gemini86
Well-known member
^^^ I was under that impression also.
Rochey said:Is it also a real time streamer? I was under the impression that it buffered most of the samples, then burst it accross the USB in data, not audio, format.
ruffrecords said:Here is the result of lsusb -v
Rochey said:just for clarification, I belive the USB 1.x spec no longer exists. The USB 2.0 Spec covers low, full and high speeds.
from a simple silicon perspective, most microcontroller vendors are doing USB Full Speed onchip, with high speed requiring a separate PHY IC.
This sounds like what the Audio Precision done - tests are all done locally, then results are burst back to the PC.
There are advantages in doing this, namely making it compatible with the most pedestrian USB connection, and also providing some unique features.Andy Peters said:That's how you'd do a logic analyzer or an oscilloscope, but it seems like if real-time support for streaming data is available, why have gobs of local (on-device) storage and burst it to the computer for analysis?This sounds like what the Audio Precision done - tests are all done locally, then results are burst back to the PC.
Andy Peters said:So the device should support the advertised 192 kHz/24-bit data streams with two channels in each direction simultaneously. This all depends though on their drivers and what they intend to do with the product. It's somewhat surprising that they don't support the standard Audio Class 2.0, but then again, Windows drivers don't exist and there's a lot of required stuff in that class that might just get in their way.
Andy Peters said:Still, if someone could take a screwdriver to the case and crack it open ...
dfuruta said:Andy Peters said:Still, if someone could take a screwdriver to the case and crack it open ...
Here you go, but it's not very helpful.
abbey road d enfer said:There are advantages in doing this, namely making it compatible with the most pedestrian USB connection, and also providing some unique features.Andy Peters said:That's how you'd do a logic analyzer or an oscilloscope, but it seems like if real-time support for streaming data is available, why have gobs of local (on-device) storage and burst it to the computer for analysis?This sounds like what the Audio Precision done - tests are all done locally, then results are burst back to the PC.
In particular, one thing that frustrates me with the soundcard approach is the lack of a "processed" output. Ever since I started using an audio analyser (goes back to early 70's) I always had channel 2 of the 'scope connected to the post-processing output; that's how you see what type of noise or distortion you're facing. I agree that having an FFT spectrum is the next best thing, but not as instantly significant (FFT doesn't show phase).
I guess it could be possible with the soundcard approach to loop the processed output back to ch.2 D/A...
audiomixer said:I think this makes really sense from a developer perspective: to ensure proper operation on a windows box you must bypass the internal audio engine! re sampling, volume control, mix and whatever are a nightmare to handle. so one solution is to provide a dedicated 'tube' to the test interface and allow a ASIO connection to the application through a dedicated driver. no necessity to buffer audio, just sending it through a other path directly to the application.
I use a Duran Audio Interface that does exactly that and it is the most stable interface you can get a on a windows box.
I'm very familiar with the notion of assessing transfer function by comparing the DUT output to its input, using a random signal as stimulus, including its advantages and drawbacks. Ever since the days dbx announced that "your audience will love your test signal" (dbx RTA-1).Andy Peters said:Smaart does its transfer-function measurements using two inputs on the audio device. Your stimulus output (which can be the soundcard, or something else, like a function/noise generator) is split and feeds both the DUT and the second soundcard input. The first soundcard input comes from the DUT output. Of course transfer function gives both phase and amplitude.
(In some cases, like doing acoustics measurements or when your DUT has digital latency, you have to determine the delay through the DUT and add a complementary delay on the direct/reference input.)
Finally, Smaart has a trick setting which enables the use of the program's internal signal generator as the reference channel, so there's no need for a splitter and loopback.
-a
abbey road d enfer said:When you use an AP (and just about any audio analyser), there is a "processed" output, which is basically the signal hitting the "meter". In particular in THD mode, this output, taken at the output of the notch filter, contains all the residuals - harmonics and noise.
Observing this signal on a 'scope is of enormous interest for anyone who knows their Lissajous.
In order to do that with a soundcard, the software should route the output of the virtual notch filter to the ch.2 DAC input.
That may be doable if the THD processing was done by emulating the analog THD measurement, but in most cases, soundcard THD measurement is done by FFT. Reconstructing the residual signal by recombining the individual harmonics is feasible, but phase information is essential for making it useful.
In the particular case of QA, it seems the phase information is lost.
Of course, audio equipment is made to pass audio, so audio monitoring is de rigueur.JohnRoberts said:abbey road d enfer said:When you use an AP (and just about any audio analyser), there is a "processed" output, which is basically the signal hitting the "meter". In particular in THD mode, this output, taken at the output of the notch filter, contains all the residuals - harmonics and noise.
Observing this signal on a 'scope is of enormous interest for anyone who knows their Lissajous.
In order to do that with a soundcard, the software should route the output of the virtual notch filter to the ch.2 DAC input.
That may be doable if the THD processing was done by emulating the analog THD measurement, but in most cases, soundcard THD measurement is done by FFT. Reconstructing the residual signal by recombining the individual harmonics is feasible, but phase information is essential for making it useful.
In the particular case of QA, it seems the phase information is lost.
I am familiar with looking at, and even listening to, the product output from distortion analyzers. Back in the day I used to listen to the distortion residual to help evaluate audio path quality. While eyes and meters are good for quantitative judgements, ears are better for qualitative assessment in the margin.
That is not so in my experience. When the notch filter is properly tuned, the harmonics are not shifted. That would be the case if the notch filter was too wide. Since the notch filter is a single biquad, phase shift cannot exceed 90°. When the filter is narrow enough so that the response is close to flat at the second harmonic, the phase response is close to 0.I am not very confident in the phase information in that distortion analyzer product output since most distortion analyzers tweak the phase to realize a deeper notch filter.
The look of distortion is extremely useful. Crossover distortion appears as a discontinuity right in the middle of the wave form, third appears as a loop at the beginning and end, second as a dissymetry around the vertical axis...I guess you are talking about the phase relationship between the different higher harmonic distortion components. This phase relationship should affect the look of that distortion more than the sound of it.
Obviously, while listening to the product output alone can be more revealing without the masking of the fundamental present. While one can fall down the rabbit hole doing this, since masking is the real world.abbey road d enfer said:Of course, audio equipment is made to pass audio, so audio monitoring is de rigueur.
The look of distortion is extremely useful. Crossover distortion appears as a discontinuity right in the middle of the wave form, third appears as a loop at the beginning and end, second as a dissymetry around the vertical axis...
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