again, there is as such no conflict between EQing the amplitude and time domain except we cannot EQ the excess phase parts (deep dips). the main reason we want to EQ the bass is that the ear does not separate the source from the room well down here (the wonderful evolutionary pre cortex processing that allowed us to locate predators inside an untreated cave). in the bass we hear the total response which has the long ringing boom from the room modes (poles). EQing that and it goes away. it’s mathematics.
-
Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!
Paul Hales on room equalization on Soundstage! Network
- Thread starter Few
- Start date
I’d like you to answer this question: if a single low note is played on a piano or a double bass in a room, would I be unable to tell where it’s coming from?
You'd be perfectly capable of locating it, but not from the actual low frequency fundamental, as that would be made difficult or impossible to locate due to the room interaction.
But you'd still locate it from the broadband transients (bow scrape/finger pick, hammer strike), and the harmonic overtones. Your brain will then understand that the source of everything including the fundamental is the piano or the double bass, so your experience will be to be able to perfectly locate the entire sound to the instrument.
agree. recent studies indicate that we have sense of direction even down to 40Hz provided that it’s in non reverberant space (eg outside). it must be the standing waves inside a room that spoils it. a large concert hall moves the problem area with discrete strong modes well below the deepest instruments. ( eg fundamental in the 1-3 Hz and already diffuse at 50Hz) unfortunately, the normal living or mixing room puts the problems right on top of the bass instruments
Thank you for the reply.
However, the microphone that captures those harmonics also captures the room's temporal spatial characteristics. During playback, all this information allows our brain to reconstruct the piano spatially; the brain does not rely solely on the fundamental frequency within our listening room to reconstruct the piano. Therefore, I consider it impossible to compare "direct sound" correction with passive correction
Currently I must admit I am not sure what your point is. Can you please elaborate or rephrase what your stance is, and how it contradicts (if it does) anything I have said or anyone else has said in this thread?
@Suono I am partly guessing what your point is, so let me know if I am misunderstanding.
You're saying the microphone is recording both the room and the direct sound of the original recording venue. This can be true, but often instruments are Close-miced, and reverb is added during production. But assuming it is captured from the recording venue, it is still the higher frequency elements of the sound that is used to localize the sound, so in essence you're saying the exact same thing I did? So EQing below 100hz will not remove our ability to still successfully "reconstruct the piano" so when the recording is reproduced. And our room modes are still problematic and altering the original recording, so we are still right to EQ them down.
There are certainly low frequency ambience-type cues in a live recording that can add to the sense of scale and room. But this will not be diminished by EQing down peaks.
The post-EQ result will be a more, not less, accurate version of the piano.
You're saying the microphone is recording both the room and the direct sound of the original recording venue. This can be true, but often instruments are Close-miced, and reverb is added during production. But assuming it is captured from the recording venue, it is still the higher frequency elements of the sound that is used to localize the sound, so in essence you're saying the exact same thing I did? So EQing below 100hz will not remove our ability to still successfully "reconstruct the piano" so when the recording is reproduced. And our room modes are still problematic and altering the original recording, so we are still right to EQ them down.
There are certainly low frequency ambience-type cues in a live recording that can add to the sense of scale and room. But this will not be diminished by EQing down peaks.
The post-EQ result will be a more, not less, accurate version of the piano.
This is incidentally the same reason why I can build systems with subwoofers that are crossed over at 100hz without localization issues. The transients and harmonic content anchors the sound to the speaker, even for tones and instruments with a 40hz fundamental.
What would you define as bass impact?
There will naturally be less bass (and potentially also feeling of impact and punch) when reducing peaks, but that's intentional, it was likely too much to begin with.
What would you define as bass impact?
There will naturally be less bass (and potentially also feeling of impact and punch) when reducing peaks, but that's intentional, it was likely too much to begin with.
The bass impact is more about the “punch in the chest”, which the overtones and transients have nothing to do with.
The question is if the lack of impact the guy is talking about in the podcast is simply because of the lesser energy happening when the peaks are shaved off, or if it indeed has something to do with the direct sound losing the impact in favor of equalizing the problems with the room acoustics.
DBA bass makes some frontal and outdoor-ish impression.
If you get close to the speaker, so that mono sound is heard from it, not stereo. Where will you hear the bass drum? In my system I noticed several times how the bass drum sound can be perceived as coming from the horn tweeter
(666 Hz crossover). Even though the subwoofer is near there. Informal random observations. I'll have to try it carefully with different signals.P.S.: I'm interested in the experiences of other listeners.
Last edited:
I doubt that long-term resonance has much\exessive impact. Or maybe this is not the right impact))(and potentially also feeling of impact and punch) when reducing peaks, but that's intentional, it was likely too much to begin with.
This is a silly argument on his part. It assumes that the direct sound in one's room is 'the' best possible direct sound you can get in that room. What are the chances of that?
At the MLP it surely is no longer the virgin 'direct sound' emerging 0.01 meters from your drivers. Nor is it what the engineer heard.
It is any of that plus what the room does to it between the drivers and your ears. We all know this.
The modal colorations the room adds to bass in a typical room (not a concert hall!) often produce imbalanced bass at the MLP.
As Amir notes, if you only listen to piano music this might not matter much ... unless it's a left-hand piano concerto (like Ravel's) . The lowest note on 88 keys is 27 Hz.
Last edited:
Bass impact is (dP/dt) change in pressure over change in time. Adding DSP filters in many cases introduces group delay and ringing which can have the effect of stretching the time over which the energy of a transient reaches the listener, reducing the impact of the transient. When you combine this with distributed subs, which don't form a point source wave front, impact can be noticeably reduced at the LP. In my experience using a point source (as close as possible anyway) for all frequencies and trying to maintain time domain integrity (energy from all the frequencies of a transient arriving at the LP at the same time) is the best way to increase "Slam" / "punch in the chest". YMMV.
Last edited:
audiofooled
Addicted to Fun and Learning
- Joined
- Apr 1, 2021
- Messages
- 756
- Likes
- 805
A kick drum transient starts with mode 0 of a circular membrane, that has no definitive pitch or timbre quality to be detected as it is primarily a body sensation. It is usually a high energy density acoustic event during which the air itself becomes nonlinear (at pressure over 140 dB). Yes, it's high dP/dt, but also a vector quantity. It is difficult to even record properly because it changes spectrum over distance as the initial wavefront modifies the time of arrival of the harmonics introduced by other modes of the membrane.
In case it is a well recorded kick drum, I see no problem in using DSP, distributed subs, to tame the room modes, and it will of course sound better in terms of pressure (what we hear). However, for reproducing the body sensation itself with even remote fidelity, time alignment across frequency is very important. One can care about this or not, just hear it, or both hear it and feel it. I think it's a fair point either way and a preference thing.
In case it is a well recorded kick drum, I see no problem in using DSP, distributed subs, to tame the room modes, and it will of course sound better in terms of pressure (what we hear). However, for reproducing the body sensation itself with even remote fidelity, time alignment across frequency is very important. One can care about this or not, just hear it, or both hear it and feel it. I think it's a fair point either way and a preference thing.
Assuming time alignment accross frequency is the goal DSP can help or hinder. Most DSP focus is on smooth FR which can conflict with time alignment. Similarly distributed subs certainly can help smooth FR but you lose the ability to somewhat recreate the original wave front and even for pure time alignment distributed subs have a small sweet spot compared to semi point source full range speakers. Chasing "slam" is hard and can conflict with other priorities.
And I think that pursuing high fidelity means, first and foremost, this.
again, there is no conflict between FR and time domain. It is a bijection (one to one map) via the Fourier Transform. If the response is minimum phase then a flat FR means a delta pulse in the time domain. The main issue in a closed room is that we get resonances where the sound bounces infinitely back and forth for seconds. This can be EQed. What cannot be EQed is the non minimum phase part like the deep dips from boundary reflection cancellation and interference between two standing wave modes. This is very position dependent and a short duration phenomenon. the reflections are also part of the experience of space. OTOH, the standing wave modes are global in the room and last for seconds which is the main problem
The key.
Similar threads
