juha-h / libwebrtcLinks
webrtc AEC module and its required dependencies module for android
☆19Updated last year
Alternatives and similar repositories for libwebrtc
Users that are interested in libwebrtc are comparing it to the libraries listed below
Sorting:
- 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☆16Updated 3 years ago
- ☆17Updated 2 years ago
- 用rtaudio来采集、播放，并用speexdsp来做回声消除。☆18Updated 7 years ago
- android rtc client for janus gateway☆33Updated 6 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
- 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 2 years ago
- Mirror of The Official WebRTC repository☆52Updated this week
- An android demo of webrtc audio process module, including aec, aecm, ns, agc and so on. The NDK part of the project is built using cmake …☆29Updated 7 years ago
- A program transcode rtsp to rtmp with FFmpeg API.☆17Updated 5 years ago
- sip client，gb28181 client☆24Updated 6 years ago
- simplest rtsp client without dependences☆18Updated 6 years ago
- Windows下音视频对讲演示程序（声学回音消除、噪音抑制、语音活动检测、自动增益控制、自适应抖动缓冲）☆36Updated 2 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。☆41Updated 2 years ago
- mpp decode h264,rga resize, drm display by hdmi☆14Updated 5 years ago
- WebRTC with libnice and srtp，use libnice and srtp to achieve WebRTC☆14Updated last month
- ☆39Updated 7 years ago
- [ABANDONED] Janus Gateway Mobile SDK☆75Updated 4 years ago
- android_live555_rtsp_client☆20Updated 5 years ago
- Implements Janus gateway video room on Android☆68Updated 3 years ago
- the webrtc client for the janus webrtc gateway☆36Updated 6 years ago
- AMediaCodec 安卓ndk硬编解码h264 acc，以及推流RTMP☆18Updated 7 years ago
- RTMP edge speed with janus-gateway☆47Updated 4 years ago
- 1）完成H264、H265、G711和AAC几种音视频编码格式数据到RTP、PS、TS包的封装处理 2）解析获取RTP、PS、TS包中的裸数据☆57Updated 5 years ago
- An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.☆35Updated 5 years ago
- ☆83Updated 5 years ago
- Baresip Applications Modules☆19Updated 3 weeks ago
- a gb28181 sip sdk based on resiprocate☆30Updated 7 years ago
- SIP UserAgent(UAS and UAC) Sample☆39Updated 8 years ago
- get vedio&audio ES from RTP packet over UDP from GBT28181 IPC☆31Updated 9 years ago
- The full C++ implementation of mediasoup☆35Updated 5 years ago