Files
livekit/pkg/sfu/connectionquality/connectionstats_test.go
T
9551c52c85 Try 2 to consolidate mime type (#3407)
* Normalize mime type and add utilities.

An attempt to normalize mime type and avoid string compares remembering
to do case insensitive search.

Not the best solution. Open to ideas. But, define our own mime types
(just in case Pion changes things and Pion also does not have red mime
type defined which should be easy to add though) and tried to use it everywhere.
But, as we get a bunch of callbacks and info from Pion, needed conversion in
more places than I anticipated. And also makes it necessary to carry
that cognitive load of what comes from Pion and needing to process it
properly.

* more locations

* test

* Paul feedback

* MimeType type

* more consolidation

* Remove unused

* test

* test

* mime type as int

* use string method

* Pass error details and timeouts. (#3402)

* go mod tidy (#3408)

* Rename CHANGELOG to CHANGELOG.md (#3391)

Enables markdown features in this otherwise already markdown'ish formatted document

* Update config.go to properly process bool env vars (#3382)

Fixes issue https://github.com/livekit/livekit/issues/3381

* fix(deps): update go deps (#3341)

Generated by renovateBot

Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>

* Use a Twirp server hook to send API call details to telemetry. (#3401)

* Use a Twirp server hook to send API call details to telemetry.

* mage generate and clean up

* Add project_id

* deps

* - Redact requests
- Do not store responses
- Extract top level fields room_name, room_id, participant_identity,
  participant_id, track_id as appropriate
- Store status as int

* deps

* Update pkg/sfu/mime/mimetype.go

* Fix prefer codec test

* handle down track mime changes

---------

Co-authored-by: Denys Smirnov <dennwc@pm.me>
Co-authored-by: Philzen <Philzen@users.noreply.github.com>
Co-authored-by: Pablo Fuente Pérez <pablofuenteperez@gmail.com>
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
Co-authored-by: Paul Wells <paulwe@gmail.com>
Co-authored-by: cnderrauber <zengjie9004@gmail.com>
2025-02-10 10:44:15 +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/mime"
"github.com/livekit/livekit-server/pkg/sfu/rtpstats"
"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 i := 0; i < 3; i++ {
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)
})
}
})
}