mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
77ee8542dd
This CL refactors RtpSender and extracts handling of sequence number assignment and timestamping of padding packets in a separate helper class. This is in preparation for allowing deferred sequencing to after the pacing stage. Bug: webrtc:11340 Change-Id: I5f8c67f3bb90780b3bdd24afa6ae28dbe9d839a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208401 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33316}
75 lines
2.7 KiB
C++
75 lines
2.7 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
|
|
#define MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
|
|
|
|
#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 {
|
|
public:
|
|
// 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_; }
|
|
|
|
private:
|
|
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
|
|
|
|
#endif // MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
|