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.
Related
I am quite new to LabVIEW and NI devices.
I am working on Active Noise Cancellation Project, where I will be using two microphones input and one loud speaker as output. I have NI myRIO 1900 and CDAQ 9178 devices in our university lab. I need to do real time audio processing, I will collect data from microphone and process it using filtered XLMS algorithm to produce anti noise from loud speaker and other microphone is error microphone. I want to process data so quickly( within 1.7 msec ) so I will have real time response at 44100 sample rate !! My question is , 'is it possible to do with labview ?? and is stream processing possible in labVIEW?? and can I achieve so small audio latencies as mentioned above ??'
I have searched for audio processing objects in labview help. I can only find 'Acquire Sound', 'Play Waveform', surprisingly 'Acquire Sound configuration ' will work only for duration of minimum of 1 second not less than that !!! I can't input the time milli seconds !!!( I am still facing problem installing myRIO, so I have used host computed VI to do this.)
Please help !! Thank You
The thing you should be looking into is the FPGA part of the myRIO. You’re never going to be able to get 1.7ms response time via the host computer. The FPGA can access the Analogue inputs and outputs, so if you can get your algorithm to compile onto the FPGA then it should work.
Yes, it is possible with LabVIEW, insofar as any algorithm you want to code up can be executed by LabVIEW. If you're asking whether there is a library that already exists to do the filtering you're wanting to do, you may want to explore the NI Sound & Vibration toolkit, which is sold separate from LabVIEW, or explore third-party libraries.
The raw waveform mathematics abilities that come with LabVIEW are fairly extensive. You should be able to code whatever transforms you want if you know the base math.
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.
I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a 'line in' jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.
The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.
So I can do this today:
Take the good sound and the camera captured sound and load each into
audacity and pretty quickly use the timeshift toot to perfectly fit
the good audio to the questionable audio from the HDMI capture and
cut the good audio to the exact size of the video. Then I can use
ffmpeg or other video editing software to replace the questionable
audio with the better audio.
But while somewhat quick and easy, it always carries with it a bit of human error and time. I'd like to automate this if possible as this process is repeated at least weekly throughout the year.
Does anyone have a suggestion if any of these ideas have merit or could suggest another approach?
I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn't see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats?
if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now-- I've never tried either of these filters yet-- just trying to optimize against blind alleys by asking for advice up front.
If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I'd take it, if it turns out reasonably reliable
Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible-- but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.
any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI?
Thanks in advance for any pointers or suggestinons!
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 want to make a program that takes recorded speech and transforms it so it sounds like it's coming from a Texas TI-99. Do you have any good ideas and resources for how to go about that?
Most of those old speech synthesizers were build directly in-chip. Perhaps you could find a synthesizer that sounds like the chip, but if you really want the original sound, you would either have to simulate the chip (I don't know if it's a simple matter, perhaps the chip internals aren't published).
I only know because I burnt out a number of the Radio Shack speech synthesizer ICs before I managed to get a SP0256-AL2 working.
If you're more of a do-it yourself type guy, you need to find out which IC actually drove the speech synthesis in a TI-99, and then build the chip up on a bread board. That's what I was trying to do back then, and I managed to get the chip to speak, but lost patience after I fried my third chip due to a mis-wiring issue when I attempted to attach it to my PC's parallel port. I think this was the book I was using back then, but there's no cover art featured so it's hard to know for sure.
If you are familiar with how to use ROM images, there seems to be a gentleman that has managed to refeverse engineer the ROM image out of a SP0256-AL2. Look here for the image and the incredible granted permission to do the work and distribute the results.
You could start with open source that does something similar: Adding Robotic/Vocoder effect to your song using Audacity