Where can I find a complete specification of the q2q protocol? I wanna try implementing it on my own also to see difference between q2q and p2p but I can't find a complete documentation.
It looks (to me) like Q2Q is just a term for the data transmission scheme[1] implemented by Vertex. And as vertex is work in progress, it is a moving target.
A good bet is to have a look at the current implementation:
http://buildbot.divmod.org/apidocs/vertex.q2q.html
[1] I am reluctant to call it a protocol until we actually unearth a specification for it. ;)
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Good day!
Problem definition:
Current implementations of Bluetooth does not allow to simply support good quality of Audio(Earphones mode) and 2-way audio transition (Headset mode).
Also, even if one would manage to set this configuration up, which have huge limitations on the hardware/software used, there is no way to handle sound input from 2 different audio devices simultaneously.
So, technically - one cannot just play the Game, communicate on the Discord, and optionally listen to some music, unless he is bound to some USB-bundled earphones. Which are usually really crappy, or really expensive. Or both.
Solution sketch:
So, I came up with an idea that one can actually build such device, using Raspberry Pi, Arduino, or even barebone-component-based stacks.
Theoretical layout of connections per-se would look somehow like that:
Idea is to create 2 "simple" devices
One, not-so-portable, that would handle several analog inputs, and one analog output
One, portable, that would handle single analog Input and Output, and could be used with any analog earphones.
"Requirements" to such system would be quite simple:
This bundle have to handle Data Transition on some distance, preferably up to 10 meters, or more.
The "Inlet" device should be portable enough to keep it in the pocket, or in an arm band, or something
Sound Quality should be at the very least on the level of Bluetooth headphones profile, or if possible - even better
If possible - it would be nice to keep the price of the Solution under 500 Euros, but I'm so tired of current state of things that I might consider raising the budget...
Don't mind the yellow buttons on the Outlet device. Those are optional, and will depend on the implementation stack :)
Question:
Can anyone advice me which component-base would be a better solution to making such a tool, and why?
And maybe someone actually knows of similar systems already existing?
Personally I would prefer anything but the barebone-components-based solution, just because I'm really rusty with that area, and it requires quite the amount of tools, to handle it properly.
While using pre-built modules can save me from buying most of the hardware tools, minifying my "hardware customization" part of this solution, leaving only software part to handle (which is my main area of expertise).
But then again, if there are some experts here, that would consider other stacks non-viable - I would really appreciate to see their reasonings.
P.S. Just to be clear: If this project will prove viable - I will implement it, and share the implementation details with the communities. I am not the first one who needs such system, and unfortunately it seems that Hardware/Software vendors are not really interested in designing similar solutions...
I happen to find a "temporary" solution.
I've came across a wireless headset, that allows to simultaneously support Wireless USB Bundle connection, and Bluetooth connection to different devices, and provide nice way of controlling sound input/output with both connections.
This was almost a pure luck, as this "feature" was not described anywhere in the specs...
Actual headset name is:
JBL Quantum 800
This does not closes the question per-se, as I still plan to implement this "Summer Project" at some point, but I believe this information might be useful to those searching for similar solutions.
I have searched far and wide for the answer to this and am surprised that there aren't more people talking about this.
There doesn't appear to be a way to ensure that the stream/track that is selected is a native camera versus a stream provided by something like ManyCam (https://manycam.com/?__c=1) or AlterCam. The use case for this is secure apps that, for security reasons, want to ensure that the video/images/data coming through the stream comes from a "legitimate" (native camera) source rather than a source that can really be anything and/or altered before it gets to the client code.
Is there any known way of ensuring this? I'm not looking for a whitelist/blacklist option, since many of the camera programs allow changing of the camera name, which ends up showing up in the label property.
Here is what i like to achieve:
I like to play around in creating "new" software / hardware instruments.
Sound processing and creation is always managed by software. But one could play the instrument via ultrasonic distance sensor for example. Another idea is to start playback when someone interrupts the light of a photoelectric barrier and so on....
So the instrument would play common sounds, but has to be used in an unusal way. For example, the ultrasonic instrument would play a sound if it detects something in a certain distance. The sound could be manipiulated in pitch for example if the distance gets smaller.
Basically i like to playback a sound sample and manipualte this in realtime.
I guess i have to use WAV samples for this, right? And which programming language do you think fits best for this task?
Edited after kevins hint: please kick me into the right direction - give me a hint where to start.
Thanks in advance
Since you're using the the Processing tag, you can try Processing.
It comes with a sound library like Minim or you can install beads which is great. There's actually a nice book on it: Sonifying Processing
You might find SuperColider fun as well.
The main thing is what are you comfortable with at the moment ?
If Processing syntax looks intimidating, you can actually try a different programming paradigm like data flow. In which case you can use PureData(free, opensource) or MaxMSP(very similar, but commercial). The idea is rather than typing instructions, you connect boxes with wires which is fun and the examples are great too.
If you're into c++ there are plenty of libraries. On the creative side, there's a nice set of libraries called OpenFrameworks that's easy and fun to use. If this is your cup of tea, have a peek at Maximilian.
Bottomline is: there are multiple options to achieve the same task. Choose the best tool for your (based on your background) or try each and see what you like best.
You asked "And which programming language do you think fits best for this task?" - I would also suggest using Processing. I have been used Processing to work with sounds previously. And in all cases I used Minim. It has many UgenS to generate sounds programmatically.
Also, you wants to integrate with some sensors. I'm not sure what types of sensors you will use, but Processing goes pretty well with different Arduino modules and sensors. Check this link for more direction.
Furthermore, you can export your project as .exe or executable .jar files. And their JS version (P5.js) works almost the same as the Java version.
I'm looking for a program that is able to recognize individual audio samples from my computer and reroute them to trigger WAV files from a library. In my project, it would need to be realtime as the latency would not be a desired result. I tried using dictation software that would recognize words to trigger opening a file and that's the direction where I want to go, but instead of words I want it to be sounds and it would happen in realtime. I'm not sure where to go and am just looking for some guidance. Does anyone have any suggestions of what I should do?
That's a fairly broad question, but I can tell you how I would do it. (Hardly the only way, but where I would start.)
If you're looking for real time input, the Java Sound library (excellent tutorial here) allows for that. (Just note that microphone input from a web page is difficult on anything, due to major security concerns, so this would be a desktop application.)
If it needs to be real time, the first thing I would suggest is stream and multithread the hell out of it. I would suggest the Java 8 Stream API, but since you're looking for subsamples that match a specific pattern, then each data point will have to be aware of the state of its neighbors, and that isn't easy with streams.
You will probably want to know if a sound roughly resembles an audio profile, so for that, I would pick a tolerance on just how close you want it to be for a match (remembering that samples may not line up 100% anyway, so "exact" is not an option), and then look up Hidden Markov Models. I suggest these because they're what voice recognition software typically uses, and while your sounds may not be voices, it will give you an idea of what has already been done.
You'll also want to maintain a limited list of audio samples in memory. Specifically, you will likely need the most recent data, because an audio signal is a time-variant signal, and you can't get a match from just one point. I wouldn't make it much longer than the longest sample you're looking to recognize, as audio takes up a boatload of memory.
Lastly (for audio), I would recommend picking a standard format for comparison. Make it as good as gets you decent results, and start high. You will want to convert everything to that format before you compare it.
Once you recognize a specific sound, it's basically a Command Pattern. Specific sounds can be mapped, even with a java.util.HashMap, to specific files, which (if there are few enough) you might even have pre-loaded.
Lastly, it's worth looking at the Java Speech API. It's not part of the JDK and it's quite dated, but you might get some good advice from its implementation.
This is of course the advice of a Java-preferring programmer, but I imagine that there might be some decent libraries in Python and Ruby to help you as well; and of course there's something in C somewhere. This may sound like a lot, but most of the material is already implemented and ready-to-go.
Hopefully this helps, let's look forward to other answers.
I am trying to change the pitch of a buffer sample using a scriptprocessor, but what kind of formulas do I need to do this? I am not looking for the exact js code, but just for some general mathematical how to. I would love to have some code for this, as the first answer has a lot of formulas where I have no idea on how to implement that in JS.
I know that this is working with time, but according to this it can be done with the FFT, but I have no idea how one should do that.
For one method of doing time-pitch modification using an FFT, look up phase vocoder. Here's one explanation of how a phase vocoder works (but a search will turn up many others): http://www.guitarpitchshifter.com/algorithm.html
I believe https://github.com/mikolalysenko/pitch-shift would be appropriate (the quality is not on par with other code, but this library is rather easy to understand/use). You can hear a demo at http://mikolalysenko.github.io/pitch-shift/.