is it possible to create a live audio stream based on other audio streams? I'm thinking of a proxy that gets two audio streams (e.g. shoutcast stream), and based on time, switches to one of them. And, if its possible, to have some time for analysis, I would implement some kind of caching so that I can stream the newly created stream time-displaced.
I already had a look on the Shoutcast server but couldn't figure out, how to config the input source as another stream. Maybe there are other projects that can handle this through a interface.
Programming language don't really matters, but Ruby is prefered.
I made my own solution with javascript based on nodejs. You can clone the repo (a personal radio app) at https://github.com/23tux/personal_node_radio.
Related
I'm trying to stream the video of my C++ 3D application (similar to streaming a game).
I have encoded an H.264 video stream with the ffmpeg library (i.e. internally to my application) and can push it to a local address, e.g. rtp://127.0.0.1:6666, which can be played by VLC or other player (locally).
I'm not particularly wedded to h.264 at this point, or rtp. I could send as srtp if that would help.
I'd like to use WebRTC to set up a connection across different machines, but can't see in the examples how to make use of this pre-existing stream - the video and audio examples are understandably focused on getting data from devices like connected web cams, or the display.
Is what I'm thinking feasible? I.e. ideally I'd just point webRTC at my rtp://127.0.0.1:6666 address and that would be the video stream source.
I am writing out an sdp file as well which can be read by VLC, could I use this in a similar way?
As noted in the comment below there is an example out there using go to weave some magic that enables an rtp stream to be shown in a browser via webRTC.
I am trying to find a more "standard" way to be able to set the source of a video track in webRTC to be the URL of an encoded stream. If there isn't one, that is valuable information to me too, as I can change tack and use a webrtc library to send frames directly.
Unfortunately FFMPEG doesn't support WebRTC output. It lacks support for ICE and DTLS-SRTP.
You will need to use a RTP -> WebRTC bridge. I wrote rtp-to-webrtc that can do this. You can do this with lots of different WebRTC clients/servers!
If you have a particular language/paradigm that you prefer happy to provide examples for those.
As a beginner at working with these kinds of real-time streaming services, I've spent hours trying to work out how this is possible, but can't seem to work out I'd precisely go about it.
I'm prototyping a personal basic web app that does the following:
In a web browser, the web application has a button that says 'Stream Microphone' - when pressed it streams the audio from the user's microphone (the user obviously has to consent to give permission to send their microphone audio) through to the server which I was presuming would be running node.js (no specific reason at this point, just thought this is how I'd go about doing it).
The server receives the audio close enough to real-time somehow (not sure how I'd do this).
I can then run ffmpeg on the command line and take the real-time audio coming in real-time and add it as the sound to a video file (let's just say I'm going to play testmovie.mp4) that I want to play.
I've looked at various solutions - such as maybe using WebRTC, RTP/RTSP, Piping audio into ffmpeg, Gstreamer, Kurento, Flashphoner and/or Wowza - but somehow they look overly complicated and usually seem to focus on video along with audio. I just need to work with audio.
As you've found there are numerous different options to receive the audio from a WebRTC enabled browser. The options from easiest to more difficult are probably:
Use a WebRTC enabled server such as Janus, Kurento, Jitsi (not sure about wowzer) etc. These servers tend to have plugin systems and one of them may already have the audio mixing capability you need.
If you're comfortable with node you could use the werift library to receive the WebRTC audio stream and then forward it to FFmpeg.
If you want to take full control over the WebRTC pipeline and potentially do the audio mixing as well you could use gstreamer. From what you've described it should be capable of doing the complete task without having to involve a separate FFmpeg process.
The way we did this is by creating a Wowza module in Java that would take the audio from the incoming stream, take the video from wherever you want it, and mix them together.
There's no reason to introduce a thrid party like ffmpeg in the mix.
There's even a sample from Wowza for this: https://github.com/WowzaMediaSystems/wse-plugin-avmix
Preface
I have read this two part tutorial (Part-1 and Part-2) by Steamroot on MPEG-DASH, and below is my understanding (please correct me if I am wrong):
The video needs to be encoded into multiple bit-rates using FFmpeg.
The encoded videos need to be transcoded (dashified) using MP4Box.
The dashified videos can be served using a web server.
Problem
I intend to live-stream an event and I need help to understand the following:
Can I club the FFmpeg and MP4Box commands into a single step? Maybe through a wrapper program so that I do not have to run them separately? Is there any other or better solution?
How do I send the dashified content to the web server? FTP? Would any vanilla web server do?
Lastly, a friend had hinted that I could also use GStreamer to achieve my objective. But, I could not find any good resource on the internet for the same. So, where (and how) does GStreamer fit in the above process?
What is the format you will be getting out of your camera for your live-event? There are a lot of solutions a lot more adapted for live streaming (the tutorial I wrote is for VOD streams only). You can check out simple solutions like Wowza Streaming Server, Nible streamer (free), etc, that take a RTMP stream and transform it into other formats (HLS, DASH, etc...).
Most of the livestreaming platforms can even do that for you (livestream.com, youtube, twitch, or even facebook now)
The dashified content will be requested as HTTP ressources by the browser or other players. In the case of a VoD stream, indeed you just need to make the dash segments available through a web-server. For live content, you need something smarter, that will encode, package the segments and make them available on the fly.
Gstreamer can transcode and transmux the original content, and can do it on the fly. You will be able to get different formats as outputs, like RTMP, HLS, and probably even mpeg-dash. Then you still need to make your content available via a webserver.
In conclusion, if you just want to transmit an occasional live event, it's probably a lot easier a platform that will ingest your RTMP stream and do all the complicated steps for you.
idea is to create a video from images provided by a user and at the same time stream the generated video to other user demanding it.
kindly tell any efficient way to do this and which language out of PHP and C# .net will be suitable.
have looked into ffmpeg to take images and convert to video and save to server and then stream .. kindly tell if this the possibility or any other method for live streaming.
regards
UPATE
consider the following scenario as I understand:
get images from server and start combining them to form a video. at the same time, stream the video to the users requesting it.. for new coming clients, stream the previously generated video from the begining and keep on sending the new video which is being generated from images to the previous clients.
kindly tell if this is possible, if so then what can be the approach. Have read something about pipes but am completely new to ffmpeg and streaming in general.
Yes, this is possible with ffmpeg. Any language that is turing complete is suitable. They are many methods of live streaming including HLS, RTP, RTMP, etc.
If you need more detailed answers. Please ask more detailed questions.
I would like to achieve the following:
Set up a proxy server to handle video requests by clients (for now, say all video requests from any Android video client) from a remote video server like YouTube, Vimeo, etc. I don't have access to the video files being requested, hence the need for a proxy server. I have settled for Squid. This proxy should process the video signal/stream being passed from the remote server before relaying it back to the requesting client.
To achieve the above, I would either
1. Need to figure out the precise location (URL) of the video resource being requested, download it really fast, and modify it as I want before HTTP streaming it back to the client as the transcoding continues (simultaneously, with some latency)
2. Access the raw byte stream, pipe it into a transcoder (I'm thinking ffmpeg) and proceed with the streaming to client (also with some expected latency).
Option #2 seems tricky to do but lends more flexibility to the kind of transcoding I would like to perform. I would have to actually handle raw data/packets, but I don't know if ffmpeg takes such input.
In short, I'm looking for a solution to implement real-time transcoding of videos that I do not have direct access to from my proxy. Any suggestions on the tools or approaches I could use? I have also read about Gstreamer (but could not tell if it's applicable to my situation), and MPlayer/MEncoder.
And finally, a rather specific question: Are there any tools out there that, given a YouTube video URL, can download the byte stream for further processing? That is, something similar to the Chrome YouTube downloader but one that can be integrated with a server-side script?
Thanks for any pointers/suggestions!
You should ask single coding questions. What you asked is more like a general "how would a write my application". A few comments though:
squid is a http proxy, video use usually streamed over e.g. rtsp.
yes there are tools that grab the rtsp url from a youtube url, be sure to understand the terms of use for the video servie before going that way though.
gstreamer has a gst-rtsp-server module that contains a rtsp server, that also can be used as a proxy for a given rtsp stream.