Recent content by neodog

I came very close to forming this thought, but missed. Your phrasing it this way triggered me to rethink everything, and maybe identify the key to the problem. I think the problem I've been facing reduces to an "area under the curve" problem: the area under the curve (especially the smoothed...

I'm basically describing a smoothing function like: over all x: value[x] = max(get_bin_values_between((x-30hz),(x+30hz)) The noise seems to have a pattern of spikes roughly every 120hz, so smoothing in terms of the maxima across blocks around that size would let me see generally how tall these...

Okay, well, that at least opens the possibility of reusing code in the RMS window, since the relevant data is there. Although I've been thinking, it might be more useful to make the value displayed in the legend configurable: it could default to the value at the cursor as it shows currently...

I can see the apparent contradiction, that's probably a conversation in its own right. I'm trying to use the RTA to analyze and improve noise floor, but struggling to find a good way to do it. Specifically, I'm hearing changes in baseline noise that sound to me like +20b but only look like...

Ah, of course, I wasn't compute-bound there, it just takes more time to fill out a 1M sample window. So FFT Length and Overlap normally govern the frequency of update, determined by the duration of FFT_Length * (1-Overlap). And ideally, it seems that every sample gets processed in real-time on...

Are the RMS values in RTA computed off of the sample stream instead of the FFT? I notice that the different windowing settings can push the spectral plot around quite substantially, but the RMS numbers never budge in response. That would help explain why there aren't any methods of recomputing...

The manual has this to say about FFT length: And about Max Overlap: Is the overlap % the amount of the block overlapping, or is it % not overlapping? I ask because I get the slowest computations setting Max Overlap to 0%, and the fastest computations setting it to 93.75%; the values in...

Okay, I'm seeing a "data offset" of 120db, and Input FS Sine of 182.5 mVrms. So for this data: To get RMS SPL: -79.84 dBFS + 120db = 40.16 RMS dB SPL To get mVrms: 182.5 * 10 ^ ( -79.84 / 20 ) = 0.01858 mVrms (18.58 uVrms) Does this look right? The 40 SPL seems about 3dB high, but...

The RTA window presents an RMS computation in a box in the upper hand corner of the graph window. When saving the aggregates as a measurement, the RMS gets added in the form of notes on that measurement: Input RMS -30.60 dBFS -31.5 dBFS C, -47.8 dBFS A -30.6 dBFS 22 - 22k UNW -56.2 dBFS >22k...

I see. So even if you use a loopback timing reference, you pretty much always have to use "Estimate IR delay" on the result, unless you are 100% sure nothing moved since the last time the offset was updated? But if "Estimate IR delay" is required, do I actually get a benefit from using a...

Perfect. I see... this actually tells me everything I needed to know. It means you can use an output as a ref input, which confirms that we can safely assume that the Ref and Output signals are always 100% identical. I had thought the ref signal might be somehow unique, as with the...

Does using a Loopback timing reference automatically compensate for the timing differences observed bout output and ref in a measurement? If so, what is the effect of having both the loopback and the offset? I got the above by using a loopback reference, and then using "Estimate IR" to...

Related question: The Preferences window requires both Output/Input and Ref IO channels to be specified. Suppose you have a soundcard with just a mono input (as with a laptop's headset jack), but which presents in Windows as a stereo input L and R, being copies of the same input. For...

I've observed a few idiosyncrasies in REW Output and ref signal configurations, I just want to bring them up to clarify or change my understanding. In preferences, it is possible to configure an output to use L+R, and a loopback ref to use either L or R. In a stereo system, isn't L+R...

I've been struggling to understand statements about distortion being at times t<0, since it's counterintuitive... i.e. surely distortion occurs after the sound that inspired it. But I think this must be a characteristic of the swept impulse analysis approach, wherein the response is analyzed...