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Adds `srs-integ`, a black-box integration suite for the SRS media server, and fixes the SRS bugs it found: over 120 of them, in every protocol. The suite tests SRS from outside the process, only through what a client, an operator or a callback server can see, so a rewritten server could pass the same cases. The suite now has about 1,450 cases (1,353 main and 99 timer), with no known bugs left: every bug a case found is fixed in this PR. It also removes some unused features, adds env support for a few more directives, and changes a few hook and API fields. ## srs-integ suite `cmd/integ` is a Go module of its own, `srs-integ`, separate from the root `srsx` module. It imports nothing from the repository. - **Run:** `make integ` makes the test media, then runs the cases in two `go test` processes at once: the main cases and the `TestTimer_` cases. `make integ-unit` runs the unit tests of the clients and analyzers, and `bin/srs-integ` gives `probe`, `publish` and `prepare` from the command line. - **Server adapter:** each server is `objs/srs -e`, configured only by env vars, on ports from `SRS_INTEG_PORTS` (14000-14999 in the Port Plan). Most cases share a Common server, split into 4 shards. Cases that need other settings run on variant servers, and groups share these where they can. - **Case classes:** Fast (≤100ms), Normal (≤500ms) and Timer (≤10s, only in `TestTimer_` groups). A case that runs over its limit fails. `KnownBug` marks a case that fails because of an SRS bug, and the case fails once the bug is fixed, so the mark can't go stale. - **Clients and analyzers written for the suite:** RTMP and RTMPS (both handshakes, any chunk size, enhanced RTMP, malformed input), FLV, TS, AAC/MP3, MP4, HLS (TS, fMP4, AES-128, SAMPLE-AES) and DASH. Each one fills a shared stream report: codecs, picture size, sequence headers, keyframes, timestamps and A/V sync. - **Clients built on libraries:** WebRTC WHIP/WHEP on pion (UDP and TCP, NACK/RTX/TWCC recorder, raw DTLS sessions), SRT on gosrt, RTSP on gortsplib, and a GB28181 PS-over-RTP publisher with no SIP. - **Test media:** generated from `doc/source.flv` by `make integ-prepare` and never committed. The files cover H.264, HEVC, VP9, AV1, AAC, Opus, MP3 and G.711, in FLV, MP4 and TS. A manifest keeps the media from being made again when nothing changed. - **Feature coverage:** the HTTP API, HTTPS API and exporter, reload, heartbeat, RTMP, RTMPS, HTTP-FLV, HTTP-TS, HTTP AAC and MP3, HTTPS, HLS, DASH, WebRTC, SRT, RTSP, GB28181, the MPEG-TS over UDP caster, every HTTP hook, security rules, forward, edge, the origin cluster, DVR, static files and VOD, signals, the command line and env settings. - **Conversion matrix:** each source (RTMP/RTMPS, WHIP, SRT, GB28181) is played by each player (RTMP, HTTP-FLV, HTTP-TS, MP3, AAC, HLS ts and fMP4, DASH, WHEP, SRT, RTSP) with each codec it can carry. The README has the matrix. - **Scale:** `-integ.scale=N` runs each case N times. A run of about 1,000 cases passes in about 11s. ## SRS bug fixes Every fix has a utest and, where a client can see it, an srs-integ case that failed before the fix. The main ones by area: - **HLS:** fMP4: - The default init file is now listed by its dir. - `hls_entry_prefix` now applies to fMP4 URIs. - MP3 streams now get an audio track in the init file. - Each publish now starts with a discontinuity. - `hls_keys` now encrypts the samples with cbcs. - A segment that fails to open is now dropped. Before, it crashed SRS. Other HLS fixes: key URIs, AES padding, HLS starts again after `hls_dispose`, the ratios, a TS republish, session names, and the `hls_ctx` of another stream's session is now checked as a new session. - **DASH:** the config, the MPD (window, update period, HEVC and MP3 codecs), init files of MP3 and of one-track streams, and dispose. - **WebRTC:** - Keyframes with an SEI, and Sender Reports and their mapping. - Opus timing, G.711, and WHEP SSRCs. - A refused WHIP publisher now frees its stream. - The start and hooks, and WebRTC over TCP. - **SRT:** only the configured latencies are set, as ffmpeg does. Also fixed: HEVC headers and CTS, frame counts, and the listener. - **RTSP:** a republish to a kept player, PLAY without SETUP, unknown methods, closes, DESCRIBE/TEARDOWN, request bodies and `on_stop`. - **GB28181:** - SSRC parsing. - HEVC to RTMP is no longer sent in legacy FLV tags. - A crash on a second media connection. - Latency with no audio, sliced frames, CTS, session lifetime, and the muxer and caster output. - **RTMP and HTTP:** - Timestamps that jump back without time correction no longer stop a player. - The republish param and the publish timeout after a pause. - The no-origin close and the RTMPS port. - HTTP ranges, including MP4 suffix ranges and VOD 416. - Redirects, mounts, unmounts and URL `..`. - CORS, API auth paths, JSONP 400 and 403 answers. - Static files and the request echo. - **Hooks:** - The heartbeat URL's query is now sent. - `on_play` clients and client types, and `on_dvr`. - A client that `on_connect` refused no longer gets `on_close`. - **Other:** DVR (MP4 info frame, audio rate, audio cut), listen ports above 65535, the env vhost check, the UDP caster, process stats, the tcmalloc rate, the encoder loop, the edge failover, STUN, coroutine joins, and the resource manager's fast-id cache (WebRTC and GB lookups never used it). ## Behavior changes - **Hooks:** - `on_publish`, `on_unpublish`, `on_play` and `on_stop` have a new `type` field after `client_id`, such as `rtmp-play` or `hls-play`, the same value as `/api/v1/clients`. - Every protocol now sends `param` with its leading `?`. - HTTP players send their own `http(s)://` tcUrl, and WHIP/WHEP clients send `webrtc://<host>/<app>`. - **API:** `/api/v1/features` adds `srt`, `rtsp`, `gb28181`, `rtc`, `simulator` and `sanitizer`. The `/api/v1/` index lists only the URLs SRS serves. - **Env support:** - The vhost security rules. - The edge protocol. - The origin cluster and coworkers. - The DVR apply filter. - **Timers in seconds:** `hls_dispose`, `dash_dispose`, `dvr_duration`, `queue_length`, `stun_timeout` and the heartbeat `interval` now accept fractions such as `0.5`. A fraction used to be cut to whole seconds, so `0.5` read as 0. - **Removed features:** a config or env var that names one of these, on or off, now makes SRS refuse to start (and `-t` fail) with `<name> is removed`: - HDS, including `--hds=on`. - The FLV POST caster and `asprocess`. - `query_latest_version` and `first_wait_for_qlv`. - The vhost `bandcheck`, `refer` and `atc_auto`. - Edge `token_traverse`. Check clients with the edge's HTTP hooks instead. ## Tests, CI and skills - **CI:** a new `integration-test` job builds `trunk/Dockerfile.test` (now with `--simulator=on`) and runs srs-integ. - It caches the test media. - It runs the main cases and the timer cases as separate steps, each retried on its own. - Limits are scaled by 4 with `-integ.limit-scale=4`. - **Unit tests:** more stable. They use random seeds, the test servers no longer share ports or leak fds, and a gtest listener fails a test that leaves a handler on a global resource manager. - **Skills:** - The `srs-develop` Gate and Full tiers run `make integ`. - `integration-tests.md` explains how to add cases. - The new `internal-audit-for-srs` skill audits each commit blind before review. --------- Co-authored-by: Claude Opus 5.5 (1M context) <noreply@anthropic.com> Co-authored-by: chatgpt-codex-connector[bot] <199175422+chatgpt-codex-connector[bot]@users.noreply.github.com>
169 lines
7.1 KiB
Go
169 lines
7.1 KiB
Go
package cases
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import (
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"fmt"
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"testing"
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"time"
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"srs-integ/internal/check"
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"srs-integ/internal/flv"
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"srs-integ/internal/media"
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"srs-integ/internal/report"
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"srs-integ/internal/rtmp"
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"srs-integ/internal/server"
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)
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// The publish timeouts of TestTimer_Rtmp_PublishTimeout. SRS checks a
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// publisher once per wait: the first wait is the first-packet timeout, each
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// later one the normal timeout. So a publisher that sends nothing is closed
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// at 100ms, one that stops after its first messages at the next 200ms check,
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// and an idle one, with no player, at the first check after 300ms idle.
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const (
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firstPacketTimeout = 100 * time.Millisecond
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normalTimeout = 200 * time.Millisecond
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kickoffForIdle = 300 * time.Millisecond
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)
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// timerSlack is how much earlier than its timeout the case may see SRS close
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// a publisher: the case starts its clock a little after SRS starts its own.
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const timerSlack = 20 * time.Millisecond
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// sharedPublishTimeout is the config of the timer server that
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// TestTimer_Rtmp_PublishTimeout and TestTimer_Heartbeat_Defaults share: the
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// short publish timeouts and kickoff_for_idle, every hook on a mock, and the
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// heartbeat every 1s with SRS's defaults, which the publish cases never read.
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func sharedPublishTimeout() server.Config {
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cfg := server.PublishConfig(server.Publish{
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FirstPacketTimeout: firstPacketTimeout,
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NormalTimeout: normalTimeout,
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KickoffForIdle: kickoffForIdle,
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})
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cfg.Hooks = true
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cfg.Heartbeat = &sharedHeartbeatDefaults
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return cfg
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}
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// TestTimer_Rtmp_PublishTimeout publishes on a variant with short publish
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// timeouts and kickoff_for_idle. The timeouts are milliseconds, so their cases
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// are Normal; the kick waits on the publish checks after a player leaves,
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// which SRS sees only after its play receive thread wakes, so its cases are
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// Timer. They share one server because the kick acts only on the checks the
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// timeouts pace.
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func TestTimer_Rtmp_PublishTimeout(t *testing.T) {
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t.Parallel()
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srs := server.Shared(t, "PublishTimeout", sharedPublishTimeout())
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// FirstPacketTimeout publishes and sends nothing: SRS closes the publisher
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// after the first-packet timeout, not the longer normal timeout.
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check.Case(t, "FirstPacketTimeout", check.Normal, func(c *check.C) {
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stream := streamName(c)
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// SRS starts the timeout during the publish, so the lower bound counts
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// from before the publish and the upper bound from after it.
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start := time.Now()
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pub := publishOnly(c, rtmpURL(srs, "live/"+stream), rtmp.FMLE)
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published := time.Now()
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closed, err := untilClosed(pub, time.Second)
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c.Must(err, "the publisher")
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d := closed.Sub(start)
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c.Check(d >= firstPacketTimeout-timerSlack, "closed %v after the publish began, want at least %v", d, firstPacketTimeout)
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d = closed.Sub(published)
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c.Check(d < normalTimeout-timerSlack, "closed %v after the publish, want before the normal timeout %v", d, normalTimeout)
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waitHook(c, srs, "on_unpublish", stream)
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})
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// NormalTimeout sends the first 200ms of the file at once and then
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// nothing: SRS closes the publisher at least the normal timeout after its
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// last message, not at the first-packet check.
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check.Case(t, "NormalTimeout", check.Normal, func(c *check.C) {
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stream := streamName(c)
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url := rtmpURL(srs, "live/"+stream)
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playerOf(c, url)
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pub := publishTags(c, url, readTags(c, media.H264HighAAC, 0, 200*time.Millisecond))
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last := time.Now()
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closed, err := untilClosed(pub, time.Second)
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c.Must(err, "the publisher")
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d := closed.Sub(last)
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c.Check(d >= normalTimeout-timerSlack, "closed %v after the last message, want at least %v", d, normalTimeout)
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waitHook(c, srs, "on_unpublish", stream)
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})
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// PauseShorterThanNormal sends the first 100ms at once, pauses 250ms,
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// longer than the first-packet timeout and shorter than the normal one,
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// then sends the next 100ms: the player gets all of it, so the later
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// messages are timed by the normal timeout.
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check.Case(t, "PauseShorterThanNormal", check.Normal, func(c *check.C) {
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url := rtmpURL(srs, "live/"+streamName(c))
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player := playerOf(c, url)
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first := readTags(c, media.H264HighAAC, 0, 100*time.Millisecond)
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later := readTags(c, media.H264HighAAC, 100*time.Millisecond, 200*time.Millisecond)
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pub := publishTags(c, url, first)
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time.Sleep(250 * time.Millisecond)
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c.Must(writeTags(pub, later), "publish after the pause")
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ref := analyzeTags(c, append(append([]*media.FLVTag{}, first...), later...))
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var a flv.Analyzer
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u := report.Until{VideoFrames: ref.Video.Frames, AudioFrames: ref.Audio.Frames}
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c.CheckNoError(readUntil(player, &a, u, time.Second), "the media after the pause")
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})
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// SteadyStreamKept publishes in real time to a player for 350ms, past
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// the first-packet check and the first normal check: the gaps between
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// the messages of a real-time stream don't close it.
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check.Case(t, "SteadyStreamKept", check.Normal, func(c *check.C) {
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url := rtmpURL(srs, "live/"+streamName(c))
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player := playerOf(c, url)
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startPublisher(c, url, media.Path(c, media.H264HighAAC), 0)
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var a flv.Analyzer
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u := report.Until{Metadata: true, VideoHeader: true, AudioHeader: true, Duration: 350 * time.Millisecond}
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c.Must(readUntil(player, &a, u, time.Second), "play")
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checkPlayed(c, &a.Report, "h264", "aac")
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})
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// KickedWhenIdle publishes in real time with no player: SRS closes the
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// publisher once it has been idle for kickoff_for_idle.
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check.Case(t, "KickedWhenIdle", check.Timer, func(c *check.C) {
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stream := streamName(c)
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start := time.Now()
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pub := startPublisher(c, rtmpURL(srs, "live/"+stream), media.Path(c, media.H264HighAAC), 0)
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closed, err := untilClosed(pub, 2*time.Second)
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c.Must(err, "the publisher")
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d := closed.Sub(start)
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c.Check(d >= kickoffForIdle-timerSlack, "kicked after %v, want at least %v", d, kickoffForIdle)
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waitHook(c, srs, "on_unpublish", stream)
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})
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// KickedAfterLastPlayerLeaves publishes in real time to a player for
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// 600ms, past the check that kicks an idle publisher, then closes the
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// player: SRS kicks the publisher at least kickoff_for_idle later.
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check.Case(t, "KickedAfterLastPlayerLeaves", check.Timer, func(c *check.C) {
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stream := streamName(c)
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url := rtmpURL(srs, "live/"+stream)
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player := playerOf(c, url)
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pub := startPublisher(c, url, media.Path(c, media.H264HighAAC), 0)
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var a flv.Analyzer
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u := report.Until{Metadata: true, VideoHeader: true, AudioHeader: true, Duration: 600 * time.Millisecond}
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c.Must(readUntil(player, &a, u, time.Second), "play while the player keeps the publisher")
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player.Close()
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left := time.Now()
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closed, err := untilClosed(pub, 2*time.Second)
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c.Must(err, "the publisher")
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d := closed.Sub(left)
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c.Check(d >= kickoffForIdle-timerSlack, "kicked %v after the player left, want at least %v", d, kickoffForIdle)
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waitHook(c, srs, "on_unpublish", stream)
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})
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}
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// untilClosed reads a publisher's connection until SRS closes it, and
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// returns when. It fails when the connection is still open after within.
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func untilClosed(pub *rtmp.Client, within time.Duration) (time.Time, error) {
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pub.SetDeadline(time.Now().Add(within))
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for {
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if _, err := pub.ReadTag(); isTimeout(err) {
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return time.Time{}, fmt.Errorf("still open after %v", within)
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} else if err != nil {
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return time.Now(), nil
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}
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}
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}
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