mingyuejingque / rtaudio_speex
用rtaudio来采集、播放,并用speexdsp来做回声消除。
☆17Updated 6 years ago
Related projects ⓘ
Alternatives and complementary repositories for rtaudio_speex
- simple cross-platform multimedia player and framework☆27Updated 3 years ago
- media sdk based on webrtc☆40Updated last year
- a video engine include receiver and sender base on webrtc☆12Updated 7 years ago
- ☆16Updated 7 years ago
- Video Jitter Buffer derived from WebRTC☆16Updated 5 years ago
- sip client,gb28181 client☆23Updated 6 years ago
- The webrtc study demo☆13Updated last year
- WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.☆21Updated 5 years ago
- LiveStream gateway for WebRTC 流媒体网关服务器☆25Updated 4 months ago
- libwebrtc_audio_preprocessing.so compile for android(AEC, AEC3,AECM,AGC,AGC2,VAD,NS),更多示例,请参见:☆12Updated 5 years ago
- VS2015 webrtc bulid solution, now open source for everyone.☆14Updated 9 months ago
- clone repo https://sourceforge.net/p/mcumediaserver/code/HEAD/tree/ doc: https://github.com/atyenoria/MCU-Media-Server/blob/master/Inst…☆11Updated 8 years ago
- webRTCtest-linux☆12Updated 6 years ago
- webrtc各个版本在不同平台的库和demo,供上层用户使用。☆13Updated 2 years ago
- 通过不同算法进行验证混音效果☆26Updated 5 years ago
- ts muxer☆28Updated 5 years ago
- rewrite WebRTC's jitter buffer for PJSIP☆14Updated 11 years ago
- get vedio&audio ES from RTP packet over UDP from GBT28181 IPC☆31Updated 9 years ago
- encode mp4 form H264 and AAC☆24Updated 6 years ago
- ☆17Updated 3 years ago
- rtsp-server rtsp-client☆14Updated 7 years ago
- ffmpeg and webrtc note☆14Updated 4 years ago
- 用于解析h265和h264的编码信息(包含hdr信息)☆19Updated 4 years ago
- An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.☆34Updated 4 years ago
- the webrtc client for the janus webrtc gateway☆37Updated 5 years ago
- 一个移植于WebRtc项目的通用基础lib库☆25Updated 5 years ago
- 基于ijkplayer的定制化修改,修复bug,开启FFmpeg硬件解码☆28Updated 5 years ago