Preconditions:
Developing an audio player for a web application.
All target browsers fully support audio tag.
No need in sprites, multiple simultaneous sounds etc, just one audio track to be played back at a moment.
Audio file has to be streamed from the server, not downloaded at once. Therefore not Web Audio API.
Why would I want to utilize howler.js or similar library instead of relying on the built-in audio tag in this scenario?
The only howler.js feature that is intriguing is “Handles edge cases and bugs across environments”.
Related
I have been trying to find a way to play live stream of audio (mp3) using Google Actions but haven't found a way to do so.
I tried Media Response as well but as mentioned in the documentation it doesn't support live stream.
I followed this thread but it doesn't have any examples to help me with.
Is it possible to play live mp3 stream using Google Actions?
I've had relatively good results with the Media Player being able to handle mp3 "streams". There are a couple of problems doing this, however:
There is a time limit on the audio playback (4 hours last time I checked, but it may have changed).
There isn't any such thing as an mp3 "stream". The player treats it as a single mp3 file that it downloads in chunks using HTTP headers, unlike some of the streaming protocols that allow for varying bitrate based on network and other conditions.
If this is an issue, one alternative might be to use the Interactive Canvas (which uses Chrome on the device) to present an HTML page that has an <audio> tag in it that you control. This gives you a little more control (most streaming protocols are either supported or have JavaScript libraries that can do the work), but there are some downsides:
This will only work on Smart Displays and Android. Smart Speakers aren't supported.
Interactive Canvas is only allowed for certain types of Actions. Currently it must be a game, a story, or an educational Action.
There is an audio stream which sends from mobile device to the server. And server sends chunks of data (due web-sockets) to the web.
The question is. What to use to play this audio in live mode, also there is should be a possibility to rewind audio back, listen to what was before..and again switch to live mode.
I considered such possibilities as Media Source API but it's not supported by Safari and Chrome on IOS, isn't it? But we need that support.
Also, there is Web Audio API which supports by modern browsers, but I'm not sure does it possible to listen to audio in live mode and rewind audio back?
Any ideas or guides on how to implement it?
I considered such possibilities as Media Source API but it's not supported by Safari and Chrome on IOS, isn't it? But we need that support.
Then, you can't use MediaSource Extensions. Thanks Apple!
And server sends chunks of data (due web-sockets) to the web.
Without MediaSource Extensions, you have no way of using this data from a web socket connection. (Unless it's PCM, or you're decoding it to PCM, in which case you could use the Web Audio API, but this is totally impractical, inefficient, and not something you should pursue.)
You have to change how you're streaming. You have a few choices:
Best Option: HLS
If you switch to HLS, you'll get the compatibility you need, as well as the ability to go back in time and what not. This is what you should do.
Mediocre Option: HTTP Progressive
This is a fine way to stream for most use cases but there isn't any built-in way to handle the stream seeking that you want. You'd have to build it, which is not worth your time since you could just use HLS.
Even More Mediocre Option: WebRTC
You could switch to WebRTC for streaming, but you have greatly increased infrastructure costs and complexity. And, you still need to figure out how you're going to handle seeking. The only reason you'd want to go the WebRTC route is if you absolutely needed the lowest latency.
I am trying to build a website and mobile app (iOS, Android) for the internet radio station.
Website users broadcast their music or radio and mobile users will just listen radio stations and chat with other listeners.
I searched a week and make a prototype with Wowza engine (using HLS and RTMP) and SHOUTcast server on Amazon EC2.
Using HLS has a delay with 5 seconds, but RTMP and SHOUTcast has 2 second delay.
With this result I think I should choose RTMP or SHOUTcast.
But I am not sure RTMP and SHOUTcast are the best protocol. :(
What protocol should I choose?
Do I need to provide a various protocol to cover all platform?
This is a very broad question. Let's start with the distribution protocol.
Streaming Protocol
HLS has the advantage of allowing users to get the stream in the bitrate that is best for their connection. Clients can scale up/down seamlessly without stopping playback. This is particularly important for video, but for audio even mobile clients are capable of playing 128kbit streams in most areas. If you intend to have a variety of bitrates available and want to change quality mid-stream, then HLS is a good protocol for you.
The downside of HLS is compatibility. iOS supports it, but that's about it. Android has HLS support but it is still buggy. (Maybe in another year or two once all the Android 3.0 folks are gone, this won't be as much of an issue.) JWPlayer has some hacks to make HLS work in Flash for desktop users.
I wouldn't bother with RTMP unless you're only concerned with Flash users.
Pure progressive streaming with HTTP is the route I almost always choose to go. Everything can play it. (Even my Palm Pilot's default media player from 12 years ago.) It's simple to implement and well understood.
SHOUTcast is effectively HTTP, but a poorly implemented version that has compatibility issues, particularly on mobile devices. It has a non-standard status line in its response which breaks a lot of clients. Icecast is a good alternative, and is what I would recommend for production use today. As another option, I have created my own streaming service called AudioPump which is HTTP as well, and has been specifically built to fix compatibility with oddball mobile clients, such as native Android players on old hardware. It isn't generally available yet, but you can contact me at brad#audiopump.co if you want to try it.
Latency
You mentioned a latency of 2 seconds being desirable. If you're getting 2-second latency with SHOUTcast, something is wrong. You don't want latency that low, particularly if you're streaming to mobile clients. I usually start with a 20-second buffer at a minimum, which is flushed to the client as fast as it can receive it. This enables immediate starting of the stream playback (as it fills up the client-side buffer so it can begin decoding) while providing some protection against buffer underruns due to network conditions. It's not uncommon for mobile users to walk around the corner of a building and lose their nice signal quality. You want your stream to survive that as best as possible, so if you have already sent the data to cover the drop out, the user doesn't have to know or care that their connection became mediocre for a short period of time.
If you do require low latency, you're looking at the wrong technology entirely. For low latency, check out WebRTC.
You certainly can tweak your traditional internet radio setup to reduce latency, but rarely is that a good idea.
Codec
Codec choice is what will dictate your compatibility more than anything else. MP3 is easily the most compatible, and AAC isn't far behind. If you go with AAC, you get better quality audio for a given bitrate. Most folks use this to reduce their bandwidth bill.
There are licensing fees with MP3, and there may be with AAC depending on what you're using for a codec. Check with a lawyer. I am not one, and the licensing is extremely complicated.
Other codecs include Vorbis and Opus. If you can use Opus, do so as the licensing is wide open and you get good quality for the bandwidth. Client compatibility here though is the killer of Opus. (Maybe in a few years it will be better.) Vorbis is a mediocre codec, but is free and clear.
On the extreme end, I have some stations doing their streaming in FLAC. This is lossless audio quality, but you're paying for 8x the bandwidth as you would with a medium quality MP3 station. FLAC over HTTP streaming compatibility is not code at the moment, but it works alright in VLC.
It is very common to support multiple codecs for your streams. Depending on your budget, if you can't do that, you're best off with MP3.
Finally on encoding, don't go from a lossy codec to another lossy codec if you can help it. Try to get the output stream as close to the input as possible. If you re-encode audio, you lose quality every time.
Recording from Browser
You mentioned users streaming from a browser. I built something like this a couple years ago with the Web Audio API where the audio is captured and then encoded and sent off to Icecast/SHOUTcast servers. Check it out here: http://demo.audiopump.co:3000/ A brief explanation of how it works is here: https://stackoverflow.com/a/20850467/362536
Anyway, I hope this helps you get started.
Streaming straight audio/mpeg (mp3 packets) has worked everywhere I've tried.
If you are developing an APP then go with AAC, if you are simply playing via web browser then you need a HTML5 Implimentation which is MP3. All custom protocols like RTMP or SHOUTcast requires additional UI to be built. There are some third party players available in open source world. You can either use them or stick to HTML5 MP3/OGG as most people now days are using chrome browser or other HTML5 complaint browsers.
I'm trying to understand what the introduction of the Web Audio API has meant for the development of web based games.
Flash games can of course do some quite advanced audio processing, and for simpler games the audio element was maybe enough. But how has Web Audio API changed the game dev scene? In terms of what can be done, supported platforms and so on.
Supported platforms are Chrome, Safari (with some prefixing caveats) and Firefox across all supported hardware/OS platforms; IE is working on development, though the longer tail of versions will take a while to deploy.
Web Audio enables very complex processing, but also very precise timing and multiple sounds; sound management is far, far easier than previously possible in HTML5. In short, Web Audio dramatically improves the story for game audio development on the Web - which, of course, was one of its goals.
I have a large amount of audio stored on my web server in a very custom format that can't be replayed by anything other than my own application. That application is a Win32 app that can connect to my web server and stream and replay that audio.
I'd really like to be able to do the streaming and replaying from within a browser, but don't know where to start. Ideally I'd like the technology to be cross-platform (unlike my current Win32 app) and cross-browser (IE 6 and above and Firefox).
My current thoughts are to look at things like:
Flash, but doesn't that only replay mp3 audio?
Java, are VMs freely available still?
Converting the audio to a WAV file on the web server and then using someone else's plugin to replay that file. I'd rather keep the conversion off the web server for performance reasons, but is still an option.
Writing my own custom plugin to do the complete stream and replay operation.
Any guidance would be most useful.
Please note that the audio is not music and that simply converting to another audio format is not trivial. The audio that is stored also changes frequently (every minute) would need constant conversion.
Why are you using a proprietary music format? I'd probably not even bother downloading a program to listen to it.
I would suggest you convert it to mp3 and then use flash.
Building your own plugin would probably be hard, there are so many different platforms you'd have to cater for, something like flash is written for them already.
Apart from converting server-side: Implement a decoder for your format in ActionScript or Java. Then you can write a Flash movie or Java applet that plays it. Both languages/runtimes should be fast enough to decode in realtime unless your format is very complex. Flash would be the more accessible of the two, since nearly everyone has the plugin installed. (It's possible that playing a raw sound buffer isn't supported by older Flash versions than 10, I'm no expert on that.) The Java plugin is definitely free, but you'd require the users to install it.
I'd go with converting the audio to WAV (or MP3) on the server. Writing your own cross-platform browser component would be a lot of work, thanks to the different ways the major OSes handle their audio APIs.
Try taking a look at shoutcast.
Basically its a server app that will stream music to any client that connects to it through a browser (effectively your own radio station). I've never used it myself but should be straight forward.
Another idea is winamp remote. Again you install the app on the server but this time you can browse your music collection on their website and play individual songs.