xiangxud / rtmp-to-webrtc
Demonstrate a RTMP server that publishes to WebRTC
☆11Updated 2 years ago
Alternatives and similar repositories for rtmp-to-webrtc:
Users that are interested in rtmp-to-webrtc are comparing it to the libraries listed below
- ☆17Updated 2 years ago
- 混音播放器,音频录、播音托管程序。支持wav、mp3、aac格式播放,rtp搭载pcma、aac发收测试,webrtc vad录音辅助(无说话时消音)、ns/nsx录、播音噪音抑制、aec/aecm回声消除、agc录音自动增益 等☆12Updated 2 years ago
- WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.☆22Updated 5 years ago
- WebRTC Chromium Open Source and integrate RNNoise. Codecs supported: AV1, VP8, VP9, H264, H265.☆19Updated 5 months 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。☆35Updated last year
- Windows下音视频对讲演示程序☆28Updated last month
- 用rtaudio来采集、播放,并用speexdsp来做回声消除。☆18Updated 6 years ago
- AMediaCodec 安卓ndk硬编解码h264 acc,以及推流RTMP☆18Updated 6 years ago
- android rtc client for janus gateway☆32Updated 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
- ☆15Updated this week
- Open source, Open mind.☆16Updated 10 months ago
- 支持国家标准 GB/T 28181-2016和拉取视频流实现缩放、码流控制、RTP推送☆16Updated 3 years ago
- 根据metaRTC5.0 修改的推流demo,原作者的推流demo是基于QT开发和编译的,我对QT不熟悉也不想安装QT,所以修改metapushSRS5 用cmake 编译的demo☆13Updated last year
- A demo application for testing mediasoup-go☆31Updated 2 years ago
- 通过不同算法进行验证混音效果☆26Updated 5 years ago
- 桃夭是套基于`Mediasoup`开发的`WebRTC`音视频信令服务,可以非常方便的扩展信令接入更多智能终端。☆29Updated last week
- h265 wasm,ffmpeg api demo☆10Updated 3 years ago
- mediasoup c++ server (SFU)☆14Updated 7 months ago
- 一个程序员的学习笔记☆21Updated last year
- The full C++ implementation of mediasoup☆33Updated 4 years ago
- sip client,gb28181 client☆23Updated 6 years ago
- golang use ffmpeg to mix the video☆10Updated last year
- cpp streamer work in dynamic modules for media develop. It include flv/mpegts/rtmp/webrtc modules, and go on developing more modules☆60Updated 3 months 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
- a video engine include receiver and sender base on webrtc☆12Updated 7 years ago
- ffmpeg-webrtc for whip and whep protocol☆64Updated last year
- LiveStream gateway for WebRTC 流媒体网关服务器☆25Updated 7 months ago