Files
livekit/pkg/sfu/connectionquality/connectionstats_test.go
T
2026-01-20 20:54:32 +05:30

904 lines
26 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 connectionquality
import (
"math"
"testing"
"time"
"github.com/stretchr/testify/require"
"github.com/livekit/livekit-server/pkg/sfu/buffer"
"github.com/livekit/livekit-server/pkg/sfu/rtpstats"
"github.com/livekit/protocol/codecs/mime"
"github.com/livekit/protocol/livekit"
"github.com/livekit/protocol/logger"
)
// -----------------------------------------------
type testReceiverProvider struct {
streams map[uint32]*buffer.StreamStatsWithLayers
lastSenderReportTime time.Time
}
func newTestReceiverProvider() *testReceiverProvider {
return &testReceiverProvider{}
}
func (trp *testReceiverProvider) setStreams(streams map[uint32]*buffer.StreamStatsWithLayers) {
trp.streams = streams
}
func (trp *testReceiverProvider) GetDeltaStats() map[uint32]*buffer.StreamStatsWithLayers {
return trp.streams
}
func (trp *testReceiverProvider) setLastSenderReportTime(at time.Time) {
trp.lastSenderReportTime = at
}
func (trp *testReceiverProvider) GetLastSenderReportTime() time.Time {
return trp.lastSenderReportTime
}
// -----------------------------------------------
func TestConnectionQuality(t *testing.T) {
trp := newTestReceiverProvider()
t.Run("quality scorer operation", func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: true,
IncludeJitter: true,
EnableBitrateScore: true,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(mime.MimeTypeOpus, false, now.Add(-duration))
cs.UpdateMuteAt(false, now.Add(-1*time.Second))
// no data and not enough unmute time should return default state which is EXCELLENT quality
cs.updateScoreAt(now)
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// best conditions (no loss, jitter/rtt = 0) - quality should stay EXCELLENT
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// introduce loss and the score should drop - 12% loss for Opus -> POOR
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 120,
PacketsLost: 30,
},
},
2: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 130,
PacketsLost: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(2.1), mos)
require.Equal(t, livekit.ConnectionQuality_POOR, quality)
// should climb to GOOD quality in one iteration if the conditions improve.
// although significant loss (12%) in the previous window, lowest score is
// bound so that climbing back does not take too long even under excellent conditions.
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// should stay at GOOD if conditions continue to be good
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// should climb up to EXCELLENT if conditions continue to be good
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// introduce loss and the score should drop - 5% loss for Opus -> GOOD
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 13,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// should stay at GOOD quality for another iteration even if the conditions improve
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// should climb up to EXCELLENT if conditions continue to be good
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// mute when quality is POOR should return quality to EXCELLENT
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 30,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(2.1), mos)
require.Equal(t, livekit.ConnectionQuality_POOR, quality)
now = now.Add(duration)
cs.UpdateMuteAt(true, now.Add(1*time.Second))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// unmute at specific time to ensure next window does not satisfy the unmute time threshold.
// that means even if the next update has 0 packets, it should hold state and stay at EXCELLENT quality
cs.UpdateMuteAt(false, now.Add(3*time.Second))
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// next update with no packets,
// but last RTCP is not set, should knock quality down to POOR
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(2.1), mos)
require.Equal(t, livekit.ConnectionQuality_POOR, quality)
// another dry spell, but last RTCP is not stale, should keep quality at POOR
now = now.Add(duration)
trp.setLastSenderReportTime(now.Add(time.Second))
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(2.1), mos)
require.Equal(t, livekit.ConnectionQuality_POOR, quality)
// yet another dry spell, but last RTCP is stale, should knock down quality at LOST
now = now.Add(duration)
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(1.3), mos)
require.Equal(t, livekit.ConnectionQuality_LOST, quality)
// mute when LOST should not bump up score/quality
now = now.Add(duration)
cs.UpdateMuteAt(true, now.Add(1*time.Second))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(1.3), mos)
require.Equal(t, livekit.ConnectionQuality_LOST, quality)
// unmute and send packets to bring quality back up
now = now.Add(duration)
cs.UpdateMuteAt(false, now.Add(2*time.Second))
for range 3 {
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 0,
},
},
})
cs.updateScoreAt(now.Add(duration))
now = now.Add(duration)
}
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// with lesser number of packet (simulating DTX).
// even higher loss (like 10%) should not knock down quality due to quadratic weighting of packet loss ratio
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 50,
PacketsLost: 5,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// mute/unmute to bring quality back up
now = now.Add(duration)
cs.UpdateMuteAt(true, now.Add(1*time.Second))
cs.UpdateMuteAt(false, now.Add(2*time.Second))
// RTT and jitter can knock quality down.
// at 2% loss, quality should stay at EXCELLENT purely based on loss, but with added RTT/jitter, should drop to GOOD
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 5,
RttMax: 400,
JitterMax: 30000,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// mute/unmute to bring quality back up
now = now.Add(duration)
cs.UpdateMuteAt(true, now.Add(1*time.Second))
cs.UpdateMuteAt(false, now.Add(2*time.Second))
// bitrate based calculation can drop quality even if there is no loss
cs.AddBitrateTransitionAt(1_000_000, now)
cs.AddBitrateTransitionAt(2_000_000, now.Add(2*time.Second))
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
Bytes: 8_000_000 / 8 / 5,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
// test layer mute via UpdateLayerMute API
cs.AddBitrateTransitionAt(1_000_000, now)
cs.AddBitrateTransitionAt(2_000_000, now.Add(2*time.Second))
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
Bytes: 8_000_000 / 8 / 5,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
now = now.Add(duration)
cs.UpdateLayerMuteAt(true, now)
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// unmute layer
cs.UpdateLayerMuteAt(false, now.Add(2*time.Second))
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
Bytes: 8_000_000 / 8 / 5,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
// pause
now = now.Add(duration)
cs.UpdatePauseAt(true, now)
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(2.1), mos)
require.Equal(t, livekit.ConnectionQuality_POOR, quality)
// resume
cs.UpdatePauseAt(false, now.Add(2*time.Second))
// although conditions are perfect, climbing back from POOR (because of pause above)
// will only climb to GOOD.
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
Bytes: 8_000_000 / 8 / 5,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality = cs.GetScoreAndQuality()
require.Greater(t, float32(4.1), mos)
require.Equal(t, livekit.ConnectionQuality_GOOD, quality)
})
t.Run("quality scorer dependent rtt", func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: false,
IncludeJitter: true,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(mime.MimeTypeOpus, false, now.Add(-duration))
cs.UpdateMuteAt(false, now.Add(-1*time.Second))
// RTT does not knock quality down because it is dependent and hence not taken into account
// at 2% loss, quality should stay at EXCELLENT purely based on loss. With high RTT (700 ms)
// quality should drop to GOOD if RTT were taken into consideration
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 5,
RttMax: 700,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
})
t.Run("quality scorer dependent jitter", func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: true,
IncludeJitter: false,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(mime.MimeTypeOpus, false, now.Add(-duration))
cs.UpdateMuteAt(false, now.Add(-1*time.Second))
// Jitter does not knock quality down because it is dependent and hence not taken into account
// at 2% loss, quality should stay at EXCELLENT purely based on loss. With high jitter (200 ms)
// quality should drop to GOOD if jitter were taken into consideration
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
1: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 250,
PacketsLost: 5,
JitterMax: 200,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, float32(4.6), mos)
require.Equal(t, livekit.ConnectionQuality_EXCELLENT, quality)
})
t.Run("codecs - packet", func(t *testing.T) {
type expectedQuality struct {
packetLossPercentage float64
expectedMOS float32
expectedQuality livekit.ConnectionQuality
}
testCases := []struct {
name string
mimeType mime.MimeType
isFECEnabled bool
packetsExpected uint32
expectedQualities []expectedQuality
}{
// NOTE: Because of EWMA (Exponentially Weighted Moving Average), these cut off points are not exact
// "audio/opus" - no fec - 0 <= loss < 2.5%: EXCELLENT, 2.5% <= loss < 7.5%: GOOD, >= 7.5%: POOR
{
name: "audio/opus - no fec",
mimeType: mime.MimeTypeOpus,
isFECEnabled: false,
packetsExpected: 200,
expectedQualities: []expectedQuality{
{
packetLossPercentage: 1.0,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
packetLossPercentage: 4.0,
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
packetLossPercentage: 9.2,
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
},
},
// "audio/opus" - fec - 0 <= loss < 3.75%: EXCELLENT, 3.75% <= loss < 11.25%: GOOD, >= 11.25%: POOR
{
name: "audio/opus - fec",
mimeType: mime.MimeTypeOpus,
isFECEnabled: true,
packetsExpected: 200,
expectedQualities: []expectedQuality{
{
packetLossPercentage: 3.0,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
packetLossPercentage: 4.4,
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
packetLossPercentage: 15.0,
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
},
},
// "audio/red" - no fec - 0 <= loss < 5%: EXCELLENT, 5% <= loss < 15%: GOOD, >= 15%: POOR
{
name: "audio/red - no fec",
mimeType: mime.MimeTypeRED,
isFECEnabled: false,
packetsExpected: 200,
expectedQualities: []expectedQuality{
{
packetLossPercentage: 4.0,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
packetLossPercentage: 6.0,
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
packetLossPercentage: 19.5,
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
},
},
// "audio/red" - fec - 0 <= loss < 7.5%: EXCELLENT, 7.5% <= loss < 22.5%: GOOD, >= 22.5%: POOR
{
name: "audio/red - fec",
mimeType: mime.MimeTypeRED,
isFECEnabled: true,
packetsExpected: 200,
expectedQualities: []expectedQuality{
{
packetLossPercentage: 6.0,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
packetLossPercentage: 10.0,
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
packetLossPercentage: 30.0,
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
},
},
// "video/*" - 0 <= loss < 2%: EXCELLENT, 2% <= loss < 6%: GOOD, >= 6%: POOR
{
name: "video/*",
mimeType: mime.MimeTypeVP8,
isFECEnabled: false,
packetsExpected: 200,
expectedQualities: []expectedQuality{
{
packetLossPercentage: 1.0,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
packetLossPercentage: 3.5,
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
packetLossPercentage: 8.0,
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
},
},
}
for _, tc := range testCases {
t.Run(tc.name, func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: true,
IncludeJitter: true,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(tc.mimeType, tc.isFECEnabled, now.Add(-duration))
for _, eq := range tc.expectedQualities {
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
123: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: tc.packetsExpected,
PacketsLost: uint32(math.Ceil(eq.packetLossPercentage * float64(tc.packetsExpected) / 100.0)),
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, eq.expectedMOS, mos)
require.Equal(t, eq.expectedQuality, quality)
now = now.Add(duration)
}
})
}
})
t.Run("bitrate", func(t *testing.T) {
type transition struct {
bitrate int64
offset time.Duration
}
testCases := []struct {
name string
transitions []transition
bytes uint64
expectedMOS float32
expectedQuality livekit.ConnectionQuality
}{
// NOTE: Because of EWMA (Exponentially Weighted Moving Average), these cut off points are not exact
// 1.0 <= expectedBits / actualBits < ~2.7 = EXCELLENT
// ~2.7 <= expectedBits / actualBits < ~20.1 = GOOD
// expectedBits / actualBits >= ~20.1 = POOR
{
name: "excellent",
transitions: []transition{
{
bitrate: 1_000_000,
},
{
bitrate: 2_000_000,
offset: 3 * time.Second,
},
},
bytes: 6_000_000 / 8,
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
name: "good",
transitions: []transition{
{
bitrate: 1_000_000,
},
{
bitrate: 2_000_000,
offset: 3 * time.Second,
},
},
bytes: uint64(math.Ceil(7_000_000.0 / 8.0 / 4.2)),
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
name: "poor",
transitions: []transition{
{
bitrate: 2_000_000,
},
{
bitrate: 1_000_000,
offset: 3 * time.Second,
},
},
bytes: uint64(math.Ceil(8_000_000.0 / 8.0 / 75.0)),
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
}
for _, tc := range testCases {
t.Run(tc.name, func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: true,
IncludeJitter: true,
EnableBitrateScore: true,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(mime.MimeTypeVP8, false, now)
for _, tr := range tc.transitions {
cs.AddBitrateTransitionAt(tr.bitrate, now.Add(tr.offset))
}
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
123: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 100,
Bytes: tc.bytes,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, tc.expectedMOS, mos)
require.Equal(t, tc.expectedQuality, quality)
})
}
})
t.Run("layer", func(t *testing.T) {
type transition struct {
distance float64
offset time.Duration
}
testCases := []struct {
name string
transitions []transition
expectedMOS float32
expectedQuality livekit.ConnectionQuality
}{
// NOTE: Because of EWMA (Exponentially Weighted Moving Average), these cut off points are not exact
// each spatial layer missed drops o quality level
{
name: "excellent",
transitions: []transition{
{
distance: 0.5,
},
{
distance: 0.0,
offset: 3 * time.Second,
},
},
expectedMOS: 4.6,
expectedQuality: livekit.ConnectionQuality_EXCELLENT,
},
{
name: "good",
transitions: []transition{
{
distance: 1.0,
},
{
distance: 1.5,
offset: 2 * time.Second,
},
},
expectedMOS: 4.1,
expectedQuality: livekit.ConnectionQuality_GOOD,
},
{
name: "poor",
transitions: []transition{
{
distance: 2.0,
},
{
distance: 2.6,
offset: 1 * time.Second,
},
},
expectedMOS: 2.1,
expectedQuality: livekit.ConnectionQuality_POOR,
},
}
for _, tc := range testCases {
t.Run(tc.name, func(t *testing.T) {
cs := NewConnectionStats(ConnectionStatsParams{
IncludeRTT: true,
IncludeJitter: true,
ReceiverProvider: trp,
Logger: logger.GetLogger(),
})
duration := 5 * time.Second
now := time.Now()
cs.StartAt(mime.MimeTypeVP8, false, now)
for _, tr := range tc.transitions {
cs.AddLayerTransitionAt(tr.distance, now.Add(tr.offset))
}
trp.setStreams(map[uint32]*buffer.StreamStatsWithLayers{
123: {
RTPStats: &rtpstats.RTPDeltaInfo{
StartTime: now,
EndTime: now.Add(duration),
Packets: 200,
},
},
})
cs.updateScoreAt(now.Add(duration))
mos, quality := cs.GetScoreAndQuality()
require.Greater(t, tc.expectedMOS, mos)
require.Equal(t, tc.expectedQuality, quality)
})
}
})
}