mirror of
https://github.com/livekit/livekit.git
synced 2026-10-15 15:30:14 +00:00
Raja SubramanianandClaude Fable 5.1
9b20c1da57
* sfu/buffer: stop allocating per ingested packet Buffer.Write allocated six objects for every RTP packet. They were the parsed rtp.Packet, a copy of it in patchExtPacket, and the flow state, whose address two rare log calls took. The rest were an AbsCaptureTime, present on about one packet per second, and the extension slice that pion grows with append. Unmarshal into a pooled scratch packet. Give ExtPacket its own backing rtp.Packet and an [8]rtp.Extension array, and copy the header into them. Patch the payload in place. If the abs-capture-time extension is present, parse it. Log a copy of the flow state. Per packet, Write plus ReadExtended. Opus: 6 allocs, 448 B -> 0. VP8: 6 allocs, 416 B -> 1 alloc, 32 B. Opus 240 -> 142 ns, VP8 269 -> 204 ns. Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com> * sfu/buffer: point extension payloads at the bucket copy on read Queued ExtPackets kept extension payload slices into the write input, which the transport reuses for the next packet. ReadExtended now parses the header again from the bucket copy, so extensions and CSRCs point into RawPacket like the payload does. The test writes every packet through one input buffer and checks the audio level of each packet read back. Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com> * sfu/buffer: test that a dropped padding packet keeps the shifted sequence number on read The bucket stores the shifted number, so re-parsing the header on read returns it. This test guards that. Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com> --------- Co-authored-by: Claude Fable 5.1 <noreply@anthropic.com>
617 lines
15 KiB
Go
617 lines
15 KiB
Go
// Copyright 2023 LiveKit, Inc.
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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package buffer
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import (
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"encoding/binary"
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"fmt"
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"math"
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"sync"
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"testing"
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"time"
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"github.com/pion/rtcp"
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"github.com/pion/rtp"
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"github.com/pion/sdp/v3"
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"github.com/pion/webrtc/v4"
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"github.com/stretchr/testify/require"
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"github.com/livekit/mediatransportutil/pkg/nack"
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act "github.com/livekit/livekit-server/pkg/sfu/rtpextension/abscapturetime"
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"github.com/livekit/livekit-server/pkg/sfu/utils"
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)
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var h265Codec = webrtc.RTPCodecParameters{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: "video/h265",
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ClockRate: 90000,
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RTCPFeedback: []webrtc.RTCPFeedback{{
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Type: "nack",
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}},
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},
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PayloadType: 116,
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}
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var vp8Codec = webrtc.RTPCodecParameters{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: "video/vp8",
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ClockRate: 90000,
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RTCPFeedback: []webrtc.RTCPFeedback{{
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Type: "nack",
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}},
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},
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PayloadType: 96,
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}
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var opusCodec = webrtc.RTPCodecParameters{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: "audio/opus",
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ClockRate: 48000,
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},
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PayloadType: 111,
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}
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func TestNack(t *testing.T) {
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t.Run("nack normal", func(t *testing.T) {
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buff := NewBuffer(123, 1, 1)
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buff.codecType = webrtc.RTPCodecTypeVideo
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require.NotNil(t, buff)
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var wg sync.WaitGroup
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// 5 tries
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wg.Add(5)
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buff.OnRtcpFeedback(func(fb []rtcp.Packet) {
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for _, pkt := range fb {
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switch p := pkt.(type) {
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case *rtcp.TransportLayerNack:
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if p.Nacks[0].PacketList()[0] == 1 && p.MediaSSRC == 123 {
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wg.Done()
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}
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}
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}
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})
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buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{vp8Codec},
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}, vp8Codec.RTPCodecCapability, 0)
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rtt := uint32(20)
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buff.nacker.SetRTT(rtt)
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for i := range 15 {
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if i == 1 {
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continue
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}
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if i < 14 {
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time.Sleep(time.Duration(float64(rtt)*math.Pow(nack.NackQueueParamsDefault.BackoffFactor, float64(i))+10) * time.Millisecond)
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} else {
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time.Sleep(500 * time.Millisecond) // even a long wait should not exceed max retries
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}
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pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: uint16(i),
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Timestamp: uint32(i),
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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b, err := pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(b)
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require.NoError(t, err)
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}
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wg.Wait()
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})
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t.Run("nack with seq wrap", func(t *testing.T) {
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buff := NewBuffer(123, 1, 1)
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buff.codecType = webrtc.RTPCodecTypeVideo
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require.NotNil(t, buff)
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var wg sync.WaitGroup
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expects := map[uint16]int{
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65534: 0,
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65535: 0,
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0: 0,
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1: 0,
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}
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wg.Add(5 * len(expects)) // retry 5 times
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buff.OnRtcpFeedback(func(fb []rtcp.Packet) {
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for _, pkt := range fb {
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switch p := pkt.(type) {
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case *rtcp.TransportLayerNack:
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if p.MediaSSRC == 123 {
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for _, v := range p.Nacks {
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v.Range(func(seq uint16) bool {
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if _, ok := expects[seq]; ok {
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wg.Done()
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} else {
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require.Fail(t, "unexpected nack seq ", seq)
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}
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return true
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})
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}
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}
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}
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}
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})
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buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{vp8Codec},
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}, vp8Codec.RTPCodecCapability, 0)
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rtt := uint32(30)
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buff.nacker.SetRTT(rtt)
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for i := range 15 {
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if i > 0 && i < 5 {
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continue
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}
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if i < 14 {
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time.Sleep(time.Duration(float64(rtt)*math.Pow(nack.NackQueueParamsDefault.BackoffFactor, float64(i))+10) * time.Millisecond)
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} else {
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time.Sleep(500 * time.Millisecond) // even a long wait should not exceed max retries
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}
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pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: uint16(i + 65533),
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Timestamp: uint32(i),
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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b, err := pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(b)
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require.NoError(t, err)
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}
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wg.Wait()
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})
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}
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func TestNewBuffer(t *testing.T) {
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tests := []struct {
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name string
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}{
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{
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name: "Must not be nil and add packets in sequence",
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},
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}
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for _, tt := range tests {
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t.Run(tt.name, func(t *testing.T) {
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var TestPackets = []*rtp.Packet{
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{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: 65533,
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SSRC: 123,
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},
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},
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{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: 65534,
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SSRC: 123,
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},
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Payload: []byte{1},
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},
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{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: 2,
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SSRC: 123,
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},
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},
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{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: 65535,
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SSRC: 123,
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},
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},
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}
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buff := NewBuffer(123, 1, 1)
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buff.codecType = webrtc.RTPCodecTypeVideo
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require.NotNil(t, buff)
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buff.OnRtcpFeedback(func(_ []rtcp.Packet) {})
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buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{vp8Codec},
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}, vp8Codec.RTPCodecCapability, 0)
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for _, p := range TestPackets {
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buf, _ := p.Marshal()
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_, _ = buff.Write(buf)
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}
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require.Equal(t, uint16(2), buff.rtpStats.HighestSequenceNumber())
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require.Equal(t, uint64(65536+2), buff.rtpStats.ExtendedHighestSequenceNumber())
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})
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}
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}
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func TestFractionLostReport(t *testing.T) {
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buff := NewBuffer(123, 1, 1)
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require.NotNil(t, buff)
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var wg sync.WaitGroup
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// with loss proxying
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wg.Add(1)
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buff.SetAudioLossProxying(true)
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buff.SetLastFractionLostReport(55)
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buff.OnRtcpFeedback(func(fb []rtcp.Packet) {
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for _, pkt := range fb {
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switch p := pkt.(type) {
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case *rtcp.ReceiverReport:
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for _, v := range p.Reports {
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require.EqualValues(t, 55, v.FractionLost)
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}
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wg.Done()
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}
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}
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})
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buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{opusCodec},
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}, opusCodec.RTPCodecCapability, 0)
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for i := range 15 {
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pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 111,
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SequenceNumber: uint16(i),
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Timestamp: uint32(i),
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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b, err := pkt.Marshal()
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require.NoError(t, err)
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if i == 1 {
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time.Sleep(1 * time.Second)
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}
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_, err = buff.Write(b)
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require.NoError(t, err)
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}
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wg.Wait()
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wg.Add(1)
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buff.SetAudioLossProxying(false)
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buff.OnRtcpFeedback(func(fb []rtcp.Packet) {
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for _, pkt := range fb {
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switch p := pkt.(type) {
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case *rtcp.ReceiverReport:
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for _, v := range p.Reports {
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require.EqualValues(t, 0, v.FractionLost)
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}
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wg.Done()
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}
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}
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})
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buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{opusCodec},
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}, opusCodec.RTPCodecCapability, 0)
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for i := range 15 {
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pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 111,
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SequenceNumber: uint16(i),
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Timestamp: uint32(i),
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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b, err := pkt.Marshal()
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require.NoError(t, err)
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if i == 1 {
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time.Sleep(1 * time.Second)
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}
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_, err = buff.Write(b)
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require.NoError(t, err)
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}
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wg.Wait()
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}
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func TestCodecChange(t *testing.T) {
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// codec change before bind
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buff := NewBuffer(123, 1, 1)
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require.NotNil(t, buff)
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changedCodec := make(chan webrtc.RTPCodecParameters, 1)
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buff.OnCodecChange(func(rp webrtc.RTPCodecParameters) {
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select {
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case changedCodec <- rp:
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default:
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t.Fatalf("codec change not consumed")
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}
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})
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buff.OnStreamRestart(func(reason string) {
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require.Equal(t, "codec-change", reason)
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// read once to clear pending restart
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var buf [1500]byte
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extPkt, err := buff.ReadExtended(buf[:])
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require.NoError(t, err)
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require.Nil(t, extPkt)
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})
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h265Pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 116,
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SequenceNumber: 1,
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Timestamp: 1,
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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buf, err := h265Pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(buf)
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require.NoError(t, err)
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select {
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case <-changedCodec:
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t.Fatalf("unexpected codec change")
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case <-time.After(100 * time.Millisecond):
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}
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// Bind sets up VP8 as expected codec,
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// packet written to the buffer above before Bind is H.265,
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// that should trigger a codec change and a stream restart
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// when the queued packets from Write before Bind get flushed
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buff.Bind(
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webrtc.RTPParameters{
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HeaderExtensions: nil,
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Codecs: []webrtc.RTPCodecParameters{vp8Codec, h265Codec},
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},
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vp8Codec.RTPCodecCapability,
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0,
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)
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select {
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case c := <-changedCodec:
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require.Equal(t, h265Codec, c)
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case <-time.After(1 * time.Second):
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t.Fatalf("expected codec change")
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}
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// second codec change - writing VP8 packet after Bind should trigger another codec change
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vp8Pkt := rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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PayloadType: 96,
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SequenceNumber: 3,
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Timestamp: 3,
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SSRC: 123,
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},
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Payload: []byte{0xff, 0xff, 0xff, 0xfd, 0xb4, 0x9f, 0x94, 0x1},
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}
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buf, err = vp8Pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(buf)
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require.NoError(t, err)
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select {
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case c := <-changedCodec:
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require.Equal(t, vp8Codec, c)
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case <-time.After(1 * time.Second):
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t.Fatalf("expected codec change")
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}
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fmt.Printf("done second codec change\n") // REMOVE
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// out of order pkts can't cause codec change
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// rewrite the VP8 packet to start the sequence after a stream restart
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_, err = buff.Write(buf)
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require.NoError(t, err)
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h265Pkt.SequenceNumber = 2
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h265Pkt.Timestamp = 2
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buf, err = h265Pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(buf)
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require.NoError(t, err)
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select {
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case <-changedCodec:
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t.Fatalf("unexpected codec change")
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case <-time.After(100 * time.Millisecond):
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}
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// unknown codec should not cause change even if it is in order
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h265Pkt.SequenceNumber = 4
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h265Pkt.Timestamp = 4
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h265Pkt.PayloadType = 117
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buf, err = h265Pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(buf)
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require.NoError(t, err)
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select {
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case <-changedCodec:
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t.Fatalf("unexpected codec change")
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case <-time.After(100 * time.Millisecond):
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}
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// an in-order packet should change codec again
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h265Pkt.SequenceNumber = 5
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h265Pkt.Timestamp = 5
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h265Pkt.PayloadType = 116
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buf, err = h265Pkt.Marshal()
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require.NoError(t, err)
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_, err = buff.Write(buf)
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require.NoError(t, err)
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select {
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case c := <-changedCodec:
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require.Equal(t, h265Codec, c)
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case <-time.After(1 * time.Second):
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t.Fatalf("expected codec change")
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}
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}
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// Write unmarshals into reused scratch and callers reuse the input buffer, so each ExtPacket
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// must keep its own header and extensions, and Write plus ReadExtended must not allocate per packet
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func TestIngestNoAlloc(t *testing.T) {
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testCases := []struct {
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name string
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codec webrtc.RTPCodecParameters
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payload []byte
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maxAllocs float64
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}{
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{"opus", opusCodec, make([]byte, 100), 0},
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// VP8 boxes the parsed descriptor into ExtPacket.Payload
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{"vp8", vp8Codec, []byte{0x90, 0xe0, 0x80, 0x01, 0x05, 0x00, 0x01}, 1},
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}
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for _, tc := range testCases {
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t.Run(tc.name, func(t *testing.T) {
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buff := NewBuffer(123, 100, 100)
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require.NoError(t, buff.Bind(webrtc.RTPParameters{
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HeaderExtensions: []webrtc.RTPHeaderExtensionParameter{
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{URI: sdp.AudioLevelURI, ID: 1},
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{URI: act.AbsCaptureTimeURI, ID: 2},
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},
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Codecs: []webrtc.RTPCodecParameters{tc.codec},
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}, tc.codec.RTPCodecCapability, 0))
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defer buff.Close()
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marshal := func(sn uint16, level byte) []byte {
|
|
p := rtp.Packet{
|
|
Header: rtp.Header{
|
|
Version: 2,
|
|
PayloadType: uint8(tc.codec.PayloadType),
|
|
SequenceNumber: sn,
|
|
Timestamp: uint32(sn) * 960,
|
|
SSRC: 123,
|
|
},
|
|
Payload: tc.payload,
|
|
}
|
|
require.NoError(t, p.SetExtension(1, []byte{level}))
|
|
require.NoError(t, p.SetExtension(3, []byte{0, 0}))
|
|
raw, err := p.Marshal()
|
|
require.NoError(t, err)
|
|
return raw
|
|
}
|
|
|
|
// one input buffer for all writes, like the transport read loop
|
|
input := make([]byte, 1500)
|
|
write := func(sn uint16, level byte) {
|
|
n := copy(input, marshal(sn, level))
|
|
_, err := buff.Write(input[:n])
|
|
require.NoError(t, err)
|
|
}
|
|
|
|
readBuf := make([]byte, 1500)
|
|
write(1, 10)
|
|
write(2, 20)
|
|
for _, want := range []struct {
|
|
sn uint16
|
|
level byte
|
|
}{{1, 10}, {2, 20}} {
|
|
ep, err := buff.ReadExtended(readBuf)
|
|
require.NoError(t, err)
|
|
require.Equal(t, want.sn, ep.Packet.SequenceNumber)
|
|
require.Equal(t, []byte{want.level}, ep.Packet.GetExtension(1))
|
|
require.Equal(t, tc.payload, ep.Packet.Payload)
|
|
require.Nil(t, ep.AbsCaptureTimeExt)
|
|
ReleaseExtPacket(ep)
|
|
}
|
|
|
|
raw := marshal(3, 30)
|
|
sn := uint16(3)
|
|
ingest := func() {
|
|
binary.BigEndian.PutUint16(raw[2:], sn)
|
|
binary.BigEndian.PutUint32(raw[4:], uint32(sn)*960)
|
|
sn++
|
|
_, err := buff.Write(raw)
|
|
require.NoError(t, err)
|
|
ep, err := buff.ReadExtended(readBuf)
|
|
require.NoError(t, err)
|
|
ReleaseExtPacket(ep)
|
|
}
|
|
allocs := testing.AllocsPerRun(1000, ingest)
|
|
if !utils.RaceEnabled {
|
|
require.LessOrEqual(t, allocs, tc.maxAllocs, "allocations per Write plus ReadExtended")
|
|
}
|
|
})
|
|
}
|
|
}
|
|
|
|
// a dropped padding only packet shifts later sequence numbers, ReadExtended must keep the shifted number
|
|
// in the header and in RawPacket
|
|
func TestIngestDroppedPaddingKeepsAdjustedSequenceNumber(t *testing.T) {
|
|
buff := NewBuffer(123, 100, 100)
|
|
require.NoError(t, buff.Bind(webrtc.RTPParameters{
|
|
Codecs: []webrtc.RTPCodecParameters{opusCodec},
|
|
}, opusCodec.RTPCodecCapability, 0))
|
|
defer buff.Close()
|
|
|
|
write := func(sn uint16, paddingOnly bool) {
|
|
p := rtp.Packet{
|
|
Header: rtp.Header{
|
|
Version: 2,
|
|
PayloadType: uint8(opusCodec.PayloadType),
|
|
SequenceNumber: sn,
|
|
Timestamp: uint32(sn) * 960,
|
|
SSRC: 123,
|
|
},
|
|
}
|
|
if paddingOnly {
|
|
p.Header.Padding = true
|
|
p.Header.PaddingSize = 20
|
|
} else {
|
|
p.Payload = make([]byte, 100)
|
|
}
|
|
raw, err := p.Marshal()
|
|
require.NoError(t, err)
|
|
_, err = buff.Write(raw)
|
|
require.NoError(t, err)
|
|
}
|
|
|
|
write(40, false)
|
|
write(41, true)
|
|
write(42, false)
|
|
|
|
readBuf := make([]byte, 1500)
|
|
for _, wantSN := range []uint16{40, 41} {
|
|
ep, err := buff.ReadExtended(readBuf)
|
|
require.NoError(t, err)
|
|
require.Equal(t, wantSN, ep.Packet.SequenceNumber)
|
|
require.Equal(t, wantSN, binary.BigEndian.Uint16(ep.RawPacket[2:]))
|
|
ReleaseExtPacket(ep)
|
|
}
|
|
}
|
|
|
|
func BenchmarkMemcpu(b *testing.B) {
|
|
buf := make([]byte, 1500*1500*10)
|
|
buf2 := make([]byte, 1500*1500*20)
|
|
|
|
for b.Loop() {
|
|
copy(buf2, buf)
|
|
}
|
|
}
|
|
|
|
func BenchmarkExtPacketFactory(b *testing.B) {
|
|
|
|
for b.Loop() {
|
|
extPkt := ExtPacketFactory.Get().(*ExtPacket)
|
|
*extPkt = ExtPacket{}
|
|
ExtPacketFactory.Put(extPkt)
|
|
}
|
|
}
|