6ec97067de
This PR makes WebRTC a core feature of SRS and enforces C++98 compatibility by: 1. Always Enable WebRTC Support - Remove `--rtc=on|off` configuration option - WebRTC is now always enabled - Eliminate all `#ifdef SRS_RTC` conditional compilation blocks - Include WebRTC-related modules (RTC, SRTP, DTLS) in all builds - Update build scripts to always link WebRTC dependencies 2. Enforce C++98 Compatibility - Remove `--cxx11=on|off` and `--cxx14=on|off` configuration options - Force `SRS_CXX11=NO` and `SRS_CXX14=NO` in build system - Move these options to deprecated section with warnings - Ensure codebase maintains C++98 standard compatibility 3. Remove Windows/Cygwin Support - Remove all Windows and Cygwin64 conditional compilation blocks (#ifdef _WIN32, #ifdef CYGWIN64) - Delete Cygwin64 build configurations from build scripts ( auto/options.sh, auto/depends.sh, configure) - Remove Cygwin64 assembly files and State Threads platform support ( md_cygwin64.S) - Eliminate Windows-specific GitHub Actions workflows and CI/CD jobs - Remove NSIS packaging files and Windows installer generation - Delete Windows documentation and update feature lists to mark support as removed in v7.0 - Simplify OS detection to only support Unix-like systems (Linux, macOS) 4. Code Cleanup - Remove conditional WebRTC code blocks throughout the codebase - Simplify build configuration by removing WebRTC-related conditionals - Update constructor delegation patterns to be C++98 compatible - Fix vector initialization to use C++98 syntax - Eliminate Windows-specific implementations for file operations, time handling, and networking - Unified platform handling with consistent POSIX API usage --------- Co-authored-by: OSSRS-AI <winlinam@gmail.com> |
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|---|---|---|
| .. | ||
| ffmpeg-4-fit | ||
| gperftools-2-fit | ||
| gprof | ||
| gtest-fit | ||
| httpx-static | ||
| libsrtp-2-fit | ||
| openssl-1.1-fit | ||
| patches | ||
| signaling | ||
| srs-bench | ||
| srs-docs | ||
| srt-1-fit | ||
| st-srs | ||
| openssl-OpenSSL_1_0_2u.tar.gz | ||
| opus-1.3.1.tar.gz | ||
| README.md | ||
http-parser-2.1.zip
- for srs to support http callback.
- https://github.com/nodejs/http-parser
- https://github.com/ossrs/http-parser
- https://ossrs.net/lts/zh-cn/license#http-parser
nginx-1.5.7.zip
- http://nginx.org/
- for srs to support hls streaming.
srt-1-fit srt-1.5.3.tar.gz
openssl-1.1-fit openssl-1.1.1l.tar.gz
openssl-1.1.0e.zip openssl-OpenSSL_1_0_2u.tar.gz
- http://www.openssl.org/source/openssl-1.1.0e.tar.gz
- openssl for SRS(with-ssl) RTMP complex handshake to delivery h264+aac stream.
- SRTP depends on openssl 1.0.*, so we use both ssl versions.
- https://ossrs.net/lts/zh-cn/license#openssl
libsrtp-2.3.0.tar.gz
- For WebRTC, SRTP to encrypt and decrypt RTP.
- https://github.com/cisco/libsrtp/releases/tag/v2.3.0
ffmpeg-4.2.tar.gz opus-1.3.1.tar.gz
- http://ffmpeg.org/releases/ffmpeg-4.2.tar.gz
- https://github.com/xiph/opus/releases/tag/v1.3.1
- To support RTMP/WebRTC transcoding.
- https://ossrs.net/lts/zh-cn/license#ffmpeg
gtest-fit
- google test framework.
- https://github.com/google/googletest/releases/tag/release-1.11.0
gperftools-2-fit
- gperf tools for performance benchmark.
- https://github.com/gperftools/gperftools/releases/tag/gperftools-2.9.1
st-srs st-1.9.zip state-threads state-threads-1.9.1.tar.gz
- Patched ST from https://github.com/ossrs/state-threads
- https://ossrs.net/lts/zh-cn/license#state-threads
JSON
USRSCTP
links:
- state-threads: https://github.com/ossrs/state-threads
- x264: ftp://ftp.videolan.org/pub/videolan/x264/snapshots/x264-snapshot-20131129-2245-stable.tar.bz2
- lame: http://nchc.dl.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz
- yasm: http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
- speex: http://downloads.xiph.org/releases/speex/speex-1.2rc1.tar.gz