Files
livekit/pkg/sfu/testutils/data.go
Raja Subramanian acbd4ea104 Handle cases of long mute/rollover of time stamp. (#2842)
* Handle cases of long mute/rollover of time stamp.

There are cases where the track is muted for long enough for timestamp
roll over to happen. There are no packets in that window (typically
there should be black frames (for video) or silence (for audio)). But,
maybe the pause based implementation of mute is causing this.

Anyhow, use time since last packet to gauge how much roll over should
have happened and use that to update time stamp. There will be really
edge cases where this could also fail (for e. g. packet time is affected
by propagation delay, so it could theoretically happen that mute/unmute
+ packet reception could happen exactly around that rollover point and
  miscalculate, but should be rare).

As this happen per packet on receive side, changing time to `UnixNano()`
to make it more efficient to check this.

* spelling

* tests

* test util

* tests
2024-07-08 11:07:20 +05:30

104 lines
2.6 KiB
Go

// Copyright 2023 LiveKit, Inc.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
// http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.
package testutils
import (
"time"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"github.com/livekit/livekit-server/pkg/sfu/buffer"
)
// -----------------------------------------------------------
type TestExtPacketParams struct {
SetMarker bool
IsKeyFrame bool
PayloadType uint8
SequenceNumber uint16
SNCycles int
Timestamp uint32
TSCycles int
SSRC uint32
PayloadSize int
PaddingSize byte
ArrivalTime time.Time
VideoLayer buffer.VideoLayer
}
// -----------------------------------------------------------
func GetTestExtPacket(params *TestExtPacketParams) (*buffer.ExtPacket, error) {
packet := rtp.Packet{
Header: rtp.Header{
Version: 2,
Padding: params.PaddingSize != 0,
Marker: params.SetMarker,
PayloadType: params.PayloadType,
SequenceNumber: params.SequenceNumber,
Timestamp: params.Timestamp,
SSRC: params.SSRC,
},
Payload: make([]byte, params.PayloadSize),
PaddingSize: params.PaddingSize,
}
raw, err := packet.Marshal()
if err != nil {
return nil, err
}
ep := &buffer.ExtPacket{
VideoLayer: params.VideoLayer,
ExtSequenceNumber: uint64(params.SNCycles<<16) + uint64(params.SequenceNumber),
ExtTimestamp: uint64(params.TSCycles<<32) + uint64(params.Timestamp),
Arrival: params.ArrivalTime.UnixNano(),
Packet: &packet,
KeyFrame: params.IsKeyFrame,
RawPacket: raw,
}
return ep, nil
}
// --------------------------------------
func GetTestExtPacketVP8(params *TestExtPacketParams, vp8 *buffer.VP8) (*buffer.ExtPacket, error) {
ep, err := GetTestExtPacket(params)
if err != nil {
return nil, err
}
ep.KeyFrame = vp8.IsKeyFrame
ep.Payload = *vp8
return ep, nil
}
// --------------------------------------
var TestVP8Codec = webrtc.RTPCodecCapability{
MimeType: "video/vp8",
ClockRate: 90000,
}
var TestOpusCodec = webrtc.RTPCodecCapability{
MimeType: "audio/opus",
ClockRate: 48000,
}
// --------------------------------------