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.
Related
I am trying to build a web application which will need to have audio streaming functionality implemented in some way. Just to give you guys some context: It is designed to be a purely auditive experience/game/idkhowtocallit with lots of different sound assets varying in length and thus file size. The sound assets to be provided will consist of ambient sounds, spoken bits of conversation, but also long music sets (up to a couple of hours). Why I think I won't be able to just host these audio files on some server or CDN and serve them from there is, because the sound assets will need to be fetched and played dynamically (depending on user interaction) and as instantly as possible.
Most importantly, consuming larger files (like the music sets and long ambient loops) as a whole doesn't seem to be client-friendly at all to me (used data consumption on mobile networks and client-side memory usage).
Also, without any buffering or streaming mechanism, the client won't be able to start playing these files before they are downloaded completely, right? Which would add the issue of high latencies.
I've tried to do some online research on how to properly implement a good infrastructure to stream bigger audio files to clients on the server side and found HLS and MPEG-DASH. I have some experience with consuming HLS players with web players and if I understand it correctly, I would use some sort of one-time transformation process (on or after file upload) to split up the files into chunks and create the playlist and then just serve these files via HTTP. From what I understand the process should be more or less the same for MPEG-DASH. My issue with these two techniques is that I couldn't really find any documentation on how to implement JavaScript/TypeScript clients (particularly using the Web Audio API) without reinventing the wheel. My best guess would be to use something like hls.js and bind the HLS streams to freshly created audio elements and use these elements to create AudioSources in my Web Audio Graph. How far off am I? I'm trying to get at least an idea of a best practice.
To sum up what I would really appreciate to get some clarity about:
Would HLS or MPEG-DASH really be the way to go or am I missing a more basic chunked file streaming mechanism with good libraries?
How - theoretically - would I go about limiting the amount of chunks downloaded in advance on the client side to save client-side resources, which is one of my biggest concerns?
I was looking into hosting services as well, but figured that most of them are specialized in hosting podcasts (fewer but very large files). Has anyone an opinion about whether I could use these services to host and stream possibly 1000s of files with sizes ranging from very small to rather large?
Thank you so much in advance to everyone who will be bothered with helping me out. Really appreciate it.
Why I think I won't be able to just host these audio files on some server or CDN and serve them from there is, because the sound assets will need to be fetched and played dynamically (depending on user interaction) and as instantly as possible.
Your long running ambient sounds can stream, using a normal HTMLAudioElement. When you play them, there may be a little lag time before they start since they have to begin streaming, but note that the browser will generally prefetch the metadata and maybe even the beginning of the media data.
For short sounds where latency is critical (like one-shot user interaction sound effects), load those into buffers with the Web Audio API for playback. You won't be able to stream them, but they'll play as instantly as you can get.
Most importantly, consuming larger files (like the music sets and long ambient loops) as a whole doesn't seem to be client-friendly at all to me (used data consumption on mobile networks and client-side memory usage).
If you want to play the audio, you naturally have to download that audio. You can't play something you haven't loaded in some way. If you use an audio element, you won't be downloading much more than what is being played. And, that downloading is mostly going to occur on-demand.
Also, without any buffering or streaming mechanism, the client won't be able to start playing these files before they are downloaded completely, right? Which would add the issue of high latencies.
If you use an audio element, the browser takes care of all the buffering and what not for you. You don't have to worry about it.
I've tried to do some online research on how to properly implement a good infrastructure to stream bigger audio files to clients on the server side and found HLS and MPEG-DASH.
If you're only streaming a single bitrate (which for audio is usually fine) and you're not streaming live content, then there's no point to HLS or DASH here.
Would HLS or MPEG-DASH really be the way to go or am I missing a more basic chunked file streaming mechanism with good libraries?
The browser will make ranged HTTP requests to get the data it needs out of the regular static media file. You don't need to do anything special to stream it. Just make sure your server is configured to handle ranged requests... most any should be able to do this right out of the box.
How - theoretically - would I go about limiting the amount of chunks downloaded in advance on the client side to save client-side resources, which is one of my biggest concerns?
The browser does this for you if you use an audio element. Additionally, data saving settings and the detected connectivity speed may impact whether or not the browser pre-fetches. The point is, you don't have to worry about this. You'll only be using what you need.
Just make sure you're compressing your media as efficiently as you can for the required audio quality. Use a good codec like Opus or AAC.
I was looking into hosting services as well, but figured that most of them are specialized in hosting podcasts (fewer but very large files). Has anyone an opinion about whether I could use these services to host and stream possibly 1000s of files with sizes ranging from very small to rather large?
Most any regular HTTP CDN will work just fine.
One final note for you... beware of iOS and Safari. Thanks to Apple's restrictive policies, all browsers under iOS are effectively Safari. Safari is incapable of playing more than one audio element at a time. If you use the Web Audio API you have more flexibility, but the Web Audio API has no real provision for streaming. You can use a media element source node, but this breaks lock screen metadata and outright doesn't work on some older versions of iOS. TL;DR; Safari is all but useless for audio on the web, and Apple's business practices have broken any alternatives.
I'm looking for some suggestions or pointers on where to look or how to get started with a project for Windows Phone 8.1. The idea is pretty simple in my mind. I want to constantly record video to a memory stream only keeping say the last five seconds, then an event will trigger saving the video steam to a file on to the phone.
I was originally thinking I could save raw frames to a ring buffer and define the size based on the raw frame size * sample rate. Now I realize that might not work because the video provided by the MediaCapture class will be encoded. Digging on stackoverflow, I came across the idea of using MFTs but it sounds a lot more complicated than I originally had in mind.
Looking around the Development Reference material on MSDN, I'm guessing the MediaCapture class will be my friend. Can I somehow define a fixed size stream for use with MediaCapture.StartRecordToStreamAsync then on my event connect it to MediaCapture.StartRecordToStorageFileAsync? Or perhaps there might be a more appropriate way to do this that I should investigate?
I've spent plenty of time solving this problem, but it looks like I need some help. I have a web conference application which provides ability to stream live video, chat, share documents, draw on a whiteboard, share desktop, etc. And now I want to record everything that happens in taken separately so called webinar, including video and sound. So I'm looking for tools that can help with this goal.
Here's input data:
This is Adobe Flash based application
Using wowza server
Everything should be recorded on server
Many webinars can be in recording mode at the same time
Record should be represented in video (flv, mp4 or whatever)
What I've done so far and what I problems I have:
I have implemented recording on server side. But this is not a video, this is just a list of commands to recreate passed webinar. It works, but has lot's of limitations and problems with rewinding.
And now I'm testing this FLV Encoding library. I created AIR application that starts on server when record is needed, connects to taken webinar and takes screenshots from itself with BitmapData.draw() method. Works pretty neat, but has some limitation that I'm looking help with:
First of all, this is sound problem. I have no idea how to catch all
sounds from all sources in flash. So far from my tests and googling I conclude that SoundMixer.computeSpectrum() won't help me to do this. Maybe this can be done on server side by mixing all streams on the right time but I think this can lead to synchronization problems and I prefer to capture sound on client. Maybe there is way to capture audio byte array from rtmp stream somehow?
Security problems. We have 2 kinds of them. First ones are with streaming videos. BitmapData.draw() method throws exeptions even after adding <StreamAudioSampleAccess>true</StreamAudioSampleAccess>
<StreamVideoSampleAccess>true</StreamVideoSampleAccess> on server. There are lots of posts about this problem and no good solution.
But more complex problem is that YouTube videos can be opened in webinar using api player. And in this situation I have no idea how to resolve security problem. Maybe someone knows a way or workaround to use BitmapData.draw() on YouTube AS3 player?
Or maybe there is another good way to solve my recording issue?
Free Apache Openmeetings conferencing [1] has a java recording application inside which should work in 3.0 release. Just use it.
[1] http://openmeetings.apache.org/
I'm streaming a video from an Amazon CloudFront RTMP source with
video.attachNetStream(myNetStream);
myNetStream.play(myVideoFileName,0,-1);
and it's working quite well. Now, what I want to do is something like this:
video1.attachNetStream(myNetStream);
video2.attachNetStream(myNetStream);
myNetStream.play(myVideoFileName,0,-1);
This doesn't work as written because only one of the two videos will play at a time for some unknown reason. I want video1 and video2 to play the same video from a single NetStream (to save bandwidth) and remain completely in-sync with each other. How can I accomplish this?
If you are playing an FLV file directly (not streaming it from FMS), you should be able to :
Load the file with URLStream
Wait for sufficient data to start copying the data to a ByteArray object
Create as many NetStream objects as you need and use the appendBytes(bytes) method
I haven't actually tested it, and the logic of appendBytes() needs to be looked at, but theoretically it should work.
Also, it deserves a benchmark. But it's probably better than re-drawing a bitmap copy at the same rate as the video, and keep the two videos in sync.
bitmapData can't be accessed at all because of security restrictions. I'm going with plan B which is playing 2 netstreams, but reducing the file size of the one the videos by removing its audio. I'll have to wait until Amazon allow security policy access to use the bitmapData solution.
I have a custom Flash video player that uses the NetConnection and NetStream classes to stream mp4 files from a Wowza server.
Lately the playback has been having audio problems. The sound is crackled and in some cases may cause the Flash player to crash. This is not consistent between files. Meaning that a file that was played once with bad audio could be fine 5 minutes later when played again from a new instance of the player , but 5 minutes later have the same problem.
My thought direct me to thinking this was an encoding problem, but that wouldn't explain how one second the playback is fine, and once refreshed messes up again.
I should note that the Wowza server does not stream the entire file but only a portion of it based on various parameters it gets.
Any help would be greatly appreciated.
Do you get the audio crackle when you use a standard player and/or a different client machine? My first thought is that this is a custom player or client sound processing issue.
If this reproduces on any player or client, then we should also troubleshoot the server. If you end up thinking it might be server-related, please post a message on the Wowza forum: http://www.wowza.com/forums/forum.php.