dBm or dBm/RBW - Page 1
Forum section: electronics / rf-microwave

author: mattko
author_rank: Contributor
timestamp: April 11, 2022, 10:20:37 am
content: Hello,why are the y-axis units of a spectrum analyzer often described as dBm? The correct units for the y-axis would be dBm/RBW, which is an important difference. As soon as you look at wideband signals (wider than RBW), then it becomes quickly obvious that the marker on the signal does not represent the power of that wideband signal.

author: dmills
author_rank: Super Contributor
timestamp: April 11, 2022, 11:34:58 am
content: Because for the case where the signal fits inside the RBW (Which I would submit is the common case for a lot of lab work) the RBW impacts only the noise floor.Ya got to understand what you are seeking to measure as well as how your tools work.

author: Marsupilami
author_rank: Frequent Contributor
timestamp: April 11, 2022, 01:49:59 pm
content: The spectrum analyzer is an integrating instrument. With a bit of simplification in one screen position you'll see the total signal power density integrated over RBW thus the correct unit is absolute power and not spectral power density.Some analyzers have the option to show density e.g. dBm/Hz but it's only an approximation by dividing the result above by the RBW. The instrument has no notion of the actual density distribution within the single frequency bin.Following your logic, evenly distributed wideband signals would read correct marker levels, but CW would be off since all of its power is concentrated in a nearly zero bandwidth the density at that point is very high.

author: TimFox
author_rank: Super Contributor
timestamp: April 11, 2022, 01:56:23 pm
content: When the input signal is broadband noise, increasing the RBW increases the power indicated on the SA.When the input signal is a constant sine wave, so long as the RBW is centered around its frequency, the power indicated on the SA is constant.When the input signal contains harmonics, then the power in each harmonic is indicated when the SA is tuned to that harmonic frequency.

author: mattko
author_rank: Contributor
timestamp: April 12, 2022, 11:29:28 am
content: okay, so if I understand correctly... The y-axis unit is dBm, but we need to know that it was integrated over one RBW. If the signal is wider than RBW, the measurement (marker reading) will not be correct. We would have to multiply it by as much as it is wider than RBW.If we need dBm/Hz, we divide it with RBW.I think this makes sense now, yes.

author: Marsupilami
author_rank: Frequent Contributor
timestamp: April 12, 2022, 05:45:15 pm
content: That is correct.It's worth noting though, that in analog instruments the exact shape of the IF and video filters have an effect on the display, while in sampled systems the FFT windowing function contributes a lot too. (e.g. you would choose your window function for better peak magnitude accuracy display vs skirting). But unless you're hunting tenths of dBs you should be fine.

author: TimFox
author_rank: Super Contributor
timestamp: April 12, 2022, 06:59:01 pm
content: If you apply a clean sine wave of known power from a stable generator to the SA input, you will see the shape of the BW filter around the frequency of the sine wave. The peak value (in dBm or Vrms) of the display should be calibrated to the known level of the input sinusoidal voltage.

author: Marsupilami
author_rank: Frequent Contributor
timestamp: April 12, 2022, 11:41:11 pm
content: Quote from: TimFox on April 12, 2022, 06:59:01 pmThat's a good point. It's all FFT in my head, I never really had the chance to learn to think analog.

author: gf
author_rank: Super Contributor
timestamp: April 13, 2022, 06:16:59 pm
content: Quote from: Marsupilami on April 12, 2022, 11:41:11 pmWell, but a (windowed) FFT is not different in this regard. Also here, the resulting FFT spectrum is the convolution of the signal spectrum with the frequency response of the spectrum analysis filter (which in turn is the Fourier Transform of the chosen window function).

author: Marsupilami
author_rank: Frequent Contributor
timestamp: April 14, 2022, 04:58:00 am
content: Quote from: gf on April 13, 2022, 06:16:59 pmOk, fair enough, if we talk about RBW in the context of a real time sampling analyzer that will be usually true because in order to simulate the analog RBW you have to have adequate frequency resolution to smear the signal power across enough bins so that the peak (marker reading) is representative of the actual input power. What I tried to point out was that if you're just looking at any random FFT result, the magnitude of a single point will have an uncertainty related to where the input CW frequency lies between two adjacent frequency bins and you can't compensate for this purely by knowing the IF and window response.

author: gf
author_rank: Super Contributor
timestamp: April 14, 2022, 06:57:42 am
content: Quote from: Marsupilami on April 14, 2022, 04:58:00 amFor spectrum analysis purposes, a filter with a high selectivity is required anyway. The sinc frequency response of a rectangular window does not fulfil this property and is IMO not suitable for this purpose at all. But e.g. Kaiser window with beta > 13, or Gaussian window with alpha > 4.3 (just to mention two examples) have side lobes below -100dBc and a high selectivity (at the cost of a wider main lobe - there is no free lunch). Gaussian has the advantage that a qadratic interpolation of the magnitude on a dB scale is exact (if we neglect the truncation of the inifinitely long tails), aiding peak estimation between discrete frequency bins.

author: Marsupilami
author_rank: Frequent Contributor
timestamp: April 14, 2022, 05:31:23 pm
content: Quote from: gf on April 14, 2022, 06:57:42 amThat's my life philosophy.The windows in your example work, again, if you have adequate frequency resolution. (You have 10 or so bins to smear your signal over.) One case I can think of might be looking at tones very close to each other when the record length can't be increased indefinitely to get arbitrarily good frequency resolution.I might be getting on a tangent here so getting back to the original question is I wanted to point out that the only bulletproof way to get accurate power regardless of the input signal bandwidth is if you sum or integrate over a wide enough frequency range. A single marker magnitude reading will have some sort of approximating correction factor that only give accurate result for certain input and in this regard there is an extra uncertainty factor in the discrete sampled case. (I can also imagine there's something remotely equivalent in analog systems that is due to the sweep speed but again, I'm not too familiar with what's going on there.) It doesn't mean that it couldn't or shouldn't be used, but the user has to have some sort of awareness about how those values are derived and there's more nuance to that then looking at the RBW and multiplying for wideband signals.