blob: 67255164f3ab7cc1076d41d91aeb52067311ac61 [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).
PacketSequencer(uint32_t media_ssrc,
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.
// Returns false if sequencing failed, which it can do for instance if the
// packet to squence is padding on the media ssrc, but the media is mid frame
// (the last marker bit is false).
bool 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 PupulateRtpState(RtpState& state) const;
uint16_t media_sequence_number() const { return media_sequence_number_; }
uint16_t rtx_sequence_number() const { return rtx_sequence_number_; }
void UpdateLastPacketState(const RtpPacketToSend& packet);
bool PopulatePaddingFields(RtpPacketToSend& packet);
const uint32_t media_ssrc_;
const 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_;
int64_t last_capture_time_ms_;
int64_t last_timestamp_time_ms_;
bool last_packet_marker_bit_;
} // namespace webrtc