Alternatives and similar repositories for rtaudio_speex

Users that are interested in rtaudio_speex are comparing it to the libraries listed below

• zhiyong0804 / rtsp_pusher
  ☆16Updated 8 years ago
• KingsleyYau / WebRTC-Server
  LiveStream gateway for WebRTC 流媒体网关服务器
  ☆25Updated last week
• wukehu / rtsp
  rtsp-server rtsp-client
  ☆14Updated 8 years ago
• dq5070410 / native_webrtc
  a video engine include receiver and sender base on webrtc
  ☆12Updated 8 years ago
• zhangpengyf / foxrtc
  media sdk based on webrtc
  ☆40Updated 2 years ago
• X2Rtc / X2Rtc-Stream
  Open source, Open mind.
  ☆19Updated last year
• sambios / webrtc_vs2015
  VS2015 webrtc bulid solution, now open source for everyone.
  ☆15Updated last year
• ygliang2009 / boeing
  webrtc network part for reliable udp transport
  ☆31Updated 5 years ago
• zhenwu1981 / encode_mp4
  encode mp4 form H264 and AAC
  ☆25Updated 7 years ago
• peterfuture / dtplayer_c
  simple cross-platform multimedia player and framework
  ☆27Updated 3 years ago
• escoffier / sipsdk
  a gb28181 sip sdk based on resiprocate
  ☆30Updated 7 years ago
• QuickGBLink88 / StreamRecorder
  Uses ffmpeg to receive rtsp/rtmp/HLS, and save stream data to a MP4 file.
  ☆20Updated 5 years ago
• bingqingsuimeng / SipClient
  sip client，gb28181 client
  ☆24Updated 6 years ago
• larkguo / SIP_UA
  SIP UserAgent(UAS and UAC) Sample
  ☆39Updated 8 years ago
• lhwanglj / CodecMixer
  mix multi audio stream and multi video stream to one audio stream and one video stream and send it to rtmp server.
  ☆16Updated 9 years ago
• cheungreed / StreamResolver
  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
• Achillesxu / GBT28181-RTP-PS-ES
  get vedio&audio ES from RTP packet over UDP from GBT28181 IPC
  ☆31Updated 9 years ago
• pcgpcgpcg / janus_gateway_win
  the webrtc client for the janus webrtc gateway
  ☆36Updated 6 years ago
• Mihawk086 / webrtc-server
  webrtc流媒体服务器
  ☆46Updated 5 years ago
• TaoistKing / Video-Jitter-Buffer
  Video Jitter Buffer derived from WebRTC
  ☆16Updated 6 years ago
• fonnes / flexible_fec
  An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.
  ☆35Updated 5 years ago
• prudens / audioengine
  a cross-platform audio engine,including audio device ,audio processing audio codec,etc.
  ☆14Updated 7 years ago
• gongluck / 3rdparty
  三方库源码、编译脚本、二进制输出等。openssl、libsrtp、zlmediakit
  ☆25Updated last year
• redknotmiaoyuqiao / EyerGB28181
  ☆17Updated 4 years ago
• AnSwErYWJ / AudioResamplerate
  This is an audio resamplerate program based on Secret Rabbit Code and iniparser.
  ☆23Updated 4 years ago
• wexiangis / wmix
  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
  ☆15Updated 3 years ago
• ouxianghui / mediasoup-server
  mediasoup c++ server (SFU)
  ☆14Updated last year
• ireader / 3rd
  ☆31Updated 2 years ago
• sureone / ffmpeg_steam
  push a rtsp streamer to rtmp nginx server
  ☆16Updated 7 years ago
• lcmftianci / libonvif
  discover device and open rtsp url
  ☆18Updated 6 years ago