamId>::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;
PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
uint16_t sequence_number)
: PacketFeedback(-1,
arrival_time_ms,
kNoSendTime,
sequence_number,
0,
0,
0,
PacedPacketInfo()) {}
PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
const PacedPacketInfo& pacing_info)
: PacketFeedback(-1,
arrival_time_ms,
send_time_ms,
sequence_number,
payload_size,
0,
0,
pacing_info) {}
PacketFeedback::PacketFeedback(int64_t creation_time_ms,
uint16_t sequence_number,
size_t payload_size,
uint16_t local_net_id,
uint16_t remote_net_id,
const PacedPacketInfo& pacing_info)
: PacketFeedback(creation_time_ms,
kNotReceived,
kNoSendTime,
sequence_number,
payload_size,
local_net_id,
remote_net_id,
pacing_info) {}
PacketFeedback::PacketFeedback(int64_t creation_time_ms,
int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
uint16_t local_net_id,
uint16_t remote_net_id,
const PacedPacketInfo& pacing_info)
: creation_time_ms(creation_time_ms),
arrival_time_ms(arrival_time_ms),
send_time_ms(send_time_ms),
sequence_number(sequence_number),
payload_size(payload_size),
unacknowledged_data(0),
local_net_id(local_net_id),
remote_net_id(remote_net_id),
pacing_info(pacing_info) {}
PacketFeedback::PacketFeedback(const PacketFeedback&) = default;
PacketFeedback& PacketFeedback::operator=(const PacketFeedback&) = default;
PacketFeedback::~PacketFeedback() = default;
bool PacketFeedback::operator==(const PacketFeedback& rhs) const {
return arrival_time_ms == rhs.arrival_time_ms &&
send_time_ms == rhs.send_time_ms &&
sequence_number == rhs.sequence_number &&
payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
}
void RtpPacketCounter::AddPacket(const RtpPacket& packet) {
++packets;
header_bytes += packet.headers_size();
padding_bytes += packet.padding_size();
payload_bytes += packet.payload_size();
}
} // namespace webrtc