blob: 0ae069dabceb1f04985a1618fd8c84c585997e12 [file] [log] [blame]
* 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/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
// Helper class used to assign RTP sequence numbers and populate some fields for
// padding packets based on the last sequenced packets.
// This class is not thread safe, the caller must provide that.
class PacketSequencer {
// If `require_marker_before_media_padding_` is true, padding packets on the
// media ssrc is not allowed unless the last sequenced media packet had the
// marker bit set (i.e. don't insert padding packets between the first and
// last packets of a video frame).
// Packets with unknown SSRCs will be ignored.
PacketSequencer(uint32_t media_ssrc,
absl::optional<uint32_t> rtx_ssrc,
bool require_marker_before_media_padding,
Clock* clock);
// Assigns sequence number, and in the case of non-RTX padding also timestamps
// and payload type.
void Sequence(RtpPacketToSend& packet);
void set_media_sequence_number(uint16_t sequence_number) {
media_sequence_number_ = sequence_number;
void set_rtx_sequence_number(uint16_t sequence_number) {
rtx_sequence_number_ = sequence_number;
void SetRtpState(const RtpState& state);
void PopulateRtpState(RtpState& state) const;
uint16_t media_sequence_number() const { return media_sequence_number_; }
uint16_t rtx_sequence_number() const { return rtx_sequence_number_; }
// Checks whether it is allowed to send padding on the media SSRC at this
// time, e.g. that we don't send padding in the middle of a video frame.
bool CanSendPaddingOnMediaSsrc() const;
void UpdateLastPacketState(const RtpPacketToSend& packet);
void PopulatePaddingFields(RtpPacketToSend& packet);
const uint32_t media_ssrc_;
const absl::optional<uint32_t> rtx_ssrc_;
const bool require_marker_before_media_padding_;
Clock* const clock_;
uint16_t media_sequence_number_;
uint16_t rtx_sequence_number_;
int8_t last_payload_type_;
uint32_t last_rtp_timestamp_;
Timestamp last_capture_time_ = Timestamp::MinusInfinity();
Timestamp last_timestamp_time_ = Timestamp::MinusInfinity();
bool last_packet_marker_bit_;
} // namespace webrtc