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srs/cmd/integ/probe_test.go
Winlin 551ee22d1f Integ: Add the srs-integ test suite, and fix 135 bugs. v8.0.61 (#4772)
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>
2026-10-17 15:20:30 -04:00

99 lines
3.3 KiB
Go

// Copyright (c) 2026 Winlin
//
// SPDX-License-Identifier: MIT
package main
import (
"bytes"
"net/http"
"net/http/httptest"
"strings"
"testing"
"srs-integ/internal/media"
)
// TestProbeFile probes a test file: to its end with -duration 0, and only its
// first second by default.
func TestProbeFile(t *testing.T) {
file := media.Path(t, media.H264HighAAC)
var stdout, stderr bytes.Buffer
if code := run([]string{"probe", "-duration", "0", file}, &stdout, &stderr); code != 0 {
t.Fatalf("exit code %v, stderr %q", code, stderr.String())
}
for _, want := range []string{"h264 High 768x320", "30 keyframes", "750 frames", "aac 44100Hz 2ch"} {
if !strings.Contains(stdout.String(), want) {
t.Errorf("stdout %q, want it to contain %q", stdout.String(), want)
}
}
stdout.Reset()
if code := run([]string{"probe", file}, &stdout, &stderr); code != 0 {
t.Fatalf("exit code %v, stderr %q", code, stderr.String())
}
if !strings.Contains(stdout.String(), "1 keyframe,") {
t.Errorf("stdout %q, want one second with 1 keyframe", stdout.String())
}
}
// TestProbeTSAndAudioFiles probes the test media remuxed into TS, ADTS and
// MP3 files, each by its extension, to the end.
func TestProbeTSAndAudioFiles(t *testing.T) {
cases := []struct {
name string
path string
want []string
}{
{"Ts", media.Remux(t, media.H264HighAAC, "mpegts"), []string{"h264 High 768x320", "30 keyframes", "750 frames", "aac 44100Hz 2ch"}},
{"Aac", media.Remux(t, media.AACOnly, "adts"), []string{"no video", "aac 44100Hz 2ch: 1 sequence header"}},
{"Mp3", media.Remux(t, media.H264HighMP3, "mp3", "-vn", "-write_xing", "0"), []string{"no video", "mp3 44100Hz 2ch"}},
}
for _, tc := range cases {
t.Run(tc.name, func(t *testing.T) {
var stdout, stderr bytes.Buffer
if code := run([]string{"probe", "-duration", "0", tc.path}, &stdout, &stderr); code != 0 {
t.Fatalf("exit code %v, stderr %q", code, stderr.String())
}
for _, want := range tc.want {
if !strings.Contains(stdout.String(), want) {
t.Errorf("stdout %q, want it to contain %q", stdout.String(), want)
}
}
})
}
}
// TestProbeSegments probes the test media as HLS with TS and fmp4 segments,
// as DASH, served over HTTP, and as an MP4 file, to the end.
func TestProbeSegments(t *testing.T) {
whole := []string{"h264 High 768x320", "30 keyframes", "750 frames", "aac 44100Hz 2ch"}
serve := func(dir string) string {
s := httptest.NewServer(http.FileServer(http.Dir(dir)))
t.Cleanup(s.Close)
return s.URL
}
cases := []struct {
name string
url string
}{
{"HlsTs", serve(media.RemuxDir(t, media.H264HighAAC, "hls", "a.m3u8", "-hls_playlist_type", "vod")) + "/a.m3u8"},
{"HlsFmp4", serve(media.RemuxDir(t, media.H264HighAAC, "hls", "a.m3u8", "-hls_playlist_type", "vod", "-hls_segment_type", "fmp4")) + "/a.m3u8"},
{"Dash", serve(media.RemuxDir(t, media.H264HighAAC, "dash", "a.mpd")) + "/a.mpd"},
{"Mp4", media.Remux(t, media.H264HighAAC, "mp4")},
}
for _, tc := range cases {
t.Run(tc.name, func(t *testing.T) {
var stdout, stderr bytes.Buffer
if code := run([]string{"probe", "-duration", "0", tc.url}, &stdout, &stderr); code != 0 {
t.Fatalf("exit code %v, stderr %q", code, stderr.String())
}
for _, want := range whole {
if !strings.Contains(stdout.String(), want) {
t.Errorf("stdout %q, want it to contain %q", stdout.String(), want)
}
}
})
}
}