I need to do some NON-STANDARD signal processing operations with an RFID-reader, so I'd like to know if it is possible to extract antenna's individual analog (actually digital samples right after ADC) input signal samples with Motorola FX7500 (if you know how this works on FX7400 or FX9500, please do tell, could be helpful). Samples would be processed in a JAVA-based host computer program.
What I've already tried:
Investigating Motorola's own RFID3 API's possibilities, it doesn't go deep enough to actually get in touch with input analog signal samples.
Using LLRP to its full extent, it doesn't allow analog signal sample access either. RFsurvey-functionality would have been helpful to some extent, but FX7500 doesn't support it either.
Accessing RFID-reader's linux terminal, trying to find the driver function(s), that could listen the input sample stream. If current input sample(s) could be extracted from the input stream, I could (in theory) make a script, that would save a few of those sample values in a txt-file in the host computer during a tag inventory round. My linux skills are kinda bad, hence I ask this question.
The only realistic way to solution seems to be via linux terminal, so if you folks have any ideas about that (where to look and what to do), please advise!
Contents of reader:
rfidadm#FX7500abcdef:/$ ls -1
apps
bin
dev
etc
home
include
lib
linuxrc
media
mnt
platform
proc
readerconfig
run
sbin
sys
tmp
usr
var
I cannot completely rule that out, but it's highly unlikely you can get the raw signal digitized; the devices you're looking at aren't really software defined radio devices, typically.
"speaking" RFID physically is a bit different from "usual" wireless communication: The reader doesn't only observe the energy transmitted from the tag, but more importantly the fluctuations of energy extracted from the near field of the reader's antenna coil. Hence, you don't actually have a baseband of RF bandpass signal, but hardware-specific modulations of transmitted (and inversely, antenna-reflected) energy. Demodulation is hence usually done in specialized hardware.
However, do not fret: It's totally possible to build a software defined RFID reader. There have been several approaches to that, but personally, I trust these based on Ettus USRPs and/or GNU Radio best. Look through the results IEEExplore gives you, eg. this search.
Most probably this is not possible with the Motorola readers. What you can do, is use one of the RFID chipsets available on the market: either the AMS RFID IC's, or the Impinj RFID IC's. As far as I know, both IC's support retrieving the digital samples that are received. They also have a development kit to test-drive the IC's.
Related
I'm trying to make a video tutorial, so i decided to record the speeches using a TTS online service.
I use Audacity to capture the sound, and the sound was clear !
After dinning, i wanted to finish the last speeches, but the sound wasn't the same anymore, there is a background noise(parasite) which is disturbing, i removed it with Audacity, but despite this, the voice isn't the same ...
You can see here the difference between the soundtrack of the same speech before and after the occurrence of the problem.
The codec used by the stereo mix peripheral is "IDT High Definition Codec".
Thank you.
Perhaps some cable or plug got loose? Do check for this!
If you are using really cheap gear (built-in soundcard and the likes) it might very well also be a problem of electrical interference, anything from ...
Switching on some device emitting a electro magnetic field (e.g. another monitor close by)
Repositioning electrical devices on your desk
Changes in CPU load on your computer (yes i'm serious!)
... could very well cause some kinds of noises with low-fi sound hardware.
Generally, if you need help on audio sounding wrong make sure that you provide a way to LISTEN to the files, not just a visual representation.
Also in your posted waveform graphics i can see that the latter signal is more compressed, which may point to some kind of automated levelling going on somewhere in the audio chain.
I've been looking online for solutions on how to generate a simple beep with an DE2 Altera board using VHDL but I can not seem to find anything.
I've seen some things that are talking about Audio Codec but I haven't been able to get a firm grasp of what I would need to do with it.
Thanks a lot!
-Will
The "audio codec" is a separate chip (a Wolfson WM8731 in your case) which can turn information on digital form into analog information to be amplified/played through a standard sound system with amplifier/speaker/etc.
You'll need to find out how to interface to the audio codec chip from the FPGA (google a datasheet for the chip), and then send it the necessary information to generate you beep. Depending on how advanced the audio chip is, this task can vary rather much in complexity. As it's an educational board, you might be able to find people who've already done something similar, and then build on their experience.
If all you need is a simple beep though, you might also be able to create a simple circuit with a piezo-buzzer or similar yourself - that can be controlled without having to use an audio codec...
On a basic embedded systems speaker with a single line of output, wiggling the output as 0 or 1 in a for given periods produces sound.
I'd like to do something similar on a modern Linux desktop. A brief look-see of Portaudio, OpenAL, and ALSA suggests to me that most people do things at a considerable higher level. That's ok, but not what I'm looking for.
(I've never worked with sounds on Linux before, so if a tutorial exists, I'd love to see it).
Actually, it... kinda is. While you can generate the waveform yourself, you still need to use an API to queue it and send it to the audio hardware; there no longer even exists a sane way to twiddle the audio line directly. Plus you get cross-platform compatibility for free.
[...] embedded systems speaker with a single line of output, wiggling the output as 0 or 1 in a for given periods produces sound.
Sounds a lot like the old PC speaker. You might still find code for it in the Linux kernel.
I'd like to do something similar on a modern Linux desktop.
Then you need AFAIK a driver for ALSA. There you can find infos on how to write an ALSA driver. Use PWM to produce the sound.
Since there are many different sound cards and audio interfaces produced by different companies, there is no uniform way to have a low level access to them. With most sound I/O APIs what you need to do is to generate the PCM data and send that to the driver. That's pretty much the lowest level you can go.
But PCM data is very similar to the 0-1 approach you describe. It's just that you have the in-between options too. 0-1 is 1-bit audio. 8-, 16-, 24-bit audio is what you'll find on a modern sound card. There are also 32- and 64-bit float formats. But they're still similar.
I am currently thinking about what I could do to measure the time it takes from the point where the computer gets audio input (through a normal audio input on a soundcard) to the point where there's something to work with, e.g. noise cancellation or something like that.
The main problem I reckon is to measure when the audio signal was created and the synchronization of the sender and receiver.
So far I came up with the following ideas:
Use the serial port to transmit timing information
Put a timestamp into the audio signal
Transmit a recurring signal - a delay would be visible
Do you have more ideas or something that I m not seeing in mine? I thought I would find more academic work on this matter but was sad to see that this is not the case, am I searching wrong?
you can check the latency in windows with this tool they also have some great info on the site also you can read up on the ASIO drivers or try to reach out to the communities that use these tools (DJs Guitar modeling scene) another great source of information is open Source projects like JACK that have more technical information:
Latency Tool:
http://www.thesycon.de/deu/latency_check.shtml
Asio Wikipedia Page:
http://en.wikipedia.org/wiki/Audio_Stream_Input/Output
Guitar Amp Modeling:
http://www.guitarampmodeling.com/
JACK Project homepage:
http://jackaudio.org/
Hope that helps.
Im doing a dsp project and i want to take an anolog file and convert it to a digital output using a microconroller attached to a ADC on a dsp board. How would i program this in c?
Pretty much its as simple as that, atleast i think.
This is what i need.
Input --------- Output
Angolog --> Digital
Digital --> Anolog
You really need to clarify your question. Like what do you mean by analog file? File systems are binary from a programming perspective, sure the media is magnetic or other technologies and there is analog involved. An ADC goes from Analog to Digital, so it is an an analog input not a digital output.
ADC analog to digital converter, takes analog inputs to the device and converts them to digital so you can use them inside the chip, program, save to files, etc.
DAC digital to analog converter, takes digital values and converts them to analog outputs.
In both cases you need to look at the specific details for the chips and the board. From a programming perspective if nothing else you need to look into the details for the ADC and or DAC. Microcontrollers having an ADC is not uncommon, but you need to read up on how to get the ADC on that microcontroller to initiate a sample, how to know when the sample has completed and how to read the digital data once the sample has been taken. DACs are often external, sometimes serial, so you may have to bit bang spi or i2c or look into what hardware the microcontroller might have for speaking spi or i2c or if there is a dac in the microcontroller, how to use it (what registers to write, etc).
If you have a specific publicly available microcontroller board, for example an eval board, then that makes it much easier for folks here or elsewhere to show you where to look for the schematics, data sheets, etc. Otherwise, even knowing exactly which microcontroller and what I/O pins are used, would be helpful when asking such a question. There are probably lots of example programs out there that could be borrowed from. And it could be as simple as a few lines of C to an existing library, or as complicated as many lines of C with interrupt service routines, and possibly some assembler.
This is extremely dependent on your hardware and there's no information in the question that would enable a real answer.
In general, you should see the documentation for your system, especially the AD/DA parts. There should be good examples. If there's a particular problem, post a more specific question.