I have written an application that receives media files from a central server and plays those files according to a playlist. All works well.
A client has contacted us and wants to use our application to play some audio files as presentations in a kiosk-style application. So far, so good, our application can handle this no problems.
He has requested as a potential feature that we would have a number of headphone sockets at the front of the kiosk. Each headphone socket would play the same audio presentation in a different language.
I have come up with the idea of encoding a single audio file with the presentation in multiple languages, and each language in a different channel. We would then require a sound card that could decode each channel and output it on a different headphone socket.
Thing is, while I'm think the theory is sound, I have absolutely no idea whether this is feasible and what would be required to pull it off.
Any ideas?!
As a side-note: the application uses Media Player as the underlying component to handle the playback of audio and video. I'd appreciate any help as to the software we could use to generate the multi-channel audio stream and the hardware (USB sound card would be fine) that we could use to decode the stream.
Thanks!
You need to use multiple files not channels, its going to be way easier that way.
Instead of using Media Player use DirectShow (on .NET you have DirectShow.NET), In DirectShow you have the notation of Multiple files on the same graph.
You will be able to control to which audio device play which files, and your Play, Pause, Stop commands will be preformed on all files without you need to worry about syncing.
There are many samples on how to build media player like with DiectShow, extending them to use multiple files should be really easy.
For HW take a look at this (USB with 8 output channels)
I think with Shay's hardware you've got a complete solution:
Encode a 7.1 file with a different mono voice track on each channel.
Use the 8 channel output device in 7.1 mode, with a different headset in each port, and you've got it. Or, if you only have 6 languages, a 5.1 file would work. Many PC's have 5.1 outputs built in, you'd only need 3 splitters to break out the left and right channels from each jack.
You can do the encoding with Windows Media Encoder, or other pro audio tool.
Related
I have a Linode server and need to broadcast one to-many audio (they can hear but can not talk back) to a group of three to five people. I looked at WebRTC and the Janus server but it seems complete overkill. Using commercial applications like Skype, Discord etc. results in low audio quality and it is mono. Best possible audio quality and low latency (on a par with that of Skype, Discord etc.) is essential.
Any pointers would be greatly appreciated.
I can recommend building such system based on Icecast streaming. It's an old proven technology which has a latency close to real-time.
You could use any set of Icecast-enabled tools for that.
As example, here's what you an do with tools by our company:
Larix Broadcaster mobile app allows streaming in audio-only
mode.
Nimble Streamer software media server can get Larix' input and
produce Icecast stream. You can use any Icecast-enabled here
instead.
SLDP Player can play Icecast produced by Nimble
Streamer or any other Icecast-enabled server.
That can also be built with other companies products, so you can pick the right tools yourself.
A super simple setup would be to just use command line tool called ffmpeg (it also has an api) see doc at https://trac.ffmpeg.org/wiki/ffserver
Where your source audio lives just launch either the ffmpeg or ffserver
ffserver -f /etc/ffserver.conf
in that config put location of source audio and output url it will publish to ... then your client receivers can use ffplay with
ffplay <stream URL>
ffmpeg is a free open source industry workhorse for audio/video manipulation ... its the underlying technology several more visable tools like vlc use under the covers
I'd like to be able to capture the audio from the audio card of my computer and to dispatch it with WebRTC. However, I am not sure if it's possible or not to have access to the audio directly produced by my computer.
According to this repo https://github.com/niklasenbom/RecordingApp/blob/master/app.js there is a system audio stuff but not sure if it's what I'm looking for.
Thanks,
You can do it by using NAudio. Actually I did the same project myself and will put it in GitHub in a few weeks and update this answer. You can configure the frequency etc. and use it's OnDataAvailable event to dispatch the sound to registered clients.
I am to test voice recognition programs. Some which I have access to the code and others where I don't.
Sadly my (beautiful) voice is not perfect, so when I am reading a text it sounds slightly different each time. Which makes the testing difficult and time consuming. Giving that I can tweak a lot of parameters.
So I was wondering if there was a way to record my own voice (already done). And then play it as normal microphone input so the voice recognition program I am testing will see it as microphone input.
This would also help greatly if it could be done programatically in C#. So I can in my own code specify when to play what.
To play it from speakers and have the voice recognition programs listen to the microphone is not an option, because it is not the same sound on different computers/speakers/microphones.
Thanks.
Edit:
What i have found so far is to use a software sound Card simulator. But I haven't been able to find a suitable one.
Just as there are printer drivers that do not connect to a printer at all but rather write to a PDF file, analogously there are virtual audio drivers available that do not connect to a physical microphone at all but can pipe input from other sources such as files or other programs.
I hope I'm not breaking any rules by recommending free/donation software, but VB-Audio Virtual Cable should let you create a pair of virtual input and output audio devices. Then you could play an MP3 into the virtual output device and then set the virtual input device as your "microphone". In theory I think that should work.
If all else fails, you could always roll your own virtual audio driver. Microsoft provides some sample code but unfortunately it is not applicable to the older Windows XP audio model. There is probably sample code available for XP too.
In J2ME app, I want to give freedom to user to set his/her own liking tone though it is .mp3 or .wav sound with high bit rate & size. It is possible in J2ME with javax.microedition.media package??
I tried to do this with one .wav file whose size is 2.52MB and bit rate is 352kpbs but Netbeans showed me exception which is
java.lang.OutOfMemoryError
at javax.microedition.lcdui.Display$DisplayAccessor.commandAction(Display.java:1996)
at javax.microedition.lcdui.Display$DisplayManagerImpl.commandAction(Display.java:2825)
at com.sun.midp.lcdui.DefaultEventHandler.commandEvent(DefaultEventHandler.java:303)
at com.sun.midp.lcdui.AutomatedEventHandler.commandEvent(AutomatedEventHandler.java:670)
at com.sun.midp.lcdui.DefaultEventHandler$QueuedEventHandler.handleVmEvent(+186)
at com.sun.midp.lcdui.DefaultEventHandler$QueuedEventHandler.run(+57)
So is there any way to do this or I have to restrict user to use tones which is provided with app?
You cannot change the ringtone or alarm tone of the device via JavaME.
You can make an app that plays its own tone. Which formats can be played depends on the device. And how big they can be, depends on the amount of memory available on the device.
If you wish for your app to be compatible with the most possible devices, you should use MIDI or AMR files. All newer devices do also support mp3 and wav files though, as far as I know.
I have developed a pretty complex audio software for my client with plugins for Winamp, Windows Media player and VST. Now the client is interested in some method to avoid maintaining the multitude of plugins, we have no way to support all the media players out there.
The client does not care for Unix/Mac yet, so I can look only at Windows XP and Vista/7/
Basically, what we need is a way to always reliably intercept as much audio stream protocols as possible (well, except maybe ASIO, that's another story, I guess), then pass this audio through our custom effects engine and then route back to the default audio device, whatever it is.
Now I am thinking, what options do I have (theoretically).
I could use hooks. I need to hook globally older vaweOut and also DirectSound.
But will this still work on Vista/7?
I could use a virtual driver, like the author of the Virtual Audio Cable did:
http://software.muzychenko.net/eng/vac.htm
Seems a pretty daunting task. Anyway, the client will contact the author of VAC to see if he agrees to sell his source code for a reasonable price.
This driver could install itself as a default audio output device, intercept the audio stream from Windows, and pass it back to default device. Hmm, but what about various DirectSound audio buffers, do I have to mix them myself or is there any way I could tell Windows mixer to mix all for me and pass a single mixed audio stream?
It seems, this custom driver will of course kill all the hardware audio acceleration, but we can live with that, if we warn our customers about this issue.
As I understand, the most current Windows driver standard is WDF.
But maybe it does not work for audio on Windows Vista/7?
I know, Vista/7 has a different audio stack from XP.
If I can do it using WDF, what driver should I write - kernel mode or user mode?
Maybe I am missing more elegant and simple options to intercept, process and route audio on Windows?
Try Virtual Audio Streaming SDK. Also virutal sound card and let you read/process audio data in realtime.
http://www.virtualaudiostreaming.net/sdk-license.html