I need to record audio through the HTML5, but you need to write directly in MP3. The reason is that in the standard version of the sound is written in WAV, 5 minutes = 500Mbit - this is a very high resolution. Write to WAV and then recode also tried (and client-side and server-side) - it is a very expensive process and takes a lot of time and resources (5 minutes of recording time is 5 minutes).
I need to write the sound directly into the format of MP3, convert for me is not suitable.I would be very grateful for any help.
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I'm trying to make a live streaming system without P2P connection by nodejs.
Video chunks are recorded each 3 seconds and sent to server via webrtc and they save simply as .wemb base64 files.
But when the client is trying to see the whole video, there is flickering problem.
Each time a 3 secs video ends the src attribute of video tag changes to next chunk.
I tried to save some upcoming ones to a buffer in order to prevent delays, but still it is considerable.
I used MediaSource API to add source buffer but id didn't work on that specific webm encoding either.
The codes are huge and complicated and doesn't help much, but if they are necessary in order to correct the concept I can put then too.
Thanks.
I need to develop something similar, but all I've got so far is a Chrome app which uses Whammy.js for encode de webm images from a desktopCapture stream and encode them to a .webm video, but it is extremely slow (almost 5 minutes for 30 seconds) and I can't record the system sound.
I tested this Screencastify app and I think it does a pretty decent job, it records even in fullHd fast and also can record the system sound. But how does this work? AFAIK Chrome doesn't have an API for recording the system sound and encode video that quickly.
Screencastify uses Native Client.
I have thousands of mp3 files in a database and a website that lets people listen to them. I'm using a flash player but want to move to html5 audio player. Does this mean that I need to make ogg and wave versions of all my audio files? What is a good approach to making these files more accessible?
In short, yes you need to support multiple formats. (Assuming you care about decent browser support.)
If you are lacking disk space, don't get a lot of traffic, and don't mind some delay before the data gets to the user, you can convert these on the fly. Just write some code so that on request, it checks the conversion cache to see if you have already converted the file. If not, convert it on the fly (with something like FFMPEG) and write the data to disk at the same time you are writing it to the client.
As Imre pointed out, browser support is changing all the time. Look up what codecs are supported from time to time.
I'm currently working on a dynamic MP3 player in AS3. The player will also support continuous (in length) radio streams.
Because my player will include a 'seek' bar, I allow the user to seek through the Sound object's data. Now I know that with a continuous stream, data being stored on the users RAM will never stop, as downloading will never stop on a continuous stream. This means, after a few hours of streaming, allot of RAM is being used by my app. I've tested the app on my own machine, running a very high spec, and the app crashes in my browser. When i say the app crashes, I mean the whole of Flash, meaning I have to restart my browser in order to use Flash again. I know my app is the cause as Flash has never crashed in the past. It only does it when my app has been streaming for 2+ hours.
So what I want to do is only allow the user to cache up to an hours worth of audio. After an hour, I want to clear the first half of the sound objects data, meaning that only the most recent half hours audio is stored and available for seeking.
So I have my stream:
var soundObj:Sound = new Sound();
soundObj.load(new URLRequest('stream.mp3'));
//ect ect
and sound is where the data is stored. So my question: How would I clear the first 30 mins of audio from that object?
Perhaps the Sound class is not meant to reliably play "unlimited" MP3 files, which seems to be your case. It is made to play normal MP3 "songs". Two hours of MP3 sound can easily accumulate to be larger than 200 megabytes of data.
But there is a good solution - use NetConnection and NetStream classes to stream audio instead. There are many tutorials out there. You will also be able to stream your MP3s, just a bit differently - a central server will be involved, which will transcode these MP3s on the fly, delivering it to you in a true "streaming" manner. One of such servers is Adobe Flash Media Server, an overpriced piece of work from Adobe. A lot of free and open-source alternatives exist which will work fine for your purposes - Red5, nginx-rtmp to name a few, that I have tested myself.
I'm recording a mono audio stream using a PIC at 8-bit 8Khz and streaming it raw to another microprocessor that houses a web server. I'm currently buffering the data and turning it into a wav file that gets played in the browser. What I'd like to be able to do is continuously stream the audio as it's being recorded without putting a lot of encoding overhead on the second processor. I've been searching but most searches turn up just streaming from a stored file, but since the file size isn't known ahead of time I'm not sure how to do this without the overhead of mp3 encoding.
You may find that simply creating a WAV file (or other raw format) that keeps growing will, in most players/browser plugins, cause the file to act as a live stream. This is, I believe, basically how like Ogg streaming and similar works. Because the player begins playing before it is done downloading anyway, it keeps playing and downloading until the end of the file, but the file has no end so it just keeps going.
Vlc media player can stream flv and many other formats.