winlinvip / ai-translationLinks
This solution is not good enough, we're researching a better version: https://github.com/winlinvip/vod-translator so we archive this repository.
☆22Updated last year
Alternatives and similar repositories for ai-translation
Users that are interested in ai-translation are comparing it to the libraries listed below
Sorting:
- Apply https://github.com/k2-fsa/sherpa-ncnn in live streaming and WebRTC☆21Updated 2 years ago
- Speech-end detection library, based on WebRTC's VAD engine☆26Updated 3 months ago
- Mirror of The Official WebRTC repository☆52Updated this week
- cpp streamer work in dynamic modules for media develop. It include flv/mpegts/rtmp/webrtc modules, and go on developing more modules☆68Updated last month
- 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。☆39Updated 2 years ago
- Media Management System: ingestion, processing, encoding, delivery, ...☆40Updated last week
- BFCP message/client/server libraries☆13Updated 6 years ago
- A stable and easy-to-use Python wrapper for FFmpeg-C-API with ctypes & cuda☆92Updated last month
- The full C++ implementation of mediasoup☆35Updated 5 years ago
- ☆18Updated last year
- Support WebRTC(WHIP) for FFmpeg.☆156Updated last week
- Stream FFMPEG based Audio and Video using WebRtc. It is the most fastest P2P based streamer which gets Audio and Video from FFMPEG and th…☆16Updated 3 years ago
- WebRTC Chromium Open Source and integrate RNNoise. Codecs supported: AV1, VP8, VP9, H264, H265.☆25Updated last year
- streaming proxy server☆13Updated last year
- Simple C++ wrapper of the SRT protocol for building Server/Client transport solutions☆20Updated last year
- Implment WebRTC H264 encoder by calling OBS's internal encoder in order to use x264 and some hardware H264 encoders for 1080P acceleratio…☆55Updated 6 years ago
- 一个程序员的学习笔记☆22Updated 2 years ago
- srt encoder☆101Updated 2 years ago
- ☆17Updated 2 years ago
- A package used to test webrtc apm functions, such as aec, ns☆16Updated 6 years ago
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Updated 6 years ago
- 快直播传输层SDK☆17Updated 2 months ago
- voip packet-loss concealment algorithm derived from WebRTC neteq module☆30Updated 3 years ago
- Dugon distributed mediaserver (mediasoup wrapper)☆24Updated 8 months ago
- WebRTC GIPS C/C++ API demos☆58Updated 12 years ago
- SRT server base ZLToolKit for intergrate to ZLM☆10Updated 3 years ago
- webRTCtest-linux☆12Updated 7 years ago
- libwebrtc_audio_preprocessing.so compile for android(AEC, AEC3,AECM,AGC,AGC2,VAD,NS),更多示例,请参见:☆12Updated 6 years ago
- Noise Suppression Module Port From WebRTC☆73Updated 2 years ago
- Live media streaming. High performance Http, secure websocket and webrtc server. Supports H264, Opus and Mp3.☆16Updated 2 weeks ago