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/*
* Copyright (c) 2021 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/source/packet_sequencer.h"
#include <cstddef>
#include <cstdint>
#include <optional>
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/random.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace {
// RED header is first byte of payload, if present.
constexpr size_t kRedForFecHeaderLength = 1;
// Timestamps use a 90kHz clock.
constexpr uint32_t kTimestampTicksPerMs = 90;
} // namespace
PacketSequencer::PacketSequencer(uint32_t media_ssrc,
std::optional<uint32_t> rtx_ssrc,
bool require_marker_before_media_padding,
Clock* clock)
: media_ssrc_(media_ssrc),
rtx_ssrc_(rtx_ssrc),
require_marker_before_media_padding_(require_marker_before_media_padding),
clock_(clock),
media_sequence_number_(0),
rtx_sequence_number_(0),
last_payload_type_(-1),
last_rtp_timestamp_(0),
last_packet_marker_bit_(false) {
Random random(clock_->TimeInMicroseconds());
// Random start, 16 bits. Upper half of range is avoided in order to prevent
// SRTP wraparound issues during startup. See this unit test for details:
// SrtpSessionTest.ProtectUnprotectWrapAroundRocMismatch
// Sequence number 0 is avoided for historical reasons, presumably to avoid
// debugability or test usage conflicts.
constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff; // 2^15 - 1.
media_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
rtx_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
}
void PacketSequencer::Sequence(RtpPacketToSend& packet) {
if (packet.Ssrc() == media_ssrc_) {
if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
// Retransmission of an already sequenced packet, ignore.
return;
} else if (packet.packet_type() == RtpPacketMediaType::kPadding) {
PopulatePaddingFields(packet);
}
packet.SetSequenceNumber(media_sequence_number_++);
if (packet.packet_type() != RtpPacketMediaType::kPadding) {
UpdateLastPacketState(packet);
}
} else if (packet.Ssrc() == rtx_ssrc_) {
if (packet.packet_type() == RtpPacketMediaType::kPadding) {
PopulatePaddingFields(packet);
}
packet.SetSequenceNumber(rtx_sequence_number_++);
} else {
RTC_DCHECK_NOTREACHED() << "Unexpected ssrc " << packet.Ssrc();
}
}
void PacketSequencer::SetRtpState(const RtpState& state) {
media_sequence_number_ = state.sequence_number;
last_rtp_timestamp_ = state.timestamp;
last_capture_time_ = state.capture_time;
last_timestamp_time_ = state.last_timestamp_time;
}
void PacketSequencer::PopulateRtpState(RtpState& state) const {
state.sequence_number = media_sequence_number_;
state.timestamp = last_rtp_timestamp_;
state.capture_time = last_capture_time_;
state.last_timestamp_time = last_timestamp_time_;
}
void PacketSequencer::UpdateLastPacketState(const RtpPacketToSend& packet) {
// Remember marker bit to determine if padding can be inserted with
// sequence number following `packet`.
last_packet_marker_bit_ = packet.Marker();
// Remember media payload type to use in the padding packet if rtx is
// disabled.
if (packet.is_red()) {
RTC_DCHECK_GE(packet.payload_size(), kRedForFecHeaderLength);
last_payload_type_ = packet.PayloadBuffer()[0];
} else {
last_payload_type_ = packet.PayloadType();
}
// Save timestamps to generate timestamp field and extensions for the padding.
last_rtp_timestamp_ = packet.Timestamp();
last_timestamp_time_ = clock_->CurrentTime();
last_capture_time_ = packet.capture_time();
}
void PacketSequencer::PopulatePaddingFields(RtpPacketToSend& packet) {
if (packet.Ssrc() == media_ssrc_) {
RTC_DCHECK(CanSendPaddingOnMediaSsrc());
packet.SetTimestamp(last_rtp_timestamp_);
packet.set_capture_time(last_capture_time_);
packet.SetPayloadType(last_payload_type_);
return;
}
RTC_DCHECK(packet.Ssrc() == rtx_ssrc_);
if (packet.payload_size() > 0) {
// This is payload padding packet, don't update timestamp fields.
return;
}
packet.SetTimestamp(last_rtp_timestamp_);
packet.set_capture_time(last_capture_time_);
// Only change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
if (last_timestamp_time_ > Timestamp::Zero()) {
TimeDelta since_last_media = clock_->CurrentTime() - last_timestamp_time_;
packet.SetTimestamp(packet.Timestamp() +
since_last_media.ms() * kTimestampTicksPerMs);
if (packet.capture_time() > Timestamp::Zero()) {
packet.set_capture_time(packet.capture_time() + since_last_media);
}
}
}
bool PacketSequencer::CanSendPaddingOnMediaSsrc() const {
if (last_payload_type_ == -1) {
return false;
}
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (require_marker_before_media_padding_ && !last_packet_marker_bit_) {
return false;
}
return true;
}
} // namespace webrtc
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