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:
- 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。☆37Updated 2 years ago
- Mirror of The Official WebRTC repository☆52Updated this week
- ☆2Updated 2 years ago
- ☆17Updated last year
- cpp streamer work in dynamic modules for media develop. It include flv/mpegts/rtmp/webrtc modules, and go on developing more modules☆67Updated 2 weeks ago
- Speech-end detection library, based on WebRTC's VAD engine☆26Updated 3 months ago
- Apply https://github.com/k2-fsa/sherpa-ncnn in live streaming and WebRTC☆21Updated 2 years ago
- Support WebRTC(WHIP) for FFmpeg.☆155Updated this week
- Thesis code☆14Updated 3 years ago
- Media Management System: ingestion, processing, encoding, delivery, ...☆40Updated last week
- ☆17Updated 2 years ago
- WebRTC Chromium Open Source and integrate RNNoise. Codecs supported: AV1, VP8, VP9, H264, H265.☆23Updated 11 months ago
- WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.☆22Updated 5 years ago
- The full C++ implementation of mediasoup☆35Updated 5 years ago
- voip packet-loss concealment algorithm derived from WebRTC neteq module☆30Updated 3 years ago
- Implment WebRTC H264 encoder by calling OBS's internal encoder in order to use x264 and some hardware H264 encoders for 1080P acceleratio…☆54Updated 6 years ago
- a video engine include receiver and sender base on webrtc☆12Updated 8 years ago
- ☆11Updated 2 years ago
- 一个程序员的学习笔记☆22Updated 2 years ago
- media server based on c++17, support webrtc/rtmp/httpflv/hls/websocket flv☆91Updated 5 months ago
- WebRTC AudioProc (AEC, VAD, NS...)☆104Updated 4 years ago
- webrtc各个版本在不同平台的库和demo,供上层用户使用。☆13Updated 3 years ago
- Audio Loudness Normalization Filter Port From FFmpeg☆11Updated 6 years ago
- Snowem is a lightweight live streaming server, based on webrtc technology. Its design mainly focuses on simplicity, scalability and high …☆88Updated 5 years ago
- ☆6Updated 2 years ago
- srt encoder☆100Updated 2 years ago
- 使用srs_librtmp和RawQuic通过QUIC推RTMP流,实现RTMP OVER QUIC。☆94Updated 5 years ago
- BFCP message/client/server libraries☆13Updated 6 years ago
- Acoustic Echo Canceller for Mobile Module Port From WebRTC☆200Updated 3 months ago
- Agora Solo is an open source speech codec, it was developed based on Silk with BWE(Bandwidth Extension) and MDC(Multi Description Coding)…☆239Updated 5 years ago