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Buffer size config for video and audio. (#2498)

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* Buffer size config for video and audio.

There was only one buffer size in config.
In upstream, config value was used for video.
Audio used a hard coded value of 200 packets.

But, in the down stream sequencer, the config value was used for both
video and audio. So, if video was set up for high bit rate (deep
buffers), audio sequencer ended up using a lot of memory too in
sequencer.

Split config to be able to control that and also not hard code audio.

Another optimisation here would be to not instantiate sequencer unkess
NACK is negotiated.

* deprecate packet_buffer_size
This commit is contained in:
Raja Subramanian
2024-02-21 22:58:56 +05:30
committed by GitHub
parent bd7950f56c
commit 6895eff496
7 changed files with 34 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files
config-sample.yaml
+4 -3
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@@ -94,8 +94,10 @@ rtc:
# allow_pause: true
# # allows automatic connection fallback to TCP and TURN/TLS (if configured) when UDP has been unstable, default true
# allow_tcp_fallback: true
# # number of packets to buffer in the SFU, defaults to 500
# packet_buffer_size: 500
# # number of packets to buffer in the SFU for video, defaults to 500
# packet_buffer_size_video: 500
# # number of packets to buffer in the SFU for audio, defaults to 200
# packet_buffer_size_audio: 200
# # minimum amount of time between pli/fir rtcp packets being sent to an individual
# # producer. Increasing these times can lead to longer black screens when new participants join,
# # while reducing them can lead to higher stream bitrate.
@@ -309,4 +311,3 @@ keys:
# # value less or equal than 0 means no limit.
# subscription_limit_video: 0
# subscription_limit_audio: 0
pkg/config/config.go
+10 -4
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@@ -76,7 +76,7 @@ type Config struct {
Region string `yaml:"region,omitempty"`
SignalRelay SignalRelayConfig `yaml:"signal_relay,omitempty"`
PSRPC rpc.PSRPCConfig `yaml:"psrpc,omitempty"`
// LogLevel is deprecated
// Deprecated: LogLevel is deprecated
LogLevel string `yaml:"log_level,omitempty"`
Logging LoggingConfig `yaml:"logging,omitempty"`
Limit LimitConfig `yaml:"limit,omitempty"`
@@ -91,8 +91,12 @@ type RTCConfig struct {
StrictACKs bool `yaml:"strict_acks,omitempty"`
// Number of packets to buffer for NACK
// Deprecated: use PacketBufferSizeVideo and PacketBufferSizeAudio
PacketBufferSize int `yaml:"packet_buffer_size,omitempty"`
// Number of packets to buffer for NACK - video
PacketBufferSizeVideo int `yaml:"packet_buffer_size_video,omitempty"`
// Number of packets to buffer for NACK - audio
PacketBufferSizeAudio int `yaml:"packet_buffer_size_audio,omitempty"`
// Throttle periods for pli/fir rtcp packets
PLIThrottle PLIThrottleConfig `yaml:"pli_throttle,omitempty"`
@@ -327,8 +331,10 @@ var DefaultConfig = Config{
ICEPortRangeEnd: 0,
STUNServers: []string{},
},
PacketBufferSize: 500,
StrictACKs: true,
PacketBufferSize: 500,
PacketBufferSizeVideo: 500,
PacketBufferSizeAudio: 200,
StrictACKs: true,
PLIThrottle: PLIThrottleConfig{
LowQuality: 500 * time.Millisecond,
MidQuality: time.Second,
pkg/rtc/config.go
+10 -2
View File
@@ -39,7 +39,8 @@ type WebRTCConfig struct {
}
type ReceiverConfig struct {
PacketBufferSize int
PacketBufferSizeVideo int
PacketBufferSizeAudio int
}
type RTPHeaderExtensionConfig struct {
@@ -72,6 +73,12 @@ func NewWebRTCConfig(conf *config.Config) (*WebRTCConfig, error) {
if rtcConf.PacketBufferSize == 0 {
rtcConf.PacketBufferSize = 500
}
if rtcConf.PacketBufferSizeVideo == 0 {
rtcConf.PacketBufferSizeVideo = rtcConf.PacketBufferSize
}
if rtcConf.PacketBufferSizeAudio == 0 {
rtcConf.PacketBufferSizeAudio = rtcConf.PacketBufferSize
}
// publisher configuration
publisherConfig := DirectionConfig{
@@ -129,7 +136,8 @@ func NewWebRTCConfig(conf *config.Config) (*WebRTCConfig, error) {
return &WebRTCConfig{
WebRTCConfig: *webRTCConfig,
Receiver: ReceiverConfig{
PacketBufferSize: rtcConf.PacketBufferSize,
PacketBufferSizeVideo: rtcConf.PacketBufferSizeVideo,
PacketBufferSizeAudio: rtcConf.PacketBufferSizeAudio,
},
Publisher: publisherConfig,
Subscriber: subscriberConfig,
pkg/rtc/mediatracksubscriptions.go
+4 -1
View File
@@ -103,11 +103,14 @@ func (t *MediaTrackSubscriptions) AddSubscriber(sub types.LocalParticipant, wr *
t.subscribedTracksMu.Unlock()
var rtcpFeedback []webrtc.RTCPFeedback
var maxTrack int
switch t.params.MediaTrack.Kind() {
case livekit.TrackType_AUDIO:
rtcpFeedback = t.params.SubscriberConfig.RTCPFeedback.Audio
maxTrack = t.params.ReceiverConfig.PacketBufferSizeAudio
case livekit.TrackType_VIDEO:
rtcpFeedback = t.params.SubscriberConfig.RTCPFeedback.Video
maxTrack = t.params.ReceiverConfig.PacketBufferSizeVideo
}
codecs := wr.Codecs()
for _, c := range codecs {
@@ -130,7 +133,7 @@ func (t *MediaTrackSubscriptions) AddSubscriber(sub types.LocalParticipant, wr *
BufferFactory: sub.GetBufferFactory(),
SubID: subscriberID,
StreamID: streamID,
MaxTrack: t.params.ReceiverConfig.PacketBufferSize,
MaxTrack: maxTrack,
PlayoutDelayLimit: sub.GetPlayoutDelayConfig(),
Pacer: sub.GetPacer(),
Trailer: trailer,
pkg/rtc/participant_internal_test.go
+1 -1
View File
@@ -731,7 +731,7 @@ func newParticipantForTestWithOpts(identity livekit.ParticipantIdentity, opts *p
if err != nil {
panic(err)
}
ff := buffer.NewFactoryOfBufferFactory(500)
ff := buffer.NewFactoryOfBufferFactory(500, 200)
rtcConf.SetBufferFactory(ff.CreateBufferFactory())
grants := &auth.ClaimGrants{
Video: &auth.VideoGrant{},
pkg/rtc/room.go
+2 -2
View File
@@ -87,7 +87,7 @@ type Room struct {
joinedAt atomic.Int64
// time that the last participant left the room
leftAt atomic.Int64
holds atomic.Int32
holds atomic.Int32
lock sync.RWMutex
@@ -164,7 +164,7 @@ func NewRoom(
participantOpts: make(map[livekit.ParticipantIdentity]*ParticipantOptions),
participantRequestSources: make(map[livekit.ParticipantIdentity]routing.MessageSource),
hasPublished: make(map[livekit.ParticipantIdentity]bool),
bufferFactory: buffer.NewFactoryOfBufferFactory(config.Receiver.PacketBufferSize),
bufferFactory: buffer.NewFactoryOfBufferFactory(config.Receiver.PacketBufferSizeVideo, config.Receiver.PacketBufferSizeAudio),
batchedUpdates: make(map[livekit.ParticipantIdentity]*participantUpdate),
closed: make(chan struct{}),
trailer: []byte(utils.RandomSecret()),
pkg/sfu/buffer/factory.go
+3 -3
View File
@@ -28,17 +28,17 @@ type FactoryOfBufferFactory struct {
audioPool *sync.Pool
}
func NewFactoryOfBufferFactory(trackingPackets int) *FactoryOfBufferFactory {
func NewFactoryOfBufferFactory(trackingPacketsVideo int, trackingPacketsAudio int) *FactoryOfBufferFactory {
return &FactoryOfBufferFactory{
videoPool: &sync.Pool{
New: func() interface{} {
b := make([]byte, trackingPackets*bucket.MaxPktSize)
b := make([]byte, trackingPacketsVideo*bucket.MaxPktSize)
return &b
},
},
audioPool: &sync.Pool{
New: func() interface{} {
b := make([]byte, bucket.MaxPktSize*200)
b := make([]byte, trackingPacketsAudio*bucket.MaxPktSize)
return &b
},
},