dandri/livekit
1
0
Fork 0
mirror of https://github.com/livekit/livekit.git synced 2026-08-28 11:44:43 +00:00
Code Issues Packages Projects Releases Wiki Activity

initialize bucket size by publish bitrates (#2763)

Browse Source
This commit is contained in:
cnderrauber
2024-06-06 14:31:20 +08:00
committed by GitHub
parent f1886ece42
commit 908baeb942
3 changed files with 20 additions and 7 deletions
Download Patch File Download Diff File Expand all files Collapse all files
pkg/rtc/mediatrack.go
+6 -1
View File
@@ -353,11 +353,16 @@ func (t *MediaTrack) AddReceiver(receiver *webrtc.RTPReceiver, track *webrtc.Tra
t.SetSimulcast(true)
}
var bitrates int
if len(ti.Layers) > int(layer) {
bitrates = int(ti.Layers[layer].GetBitrate())
}
if t.IsSimulcast() {
t.MediaTrackReceiver.SetLayerSsrc(mime, track.RID(), uint32(track.SSRC()))
}
buff.Bind(receiver.GetParameters(), track.Codec().RTPCodecCapability)
buff.Bind(receiver.GetParameters(), track.Codec().RTPCodecCapability, bitrates)
// if subscriber request fps before fps calculated, update them after fps updated.
buff.OnFpsChanged(func() {
pkg/sfu/buffer/buffer.go
+9 -1
View File
@@ -194,7 +194,7 @@ func (b *Buffer) SetAudioLossProxying(enable bool) {
b.enableAudioLossProxying = enable
}
func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability) {
func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability, bitrates int) {
b.Lock()
defer b.Unlock()
if b.bound {
@@ -264,6 +264,14 @@ func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapabili
}
}
}
if bitrates > 0 {
pps := bitrates / 8 / 1200
for pps > b.bucket.Capacity() {
if b.bucket.Grow() >= b.maxVideoPkts {
break
}
}
}
default:
b.codecType = webrtc.RTPCodecType(0)
pkg/sfu/buffer/buffer_test.go
+5 -5
View File
@@ -68,7 +68,7 @@ func TestNack(t *testing.T) {
buff.Bind(webrtc.RTPParameters{
HeaderExtensions: nil,
Codecs: []webrtc.RTPCodecParameters{vp8Codec},
}, vp8Codec.RTPCodecCapability)
}, vp8Codec.RTPCodecCapability, 0)
rtt := uint32(20)
buff.nacker.SetRTT(rtt)
for i := 0; i < 15; i++ {
@@ -127,7 +127,7 @@ func TestNack(t *testing.T) {
buff.Bind(webrtc.RTPParameters{
HeaderExtensions: nil,
Codecs: []webrtc.RTPCodecParameters{vp8Codec},
}, vp8Codec.RTPCodecCapability)
}, vp8Codec.RTPCodecCapability, 0)
rtt := uint32(30)
buff.nacker.SetRTT(rtt)
for i := 0; i < 15; i++ {
@@ -193,7 +193,7 @@ func TestNewBuffer(t *testing.T) {
buff.Bind(webrtc.RTPParameters{
HeaderExtensions: nil,
Codecs: []webrtc.RTPCodecParameters{vp8Codec},
}, vp8Codec.RTPCodecCapability)
}, vp8Codec.RTPCodecCapability, 0)
for _, p := range TestPackets {
buf, _ := p.Marshal()
@@ -229,7 +229,7 @@ func TestFractionLostReport(t *testing.T) {
buff.Bind(webrtc.RTPParameters{
HeaderExtensions: nil,
Codecs: []webrtc.RTPCodecParameters{opusCodec},
}, opusCodec.RTPCodecCapability)
}, opusCodec.RTPCodecCapability, 0)
for i := 0; i < 15; i++ {
pkt := rtp.Packet{
Header: rtp.Header{SequenceNumber: uint16(i), Timestamp: uint32(i)},
@@ -261,7 +261,7 @@ func TestFractionLostReport(t *testing.T) {
buff.Bind(webrtc.RTPParameters{
HeaderExtensions: nil,
Codecs: []webrtc.RTPCodecParameters{opusCodec},
}, opusCodec.RTPCodecCapability)
}, opusCodec.RTPCodecCapability, 0)
for i := 0; i < 15; i++ {
pkt := rtp.Packet{
Header: rtp.Header{SequenceNumber: uint16(i), Timestamp: uint32(i)},