I'm in testing stage of launching an online radio. I'm using AWS CloudFormation stack with Adobe Media Server.
My existing instance type is m1.large and my Flash Media Live Encoder is streaming mp3 at 128kbps which i think is pretty normal but it's producing a stream that isn't smooth & stable at all and seems to have a lot of breaks.
Should i pick an instance type with higher specs?
I'm running my test directly off of LiveHLSManifest link that opens on my iPhone's Safari and plays on browser's build-in player..which doesn't set any buffering on client side - could this be the issue?
Testing HLS/HDS links directly on iPhone's Safari was a bad idea. I relied on built-in players already having some sort of buffering configuration by default but noo... I was able to receive stable & smooth stream when i used players like Strobe Media Playback, FlowPlayer etc.. Hopefully, this answer will save someone some time.
Related
I am trying to loop audio from my Icecast server 24/7.
I have seen examples where people talk about storing their audio files on the EC2 instance or in an S3 bucket.
Do I also need a source client running on my EC2 Instance to be able to stream audio to the server? Or is there a way to play static files from Icecast?
Icecast and SHOUTcast servers work by passing a live audio stream from a source on to the users. You need something to produce a single audio stream in realtime from those source files.
The flow looks something like this:
Basically, you'll need to do everything you would in a normal radio studio, but automated. You'll stream the files from your bucket, play them to a raw audio stream, send that stream to your encoder to be compressed with the codec, and then sent to your streaming servers for distribution.
You can't simply push your audio files as-is to the Icecast server, for a few reasons:
Stream must be realtimeThe server doesn't really know or care about the timing of the stream. It takes the data its given and sends that off to the client. Therefore, if you push data faster than realtime, the server will attempt to deliver it to the client at this faster rate. Some clients will attempt to buffer this fast stream, but most will put backpressure on the stream, causing the TCP window to close, causing the client to eventually get far enough behind that the server drops the connection.
Consistent format is requiredChances are, your source files have varying sample rate, channel count, and even codec. Most clients are unable to take a change in sample rate or channel count mid-stream. I don't know of any client that supports a codec change mid-stream. (Theoretically possible with Ogg and Matroska/WebM, but yeah... not worth messing with.)
Stream should be free of ID3 tags and other file format cruftIf you simply PUT your files directly to your Icecast server, the output stream will contain more than just the audio data. At a minimum, you'd want to remove all that. Depending on your container format, you'll need to deal with timestamps as well.
Solutions
There are a handful of ways to solve this:
Radio automation softwareMany folks simply run something like RadioDJ on cloud-based servers. If you already have a radio station that uses automation, this might be a good solution. It can be expensive though, and not as flexible. You could even go as low as VLC or something for playout, but then you wouldn't have music transitions and what not.
Custom playout script (recommended)I use a browser engine, such as Chromium, and script my channels with normal JavaScript. From there, I take the output stream and pass it off to FFmpeg to encode and send to the streaming servers. This works really well, as I can do all my work in a language everybody knows, and I have easy access to data on cloud-hosted services. I can use the Web Audio API to mix and blend audio based on what's happening in realtime. As an alternative, there is Liquidsoap, but I do not recommend it these days as its language is difficult to deal with and it is not as flexible as a browser engine.
I am working on a live-streaming prototype, I have been reading a lot about how live-streaming works and many different approaches but I still can't find a live-streaming stack that suits my needs...
These are the requirements for my prototype:
1)The video/audio recording must come from a web browser using the webcam, the idea is that the client preferably shouldn't need to install plugins or do anything complicated(maybe installing Flash player plugin is acceptable, only for recording the video, the viewers should be able to view the stream without plugins).
2)It can't be peer to peer since I also need to store the entire video in my server (or in Amazon s3 servers for example) for viewing later.
3)The viewers should also be able to watch the stream without the need of installing anything, from their web browsers, say Chrome and Firefox for example. We want to use the HTML5 video tag if possible.
4)The prototype should be constructed without expending money preferably. I have seen that AWS-Cloudfront and Wowza offer free trials so we are thinking about using these 2 services.
5)The prototype should be able to maintain 1 live stream at a time and 2 viewers, just that, so there are no restrictions regarding this.
Any suggestions?
I am specially stuck/confused with the uploading/encoding video part of the architecture(I am new to streaming and all the formats/codecs/protocols/technologies are making it really hard to digest).
As of right now, I came across WebRTC that apparently allows me to do what I want, record and encode video from the browser using the webcam, but this API only works with HTTPS sites. Are there any alternatives that work with HTTP sites?
The other part that I am not completely sure about is the need for an encoding server, for example Wowza Streaming Engine, why do I need it? Isn't it enough if I use for example WebRTC for encoding the video and then I just send it to the distribution service (AWS-Cloudfront for example)? I do understand that the encoding server would allow me to support many different devices since it will create lots of different encodings and serve many different HTTP protocols, but do I need it for this prototype? I just want to make a 1 format (MP4 for example) live-stream that can be viewed in 2 web browsers, that's all, I don't need variety of formats nor support for different bandwidths or devices.
Base on your requirement, WebRTC is good way.
API only works with HTTPS sites. Are there any alternatives that work
with HTTP sites?
No. Currently Firefox is only browser is allow WebRTC on HTTP, but finally it need HTTPS
For doing this prototype you need to go with the Wowza WebRTC.
While going with wowza all the streams are delivered from the wowza only.So it become a routed WebRTC.
Install Wowza - https://www.wowza.com/docs/how-to-install-and-configure-wowza-streaming-engine
Enable the WebRTC - https://www.wowza.com/docs/how-to-use-webrtc-with-wowza-streaming-engine
Downaload and configure the Streamlock. or Selfsigned JKS file - https://www.wowza.com/docs/how-to-request-an-ssl-certificate-from-a-certificate-authority
Download the sample WebRTC - https://www.wowza.com/_private/webrtc/
Publish stream using the Publish HTML and Play through the Play HTML ( Supported Chrome,Firefox & Opera Browsers)
For MP4 files in WebRTC : you need to enable the transcoder with h264 & aac. Also you need to enable the option Record all the incoming Streams in the properties of application which you are creating for the WebRTC ( Not the DVR ).Using the File writer module save all the recorded files in a custom location.By using a custom script(Bash,Python) Move all the Transcoded files to the s3 bucket, Deliver through cloudfront.
I am working on a face recognition project using Flask as my web server running on a Ubuntu 14.04 Machine. I am using OpenCV 2.4.9 as my image processing software which is written using Python2.7. I would like to be able to access a clients webcam through their browser to capture a image or frame from the webcam stream and send it back to the server to be processed. Is there an easy way using python to obtain access to the clients webcam or is it possible to use JavaScript in conjunction with my current code.
I'll assume that you are more interested in architectural decisions for you application that specific implementation details. You will need to use client side and server side for this application.
Client side is html page with javascript that will capture images from web cam. There are many resources on internet about this topic. This article explains how it works with some examples. I would recommend to use some javascript library like this one
The next thing is to decide how client application and server side transfers image data. In case you would like to stream webcam video to server, do some computation and stream data back to client application, WebSockets are your friend. This tutorial describes how to set up flask application for websockets.
Much easier approach is to POST image data to the server, do some computation and respond to client. Downside of this approach is that it's not suitable for continuous video processing. But you can use it for single video frame processing. Otherwise you would flooded your server with requests.
The last thing to decide is how much processing is done to images on server side. If you would do some extensive computation that takes long time, I would recommend celery for background tasks. HOWEVER this would change architecture considerably.
For a proof of concept, I would recommend following. Take single image with webcam, post it to server, do quick computation on image and respond with what you've had computed.
Good luck.
I want to create a service that streams live traffic video to either a client browser or a client processor (which will actually process the video). I want real video, not just images that update periodically. Assume I know the basic concepts of web design (both front and back end). But assume I know nothing about streaming media.
Can someone point me in the direction to get started?
I need information concerning software, frameworks (especially if it's compatible with Ruby on Rails), encoders, converters, protocols, ... - thanks!
What about something like tokbox?
http://www.tokbox.com/
I havent personally used it. However I have visited an on line video podcast that uses this technology. Good quality streaming.
So we have some server with some address port and ip. we are developing that server so we can implement on it what ever we need for help. What are standard/best practices for data transfer speed management between C++ windows client app and server (C++)?
My main point is in how to get how much data can be uploaded/downloaded from/to client via his low speed network to my relatively super fast server. (I need it for set up of his live stream Audio/Video bit rate)
My try on explaining number 3.
We do not care how fast is our server. It is always faster than needed. We care about client tyring to stream out to our server his media. he streams encoded (via ffmpeg) live video data to our server. But he has say ADSL with 500kb/s of outgoing traffic. Also he uses some ICQ or what so ever so he has less than 500 kb/s per second. And he wants to stream live video! So we need to set up our ffmpeg to encode video with respect to the bit rate user can provide. We develop server side and client side. We need a way of finding out how much user can upload per second currently (so value can change dynamically over time)
Check this CodeProject Article
it's dot-net but you can try figure out the technique from there.
I found what I wanted. "thrulay, network capacity tester" A C++ code library for Available bandwidth tracking in real time on clients. And there is "Spruce" and it is also oss. It is made using some of linux code but I use Boost library so it will be easy to rewrite.
Offtop: I want to report that there is some group of people on SO down voting on all questions on this topic - I do not know why they are so angry but they deffenetly exist.