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Winlin 069016c714 SRT: Fix the timestamps of 48 kHz AAC from SRT to RTMP. v8.0.64 (#4786) 
Fixes #4762: with `srt_to_rtmp on`, 48 kHz AAC published over SRT got
wrong per-frame timestamps on the RTMP side, so the audio of RTMP
playback, HLS and forwards crackled. This replaces #4765, and keeps its
author's credit for what was learned from it.

## Cause

An SRT publisher such as FFmpeg packs several AAC frames into one PES,
which carries a single PTS. `SrsSrtFrameBuilder::on_ts_audio()` times
frame `i` at `pts + i * 1024 / sample_rate`, and took `sample_rate` from
the FLV sound rate, which has only 5.5, 11, 22 and 44.1 kHz:

- 48 kHz frames were stamped 23.22 ms apart instead of 21.33 ms, and the
timeline stepped at each PES.
- Every other rate FLV can't represent was wrong too: 96, 88.2, 64, 32,
24, 16, 12, 8 and 7.35 kHz.
- Players that place audio by timestamp, such as hls.js and YouTube
ingest, played gaps and overlaps. flv.js and ffplay hide it.

It regressed in the SRS 5.0 SRT rewrite. The AAC sequence header was
already right, and the UDP MPEG-TS ingest path already used the AAC
rate.

## Fix

Take the rate from the ADTS `sampling_frequency_index`, as the UDP
MPEG-TS path does, and keep 44.1 kHz for a reserved index, which has no
rate.

---------

Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
Co-authored-by: Nandisha <61039123+AlexFreik@users.noreply.github.com>
2026-10-21 21:28:21 -04:00
.agents
AI: Skills-only agent config, OpenClaw for SRSBot. v8.0.58 (#4764)
2026-10-08 21:54:23 -04:00
.claude
AI: Skills-only agent config, OpenClaw for SRSBot. v8.0.58 (#4764)
2026-10-08 21:54:23 -04:00
.github
Integ: Add the srs-integ test suite, and fix 135 bugs. v8.0.61 (#4772)
2026-10-17 15:20:30 -04:00
.kiro
AI: Skills-only agent config, OpenClaw for SRSBot. v8.0.58 (#4764)
2026-10-08 21:54:23 -04:00
.openclaw
Perf: Load and profile tools, and the first SRS and Go proxy fixes. v8.0.63 (#4785)
2026-10-21 18:46:30 -04:00
.run
Verify STUN integrity, fix SRT and WebRTC TCP republish, run SRS by env only. v8.0.48 (#4751)
2026-09-24 15:28:26 -04:00
.vscode
Improve VS Code debug and CMake build tasks. v8.0.25 (#4662)
2026-08-17 21:11:33 -04:00
cmake
OpenClaw: add and refine ST knowledge-base and learning/review skills (#4643)
2026-03-05 09:57:08 -05:00
cmd
SRT: Fix the timestamps of 48 kHz AAC from SRT to RTMP. v8.0.64 (#4786)
2026-10-21 21:28:21 -04:00
doc
Skills: Isolate test ports and test ST on every OS. v8.0.57 (#4761)
2026-10-06 09:46:29 -04:00
internal
SRT: Fix the timestamps of 48 kHz AAC from SRT to RTMP. v8.0.64 (#4786)
2026-10-21 21:28:21 -04:00
scripts
Fix failed live source crash, ST source of truth, testability. v8.0.54 (#4758)
2026-10-02 12:16:27 -04:00
skills
Perf: Load and profile tools, and the first SRS and Go proxy fixes. v8.0.63 (#4785)
2026-10-21 18:46:30 -04:00
tools
WebRTC: Add RFC 4588 RTX config, tests, tools fix. v8.0.41 (#4746)
2026-09-21 20:38:54 -04:00
trunk
SRT: Fix the timestamps of 48 kHz AAC from SRT to RTMP. v8.0.64 (#4786)
2026-10-21 21:28:21 -04:00
.clang-format
AI: Add utest to cover process module
2025-10-15 07:52:46 -04:00
.codecov.yml
AI: Improve converage for app rtc module.
2025-09-27 09:40:57 -04:00
.gitignore
Perf: Load and profile tools, and the first SRS and Go proxy fixes. v8.0.63 (#4785)
2026-10-21 18:46:30 -04:00
CONTRIBUTING.md
Update CHANGELOG for #4309. v7.0.31
2025-04-26 06:58:00 -04:00
dev-docker
JSON: Update vendored json-parser library. v8.0.18 (#4716)
2026-08-12 18:43:13 -04:00
Dockerfile
AI: Add utest to cover listener module.
2025-10-10 09:50:19 -04:00
go.mod
Proxy: Harden internal/env tests and add counterfeiter fakes. v7.0.144 (#4665)
2026-04-18 20:33:07 -04:00
go.sum
Proxy: Harden internal/env tests and add counterfeiter fakes. v7.0.144 (#4665)
2026-04-18 20:33:07 -04:00
LICENSE
RTC: Answer the video codec of the stream for WebRTC play. v8.0.36 (#4742)
2026-09-18 08:26:21 -04:00
Makefile
Perf: Load and profile tools, and the first SRS and Go proxy fixes. v8.0.63 (#4785)
2026-10-21 18:46:30 -04:00
oryx
JSON: Update vendored json-parser library. v8.0.18 (#4716)
2026-08-12 18:43:13 -04:00
README.md
Release v8.0-d1, 8.0 dev1, v8.0.63, 356563 lines.
2026-10-21 18:54:20 -04:00
SECURITY.md
Update SECURITY policy.
2023-01-30 14:20:37 +08:00
state-threads
Fix failed live source crash, ST source of truth, testability. v8.0.54 (#4758)
2026-10-02 12:16:27 -04:00
website
Claude: Report on_play and on_publish rejection to the client. v8.0.34 (#4737)
2026-09-13 17:02:25 -04:00

README.md

SRS(Simple Realtime Server)

SRS/8.0 (Free) is a simple, high-performance, AI-driven real-time media server. It supports RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH, and GB28181 on Linux and macOS across x86_64, ARMv7, AArch64, Apple M1, RISC-V, LoongArch, and MIPS architectures. Supported codecs include H.264, H.265, AV1, VP9, AAC, Opus, and G.711. See the complete list of features.

SRS Overview

Note: For more details on the single-node architecture for SRS, please visit the following link.

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Usage

Please check the Getting Started guide in English or Chinese. We highly recommend using SRS with docker:

docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
    -p 8000:8000/udp -p 10080:10080/udp ossrs/srs:6

Open http://localhost:8080/ to verify, and then stream using the following FFmpeg command:

ffmpeg -re -i ./doc/source.flv -c copy -f flv -y rtmp://localhost/live/livestream

Alternatively, stream by OBS using the following configuration:

  • Service: Custom
  • Server: rtmp://localhost/live
  • Stream Key: livestream

Play the following streams using media players:

If you'd like to use WebRTC, convert RTMP to WebRTC, or convert WebRTC to RTMP, please check out the wiki documentation in either English or Chinese.

To learn more about RTMP, HLS, HTTP-FLV, SRT, MPEG-DASH, WebRTC protocols, clustering, HTTP API, DVR, and transcoding, please check the documents in English or Chinese.

AI Agent

I recommend to use AI to understand and maintain your SRS, please follow the wiki AI Agent for details.

Contributing

The maintainers, and contributors are listed here. The maintainers who made significant contributions and maintained parts of SRS are listed below, ranked by the number of commits:

  • Winlin: Founder of the project, focusing on ST and Issues/PR. Responsible for architecture and maintenance.
  • XiaoZhihong: Concentrates on WebRTC/QUIC and SRT, with expertise in network QoS. Contributed to ARM on ST and was the original contributor for WebRTC.
  • ChenHaibo: Specializes in GB28181 and HTTP API, contributing to patches for FFmpeg with WHIP.
  • ZhangJunqin: Focused on H.265, Prometheus Exporter, and API module.
  • XiaLixin: Specializes in GB28181, with expertise in live streaming and WebRTC.
  • Jacob Su: Jacob Su has contributed to various modules of SRS.
  • ShiWei: Specializes in SRT and H.265, maintaining SRT and FLV patches for FFmpeg. An expert in codecs and FFmpeg.
  • ChenGuanghua: Focused on WebRTC/QoS and introduced the Asan toolchain to SRS.
  • LiPeng: Concentrates on WebRTC and contributes to memory management and smart pointers.
  • ZhaoWenjie: One of the earliest contributors, focusing on HDS. Has expertise in client technology.
  • WuPengqiang: Focused on H.265, initially contributed to the FFmpeg module in SRS for transcoding AAC with OPUS for WebRTC.

A huge THANK YOU goes out to:

We're really thankful to everyone in the community for helping us find bugs and improve the project. To stay in touch and keep helping our community, please check out this guide.

Of course, financial support is very important to us, so we can build the website and use AI tools to maintain the community. Please visit OpenCollective to become a backer or sponsor.

LICENSE

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Releases

Features

Please read FEATURES.

Changelog

Please read CHANGELOG.

Performance

Please read PERFORMANCE.

Architecture

Please read ARCHITECTURE.

Ports

Please read PORTS.

APIs

Please read APIS.

Mirrors

Please read MIRRORS.

Dockers

Please read DOCKERS.

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Description
SRS - это простой, высокоэффективный видеосервер реального времени, поддерживающий RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH и GB28181
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