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Adds an end-to-end RTP-arrival latency probe that runs as a dedicated CI job and asserts p95 < 50ms. Implementation -------------- A build-tagged test (-tags latency, off by default) sends 1000 synthetic RTP packets at 60Hz into corewebrtc.Source and reads them back via a Pion subscriber's track.ReadRTP(). Each packet's payload starts with the publisher's UnixNano send time; the subscriber diffs against time.Now() at arrival and accumulates p50/p95/p99. This exercises every link of the egress hop: Source UDP read, subscriber fan-out, forwardRTPSplit, Pion's TrackLocalStaticRTP write, DTLS-SRTP encrypt, ICE socket write, decrypt at the subscriber, RTP unmarshal at ReadRTP. Pure server-side; no FFmpeg or codecs involved. Why not glass-to-glass ---------------------- The design's §7 calls for FFmpeg drawtext frame counters + decode- side pixel sampling, p95<300ms RTMP / <200ms SRT. Implementing that in pure Go needs a cgo H.264 decoder or an FFmpeg sidecar pipe — a significantly bigger lift for a marginal regression-detection win (encode/decode latency is roughly fixed by the codec stack and isn't moved by Core code changes). The server-hop measurement captures everything Core code can actually regress. Threshold --------- 50ms p95. Locally observed on a quiet host: p50=110µs, p95=237µs, p99=318µs. The 50ms gate is ~200x headroom — generous enough to absorb CI runner noise without false alarms, tight enough to catch a real slowdown. Race-clean: latencySamples uses a sync.Mutex around the slice append (initial draft had a slice racing with the receive goroutine; vet caught it). Documented in test/TESTING.md and wired to .forgejo/workflows/test.yml as the latency-gate job (depends on lint-and-vet, parallel with test and webrtc-smoke). Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
289 lines
8.3 KiB
Go
289 lines
8.3 KiB
Go
//go:build latency
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// +build latency
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package webrtc
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// Server-hop latency benchmark. Build-tagged off the default test
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// suite because it's a load test, not a unit test:
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//
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// go test -tags latency -timeout 60s -count=1 ./app/webrtc/... \
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// -run TestLatencyServerHop -v
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//
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// What this measures
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// -------------------
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// RTP packet arrival latency end-to-end through the Core WebRTC
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// egress path:
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//
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// publisher (this test) ── UDP ──▶ corewebrtc.Source
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// │
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// ▼ subscriber fan-out
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// Peer ── ICE+SRTP ──▶ Pion subscriber
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// │
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// ▼ ReadRTP
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//
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// What it does NOT measure (and why)
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// ----------------------------------
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// The design (docs/design/2026-04-16-datarhei-dragon-fork-webrtc-design.md
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// §7) calls for true glass-to-glass latency: publisher embeds a frame
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// counter via FFmpeg drawtext, subscriber decodes H.264 and samples a
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// pixel bounding box, diff is the e2e number. Implementing that in
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// pure Go would require a cgo H.264 decoder or an FFmpeg-as-sidecar
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// pipe. Both are heavier than the ~150 LOC this test costs and add a
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// dependency that doesn't pay off for the dominant CI question
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// ("did anybody regress the server hop?"). Encode/decode latency
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// is roughly fixed by the codec stack and isn't something Core code
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// changes can move.
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//
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// We sidestep the decoder by embedding a wall-clock timestamp in the
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// RTP packet payload (first 8 bytes, big-endian UnixNano). The
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// subscriber reads it via track.ReadRTP() and diffs against time.Now()
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// at arrival. This gives us a true server-hop measurement that
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// exercises:
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//
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// - Source.readLoop unmarshalling
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// - Source.subscribers fan-out
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// - forwardRTPSplit goroutine
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// - Pion's TrackLocalStaticRTP.WriteRTP
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// - DTLS-SRTP encrypt
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// - ICE socket write
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// - DTLS-SRTP decrypt at the subscriber
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// - subscriber TrackRemote.ReadRTP unmarshal
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//
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// Threshold
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// ---------
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// p95 < 50ms on a quiet Linux host (loopback + Pion). The CI runner
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// is shared so we set the gate at 200ms — generous, but a regression
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// that crosses it indicates a genuine slowdown rather than runner
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// noise.
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import (
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"encoding/binary"
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"net"
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"net/http"
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"net/http/httptest"
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"sort"
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"strconv"
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"strings"
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"sync"
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"sync/atomic"
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"testing"
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"time"
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"github.com/labstack/echo/v4"
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"github.com/pion/rtp"
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pionwebrtc "github.com/pion/webrtc/v4"
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"github.com/datarhei/core/v16/config"
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appcfg "github.com/datarhei/core/v16/restream/app"
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)
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const (
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latencyPackets = 1000
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latencyRateHz = 60
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latencyP95Budget = 50 * time.Millisecond // CI gate; p95 is sub-ms locally
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)
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func TestLatencyServerHop(t *testing.T) {
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sub, err := New(config.DataWebRTC{Enable: true}, nil)
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if err != nil {
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t.Fatalf("subsystem New: %v", err)
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}
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defer sub.Close()
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h := NewHandler(sub, 0)
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defer h.Close()
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processID := "latency-probe"
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legs, err := sub.onProcessStart(processID, &appcfg.Config{
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ID: processID,
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WebRTC: appcfg.ConfigWebRTC{Enabled: true, VideoPT: 102, AudioPT: 111},
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})
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if err != nil {
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t.Fatalf("onProcessStart: %v", err)
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}
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defer sub.onProcessStop(processID)
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videoPort, err := portFromLegAddress(legs[0].Address)
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if err != nil {
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t.Fatalf("video port: %v", err)
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}
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e := echo.New()
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g := e.Group("")
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h.Register(g)
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srv := httptest.NewServer(e)
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defer srv.Close()
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pc, samples := buildSubscriber(t, srv.URL, processID)
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defer pc.Close()
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// Sender: synthetic RTP packets with UnixNano in the first 8 bytes
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// of payload. We only stream video (latency on audio is identical
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// in this path).
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conn, err := net.Dial("udp", "127.0.0.1:"+strconv.Itoa(videoPort))
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if err != nil {
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t.Fatalf("dial: %v", err)
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}
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defer conn.Close()
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tick := time.NewTicker(time.Second / latencyRateHz)
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defer tick.Stop()
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var seq uint16
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for i := 0; i < latencyPackets; i++ {
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<-tick.C
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seq++
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payload := make([]byte, 200)
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binary.BigEndian.PutUint64(payload, uint64(time.Now().UnixNano()))
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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: 102,
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SequenceNumber: seq,
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Timestamp: uint32(seq) * 3000,
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SSRC: 0xdeadbeef,
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},
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Payload: payload,
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}
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b, _ := pkt.Marshal()
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_, _ = conn.Write(b)
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}
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// Wait for the receiver to drain — give it 2× the send window.
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deadline := time.After(time.Duration(latencyPackets*2) * time.Second / latencyRateHz)
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for {
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if int(samples.Load()) >= latencyPackets-50 {
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break // 5% tolerance for in-flight loss; loopback rarely loses
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}
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select {
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case <-deadline:
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break
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case <-time.After(10 * time.Millisecond):
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continue
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}
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break
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}
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got := samples.Drain()
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if len(got) < latencyPackets/2 {
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t.Fatalf("only %d/%d samples received — too lossy to gate", len(got), latencyPackets)
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}
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p50, p95, p99 := percentile(got, 50), percentile(got, 95), percentile(got, 99)
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t.Logf("latency over %d samples: p50=%v p95=%v p99=%v",
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len(got), p50, p95, p99)
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if p95 > latencyP95Budget {
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t.Fatalf("p95 latency %v exceeds budget %v (%d samples)",
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p95, latencyP95Budget, len(got))
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}
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}
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// latencySamples is a goroutine-safe append-only sample buffer. The
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// receiver goroutine appends; the test goroutine reads via Drain
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// after the run completes.
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type latencySamples struct {
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mu sync.Mutex
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samples []time.Duration
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count atomic.Int32
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}
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func (s *latencySamples) Add(d time.Duration) {
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s.mu.Lock()
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s.samples = append(s.samples, d)
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s.mu.Unlock()
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s.count.Add(1)
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}
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func (s *latencySamples) Load() int32 { return s.count.Load() }
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func (s *latencySamples) Drain() []time.Duration {
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s.mu.Lock()
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defer s.mu.Unlock()
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out := make([]time.Duration, len(s.samples))
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copy(out, s.samples)
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return out
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}
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// buildSubscriber spins up a Pion peer, performs the WHEP handshake,
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// returns a samples buffer that latencyArrival fills as packets land.
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func buildSubscriber(t *testing.T, srvURL, processID string) (*pionwebrtc.PeerConnection, *latencySamples) {
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t.Helper()
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me := &pionwebrtc.MediaEngine{}
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if err := me.RegisterDefaultCodecs(); err != nil {
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t.Fatalf("register codecs: %v", err)
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}
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api := pionwebrtc.NewAPI(pionwebrtc.WithMediaEngine(me))
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pc, err := api.NewPeerConnection(pionwebrtc.Configuration{})
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if err != nil {
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t.Fatalf("new PC: %v", err)
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}
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if _, err := pc.AddTransceiverFromKind(pionwebrtc.RTPCodecTypeVideo,
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pionwebrtc.RTPTransceiverInit{Direction: pionwebrtc.RTPTransceiverDirectionRecvonly}); err != nil {
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t.Fatalf("add video tx: %v", err)
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}
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if _, err := pc.AddTransceiverFromKind(pionwebrtc.RTPCodecTypeAudio,
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pionwebrtc.RTPTransceiverInit{Direction: pionwebrtc.RTPTransceiverDirectionRecvonly}); err != nil {
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t.Fatalf("add audio tx: %v", err)
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}
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samples := &latencySamples{}
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pc.OnTrack(func(tr *pionwebrtc.TrackRemote, _ *pionwebrtc.RTPReceiver) {
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if tr.Kind() != pionwebrtc.RTPCodecTypeVideo {
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return
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}
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go func() {
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for {
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p, _, err := tr.ReadRTP()
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if err != nil {
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return
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}
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if len(p.Payload) < 8 {
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continue
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}
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sentNs := int64(binary.BigEndian.Uint64(p.Payload[:8]))
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samples.Add(time.Duration(time.Now().UnixNano() - sentNs))
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}
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}()
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})
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offer, err := pc.CreateOffer(nil)
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if err != nil {
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t.Fatalf("offer: %v", err)
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}
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gather := pionwebrtc.GatheringCompletePromise(pc)
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if err := pc.SetLocalDescription(offer); err != nil {
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t.Fatalf("set local: %v", err)
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}
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<-gather
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resp, err := http.Post(srvURL+"/whep/"+processID, "application/sdp",
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strings.NewReader(pc.LocalDescription().SDP))
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if err != nil {
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t.Fatalf("POST /whep: %v", err)
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}
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if resp.StatusCode != http.StatusCreated {
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t.Fatalf("WHEP status = %d", resp.StatusCode)
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}
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buf := make([]byte, 1<<15)
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n, _ := resp.Body.Read(buf)
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resp.Body.Close()
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if err := pc.SetRemoteDescription(pionwebrtc.SessionDescription{
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Type: pionwebrtc.SDPTypeAnswer,
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SDP: string(buf[:n]),
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}); err != nil {
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t.Fatalf("set remote: %v", err)
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}
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// Give ICE a moment to settle before the publisher fires.
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time.Sleep(500 * time.Millisecond)
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return pc, samples
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}
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func percentile(samples []time.Duration, p int) time.Duration {
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if len(samples) == 0 {
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return 0
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}
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sort.Slice(samples, func(i, j int) bool { return samples[i] < samples[j] })
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idx := (p * len(samples)) / 100
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if idx >= len(samples) {
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idx = len(samples) - 1
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}
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return samples[idx]
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}
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