AgoraIO-Community / SoloLinks
Agora Solo is an open source speech codec, it was developed based on Silk with BWE(Bandwidth Extension) and MDC(Multi Description Coding). With these technologies, Solo is enable to resist weak networks at low bitrates.
☆241Updated 5 years ago
Alternatives and similar repositories for Solo
Users that are interested in Solo are comparing it to the libraries listed below
Sorting:
- webrtc source code from https://chromium.googlesource.com/external/webrtc☆388Updated 4 months ago
- WebRTC编译国内加速镜像☆158Updated 5 years ago
- Noise Suppression Module Port From WebRTC☆338Updated 4 years ago
- webrtc中apm相关代码的提取,包括AEC/NS/AGC/VAD ,另外还包括mp3/aac编码器、SoundTouch☆106Updated 2 years ago
- Let's hack into WebRTC :)☆128Updated 5 years ago
- Mirror of WebRTC, update everyday. (https://webrtc.googlesource.com/src)☆301Updated 11 months ago
- webrtc audio processing☆408Updated 5 years ago
- 低延时直播(Low-Latency Streaming,LLS)是网易云信推出的低延时、强同步、高质量的直播产品。低延时直播产品基于云信 WE-CAN 全球智能路由网络,为开发者提供毫秒级延时 、多平台同步、高可靠高并发的直播服务。 集成网易云信播放器 SDK/NERTD …☆242Updated 2 years ago
- WebRTC学习资料整理,包括博客、文章、开源项目☆110Updated 6 years ago
- Hercules 是以json+lua的灵活方式控制视频混画混流mcu,简单灵活完成业务需求。☆125Updated 4 years ago
- ☆84Updated 5 years ago
- webrtc学习整理(业务和代码梳理)☆131Updated 5 years ago
- Acoustic Echo Canceller for Mobile Module Port From WebRTC☆205Updated 6 months ago
- FLV Parser.☆109Updated 6 years ago
- Voice Activity Detector Module Port From WebRTC☆179Updated 5 years ago
- Automatic Gain Control Module Port From WebRTC☆181Updated 6 years ago
- WebRTC AudioProc (AEC, VAD, NS...)☆105Updated 4 years ago
- Speex voice codec mirror - THIS IS A MIRROR, DEVELOPMENT HAPPENS AT https://gitlab.xiph.org/xiph/speex☆486Updated 4 months ago
- A c wrapper library of Google Chromium QUIC. Can be integrated into FFmpeg for playing QUIC protocol stream.Support platforms including A…☆108Updated 6 years ago
- Tech☆83Updated 8 years ago
- This is WebRtc noise suppression module demo.☆103Updated 6 years ago
- A cross-platform WebRTC/RTMP/SRT SDK base on metaRTC☆398Updated last year
- Compile module in WebRTC Native to static library☆27Updated 5 years ago
- The client library srs-librtmp of SRS(https://github.com/ossrs/srs)☆253Updated 4 years ago
- ☆228Updated 3 years ago
- Docker for https://github.com/open-webrtc-toolkit/owt-server☆106Updated 5 years ago
- Audio AEC AGC ANS(webrtc aec) process with a single lib, just 6 APIs, easy to use. 基于webrtc的极简音频3A处理上层封装,6个API,支持常见采样率、单双声道。☆42Updated 2 years ago
- webRTCtest-linux☆12Updated 7 years ago
- This repo contains the upstream webrtc stack code, with updates for Open WebRTC Toolkit.☆265Updated last year
- Yang Real-Time Communication,功能强大的视音频开发SDK。☆38Updated 2 years ago