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876411828f |
Windows: Native x64 build with MSVC. v8.0.65 (#4787)
SRS builds and runs natively on Windows x64 again, now with MSVC and without Cygwin. This change adds `srs.exe` and `srs_utest.exe`, which are tested on a Windows host and in CI, and the release attaches a Windows zip. The Cygwin64 build was removed in v7.0.60 (#4447) because it was too hard to maintain. This build sits on the native Windows support that State Threads got in v8.0.56, and every Windows feature has automated tests so it does not rot again. ## Build - A Windows-only CMake build, `trunk/cmake/windows/CMakeLists.txt`, reads the same module list as `./configure` (`trunk/auto/module_files.txt`). The Unix `./configure` build is unchanged. - Dependencies: vcpkg pins OpenSSL 3, libsrtp, libsrt, Opus and googletest in `trunk/vcpkg.json`. MSYS2 builds the bundled FFmpeg-fit with MSVC, for AAC and Opus in WebRTC. - The platform shims for sockets, file IO, time, processes, signals and system stats live in `srs_core_platform.hpp/.cpp`. App code has no new `#ifdef` spread. - `skills/srs-develop/scripts/srs-windows-runner.sh` builds, tests and packages. The Windows host, CI and the release all use it. Its `vs` step makes a Visual Studio solution. - The IDE CMake, `trunk/cmake/CMakeLists.txt`, now takes its sources from the module list too. ## Features on Windows - Live: RTMP, RTMPS, HTTP-FLV, HTTP-TS, HTTPS, HLS (TS and fMP4), DASH, DVR, static files and VOD. - HTTP API, exporter, hooks, security, edge and forward. - WebRTC WHIP and WHEP with AAC and Opus conversion, SRT through libsrt including SRT to RTMP, GB28181 and RTSP. - Ingest, transcode and exec, which start `ffmpeg.exe` through `CreateProcessW`. A job object kills the child when SRS exits. - Control: Ctrl+C quits fast. Ctrl+Break or closing the console quits gracefully. The RAW API runs `reload` and `reopen-log`. The pid file uses `LockFileEx`. Daemon mode is refused, because `srs.exe` is a console program. ## Tests and CI - **Windows host:** `srs_utest` 3325 of 3325. `cmd/integ` main and timer cases pass at `-integ.limit-scale 4`. The OBS RTMP, SRT and WHIP cases now publish to native `srs.exe` instead of WSL. - **Skipped on Windows:** six timer cases that need Unix signals. Each names the case that covers it (`cmd/integ/internal/check/skips.go`). - **CI:** a `windows-utest` job in `test.yml` on `windows-2025`. The build names `cl` for CMake, because the current image's Visual Studio 18 puts its clang on PATH. - **Release:** a `windows` job in `release.yml` attaches `SRS-Windows-x86_64-<ver>.zip`. The zip needs no DLLs beyond Windows' own. - **Linux and macOS:** the Full tier is unchanged. A Linux RTMP A/B against `develop` at 11,520 players measured +1.8% SRS CPU, below the 10% line. ## Fixes found on the way, for every OS - A forked child that fails to redirect or exec, such as for ffmpeg, now calls `_exit`. Before, with SRT started, it hung in libsrt's exit handlers, and SRS treated ffmpeg as running. - A child's `process ppid=...` banner now reaches its log: stdout is flushed before `execv`. - `mkdir_all` can create a missing dir right under the root, such as `/data/hls`. - An HEVC sequence header from WebRTC is the same for the same parameter sets. The hvcC record had uninitialized fields, so RTMP could get two headers. - With OpenSSL 3, which a Unix `--use-sys-ssl` build may use too: RSA DTLS certificates are 2048-bit. `dtls_version dtls1.0` works, but leaves out the anonymous, null and MD5 suites that security level 1 refuses. - The utest builds with RTSP or GB28181 off, the configure defaults. - `openssl3-api-check.sh` works under GCC. Before, every compile failed and it reported PASS. - Several `cmd/integ` cases no longer flake: the DASH `availabilityStartTime` bound, the RTC hook queue cases, the GB publish waits, and RTC to RTMP keyframe requests. ## Limits - ST on Windows uses WSAPoll only, with a ~15.6 ms timer. Non-blocking TCP sends are all-or-nothing. - No `/proc` stats, inotify reload, `SO_REUSEPORT`, daemon, Windows service or installer. - No ASan, gperftools, Valgrind or backtraces. - UTF-8 file names need Windows 10 1903 or later. --------- Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com> |
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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> |