libstreaming is an API that allows you, with only a few lines of code, to stream the camera and/or microphone of an android powered device using RTP over UDP.
https://github.com/fyhertz/libstreamingTags | android h264 rtsp-server aac amr mediarecorder mediarecorder-api rtp mediacodec |
Implementation | Java |
License | Apache GPL |
Platform | OS-Independent |
rtsp-simple-server is a ready-to-use and zero-dependency server and proxy that allows users to publish, read and proxy live video and audio streams through various protocols like RTSP, RTMP, HLS. It publish and read live streams to the server. It acts as a proxy and serve streams from other servers or cameras, always or on-demand.
audio streaming h264 video rtsp hls rtmp rtsp-server aac media-server rtmp-server rtp h265 obs-studio rtcp rtsp-relay rtmp-proxy rtsp-proxy video-streamingThis sample uses the camera/camcorder as the A/V source for the MediaRecorder API. A TextureView is used as the camera preview which limits the code to API 14+. This can be easily replaced with a SurfaceView to run on older devices. This sample shows how to use the MediaRecorder API. It uses the Camera as input source and displays a preview on a TextureView The sample features a button to capture the input and stop capturing afterwards.
RecordRTC is WebRTC JavaScript library for audio/video as well as screen activity recording. It supports Chrome, Firefox, Opera, Android, and Microsoft Edge. Platforms: Linux, Mac and Windows.
webrtc recordrtc mediarecorder mediastreamrecorder mediarecorder-api record-audio record-video webrtc-recording record-screen video-streaming audio-streamingThis library generate an Mp4 movie using Android MediaCodec API and apply filter, scale, and rotate Mp4.
android-library opengles video-editor video-filtering mediacodec glsl-shader h264 shader androidRTSP Server for V4L2 device capture supporting HEVC/H264/JPEG/VP8/VP9
rtsp-server v4l2 rtsp c-plus-plus hls v4l2-devicePion is the Modern Stack for Web Real-Time Communication. It implements the WebRTC API. Spend more time building and less time learning a new API. Its feature include PeerConnection API support for DataChannels, Send/Receive audio and video, Renegotiation. It provides API with direct RTP/RTCP access, API also allows developer to pass their own packetizer, Opus, PCM, H264, VP8 and VP9 packetizer, Sender/Receiver Reports, Easy integration with x264, libvpx, GStreamer and ffmpeg and lot more.
audio golang streaming video webrtc p2p webassembly wasm webcam voip ortc rtp srtp webrtc-libraries webrtc-api pion pion-webrtc live-video webrtc-serverThe simple demonstration of Android MediaExtractor and MediaCodec API. How to play video with Android MediaExtractor and MediaCodec API, with a very simple clock to sync video playback with the original FPS.
Android based RTSP Server which is able to serve live camera view to multiple RTSP clients, such as VLC. This project is not maintained anymore (in fact since 2end of 2012). It exists to share the code how to implement this back in the days.
Inspired by Audacity, this project is a multiple track playlist editor written in ES2015 using the Web Audio API. Load tracks and set cues (track cue in, cue out), fades (track fade in, fade out) and track start/end times within the playlist. I've written up some demos on github for the different audio fade types in the project.
playlist waveform audio audio-visualizer audio-player annotations audiorecorder webaudio music mediarecorder harmony playback daw audacity stem tracks multitrack editor record recording playerNode Media Server is a Node.js implementation of RTMP / HTTP-FLV / WS-FLV / HLS / DASH Media Server. It supports H.264 / H.265(flv_id=12) / AAC / MP3 / SPEEX / NELLYMOSER / G.711 / OPUS(flv_id=13), GOP cache, Rtsp / Rtmp relay, Real-time multi-resolution transcoding.
nodejs flv rtmp websocket livestream media-server hls mpeg-dash server国内外为数不多致力于极致体验的超强全自研跨平台(windows/android/iOS)流媒体内核,通过模块化自由组合,支持实时RTMP推流、RTMP播放器、RTSP播放器、录像、多路流媒体转发、音视频导播、动态视频合成、音频混音、直播互动、内置轻量级RTSP服务等,比快更快,业界真正靠谱的超低延迟直播SDK(1秒内,低延迟模式下200~400ms)。
rtmp-pusher rtmp-player android-rtmp ios-rtmp windows-rtmp daniulive rtsp-player rtmp-client android-publisher rtsp2rtmp player rtmp rtsp rtsp-relay rtmp-h265 hevc rtmpclient rtspclient rtmp-broadcaster rtsp-to-rtmpMedia streaming server based on nginx-rtmp-module. 中文说明.
rtmp http-flv gop-cache media-server nginx-rtmp virtual-hosts nginx-http-flv chunked-transmission live-streaming h264 aac flvjs dynamic-moduleThis repository contains three simple examples of how to use libstreaming. You can find out more about libstreaming here.
Allowing developers to aggregate media from Rtsp sources over Rtsp without degrading the source bandwidth. Agnostic of Video or Audio format. Decodes Jpeg / RTP
aggregation audio ieee1733 media rfc2032 rfc2326 rfc2435The KOM(S) Streaming System (komssys) implements a streaming system based on the IETF protocols RTSP, SDP, RTP/RTCP with the intention of providing a base for researchers and other developers. Komssys includes code for a server, a client, and a proxy
This project provides a free Raspbian based Raspberry Pi image with Homebridge and Homebridge Config UI X pre-installed. This image also provides a command called hb-config which helps you keep Node.js up-to-date, perform maintenance on your Homebridge server, and install additional optional software such as Pi Hole, deCONZ, Node-RED and the UniFI Controller.
raspberry-pi ui ffmpeg smarthome homebridge homekit raspberry-pi-3 raspberry-pi-2 raspberry-pi-zero-w homekit-bridge libfdk-aac wifi-setup raspberry-pi-4 homebridge-ui h264-omxHardware accelerated transcoder for Android, written in pure Java. Android does not offer straight forward way to transcode video.
This package provides an implementation of the Secure Real-time Transport Protocol (SRTP), the Universal Security Transform (UST), and a supporting cryptographic kernel. The SRTP API is documented in include/srtp.h, and the library is in libsrtp2.a (after compilation). This document describes libSRTP, the Open Source Secure RTP library from Cisco Systems, Inc. RTP is the Real-time Transport Protocol, an IETF standard for the transport of real-time data such as telephony, audio, and video, defined by RFC 3550. Secure RTP (SRTP) is an RTP profile for providing confidentiality to RTP data and authentication to the RTP header and payload. SRTP is an IETF Standard, defined in RFC 3711, and was developed in the IETF Audio/Video Transport (AVT) Working Group. This library supports all of the mandatory features of SRTP, but not all of the optional features. See the Supported Features section for more detailed information.
We have large collection of open source products. Follow the tags from
Tag Cloud >>
Open source products are scattered around the web. Please provide information
about the open source projects you own / you use.
Add Projects.