I am trying to build an open-source in-ear monitoring system. I have created the UI and was wondering how I would get the channels that are on an audio mixing console so that I can edit the channels and stream them to each musician. Is there a certain protocol that all the mixers use? You can find the project at https://gitlab.com/openstagemix. We would love to have contributors.
I can't really test whether this is the correct answer as I am trapped in my house during the coronavirus time. But, all mixers use something called OSC which is a protocol between mixers, synthesizers, etc. to computers. You can find more information here http://opensoundcontrol.org/introduction-osc.
Update:
It's neither! I am going to use the AES67 standard to receive information from my mixer and with that process the audio. This is because my mixer is ethernet capable.
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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.
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
I was wondering whether it is possible to capture audio data from other sources like the system out, FM radio, bluetooth headset, etc. I'm particularly interested in capturing audio from the FM radio and already investigated all possibilities including trying to sniff the raw bluetooth communication between the phone and the radio device with no luck. It's too bad Android only allows recording audio from the MIC.
I've looked at the Android source code and couldn't find a backdoor to allow me to do that without rooting the device. Do you, at least, have any idea how to use other devices (maybe access somehow /dev/audio) say via NDK or even better - Java (maybe Reflection?) to trick the system to capture the audio stream from say, the FM radio. (in my case I'm trying to develop the app for the HTC Desire)
PS. And for those of you who are against using undocumented APIs, please don't post here - I'm writing an app that will be for my personal use or even if I ever publish it I will warn the user of possible incompatibilities.
I've spent quite some time deciphering the audio stack, and I think you may try to hijack libaudio. You'll have trouble speaking directly to the hardware (/dev/*) because many devices use proprietary audio drivers. There's no rule in this regard.
However, the audio hardware abstraction layer (HAL) provided by /system/lib/libaudio.so should expose the API described at http://source.android.com/porting/audio.html
The Android system, and especially audioflinger, uses this libaudio HAL to find available devices, deal with routing, and of course to read/write PCM data.
So, you could hijack the interaction between audioflinger and libaudio, by renaming the later, and providing your own libaudio which decorates the real one. Doing so, you should be able to log what happens and very possibly intercept FM radio output, provided that this is not directly handled by the hardware.
Of course, all this requires rooting. Please comment if you manage to do this, that interests me.
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.