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Is it necessary to AUTOMATICALLY change the convolution filters in accordance with the changes in the sample rate of the reproduced material?

Is it important to have an AUTOMATIC change of the convolution file?

  • I don't listen to mixes or playlists - it's not tedious to change the filter when changing an album!

    Votes: 0 0.0%

  • Total voters
    11
I use TIDAL (streaming service). Yesterday I deleted JRiver because it doesn't support 192, which is a lot in Tidal's classic albums. I don't watch movies - in this case, do I need to take care of reducing latency?

I'm not sure if I understood what I quoted correctly, but I have a PEQ*39 bands, which gives a 1 dB deviation from the target from 20 to 16k. Is this enough to do something that I completely misunderstood from what is written in the continuation of the quoted words?

If you fix the frequency of the DAC, then what frequency should you choose to ensure the least harm due to resampling (judging by everything I've heard in different places, I'm not interested in latency)?
Did you ever even try to analyse supposed hi res files from Tidal? Over 90% are blanks. JRiver suports both windows driver's and ASIO including what ever DAC can output + it's own multichannel encoder and decoder (PCM and old Dolby). Your DAC eventually need USB audio 2.0 driver if it doesn't go over 96 KHz and it supposedly should be.
 
I've seen several descriptions of creating a convolution filter - I don't understand a single word in any of them except REW. One of the descriptions I saw caught my attention by praising the results and having a list of milestones-but here, too, I didn't understand a single word. if it seems reasonable and effective to you what is obtained as a result of this procedure, then I will try to understand how to implement the terms mentioned in it (the last post in this thread:

Thank you in advance for your opinion!
 
For HiFi, I have foobar talking direct to my DAC/HiFi speakers using ASIO/WASAPI.
Therefore, I only need to EQ/DSP what foobar is playing
In your opinion, which is better and gives better results - your foobar or APO in a situation where other applications besides TIDAL will not interfere with it in a dedicated TIDAL computer?
 
Did you ever even try to analyse supposed hi res files from Tidal? Over 90% are blanks. JRiver suports both windows driver's and ASIO including what ever DAC can output + it's own multichannel encoder and decoder (PCM and old Dolby). Your DAC eventually need USB audio 2.0 driver if it doesn't go over 96 KHz and it supposedly should be.
I listen to Tidal's classics, and 192 albums often sound better. In addition, I do not need to check the integrity of Tidal because I do not have the opportunity to replace tracks 192 with the same tracks 96.
My DAC also works on 786 and, apparently, has USB-2 (D90SE).
 
I've seen several descriptions of creating a convolution filter - I don't understand a single word in any of them except REW. One of the descriptions I saw caught my attention by praising the results and having a list of milestones-but here, too, I didn't understand a single word. if it seems reasonable and effective to you what is obtained as a result of this procedure, then I will try to understand how to implement the terms mentioned in it (the last post in this thread:

Thank you in advance for your opinion!

You can use REW + rePhase to generate linear-phase FIR filters, but you really need to know what you are doing. REW is measurement software, it was not designed to be filter generation software. Because of that, it lacks some very important features, such as the ability to generate linear phase crossovers. For that, you need rePhase. And it is very difficult to use. It is not impossible, but it is difficult. This is why most people prefer to use dedicated filter design software, like Acourate, Audiolense, Focus Fidelity, FIR Designer, and so on.

But even before you begin to do any of that, you need to understand what the terms "linear phase" and "minimum phase" even mean. There are people who use minphase DSP, and other people use linphase DSP. I tried to explain everything in this thread but I have received feedback in that thread from some people who said they did not understand. So have a go, try to understand the basic concepts.
 
Who says it does not support 192kHz? This is an old version of JRiver (MC31).
I found out this and received confirmation from people who said that "ANDROID does not work with JRIVER on 192" (I did not check 176) and that JRiver people ignore contacting them about this problem.
why is 192kHz so important?
At least because it exists and I don't want to be UNABLE to listen to 192 albums/tracks. (to be clear: sometimes a things are exist into TIDAL as the 192 versions ONLY).
 
In your opinion, which is better and gives better results - your foobar or APO in a situation where other applications besides TIDAL will not interfere with it in a dedicated TIDAL computer?

Actually, it depends on your use case.

1. If you have only one audio path (eg. one set of desktop speakers connected to PC), APO might be better option.

2. If you have dual or multiple audio path (eg. desktop speakers, separate DAC/HiFi speakers connected to same PC), doing the DSP inside foobar might be better.

My use case is #2.

About your example of "dedicated Tidal computer" -- if you only have one audio path, probably APO might be better for you.
 
In what sense (in what situation) does REW do this?

REW software limitation.

2025-04-25_2352 roomeqwizard EQ_filters.png




"The Generic settings allows 20 parametric filters."
 
if you only have one audio path, probably APO might be better for you.
I see, thanks! Now, please, which one is better in terms of sound QUALITY?
In addition: does the foobar interfere with the operation of the system during peak system operation situations (what processor/memory do you have)? Stuttering, glitches in encoding/equalization? The EAPO often pleases me with such decorations.
 
I found out this and received confirmation from people who said that "ANDROID does not work with JRIVER on 192" (I did not check 176) and that JRiver people ignore contacting them about this problem.

At least because it exists and I don't want to be UNABLE to listen to 192 albums/tracks. (to be clear: sometimes a things are exist into TIDAL as the 192 versions ONLY).
Which Android? You need to understand what simple I/O is. I don't mind you giving them money and listening what ever "masters" (of the universe) you want. If you are saying you will hear same master difference when played at what ever including DSD256 to 96 KHz, you won't nothing over 20 K per baby ear more realistically 16 K as mid age adult. When you learn music theory you realise it's really not critical over 12.5 KHz.
 
I see, thanks! Now, please, which one is better in terms of sound QUALITY?

I dare not answer this question for 2 reasons. One, I am not so familiar with APO. Two, I will be challenged by some people that I haven’t done ABX test therefore what I say is rubbish or hallucinations.

In addition: does the foobar interfere with the operation of the system during peak system operation situations (what processor/memory do you have)? Stuttering, glitches in encoding/equalization? The EAPO often pleases me with such decorations.

On my old powerful PC, no glitches. i7-6800K. 48Gb RAM. foobar with all the eye candy takes about 28%cpu. if i remove all the eye candy maybe <8%cpu.

long time back, i encounter intermittent playback glitches when there is huge network LAN transfer going on. there is a registry tweak to fix this issue so that windows will deprioritise network traffic. i did the tweak and after that no more audio glitching.
 
REW software limitation.

View attachment 446694
yea, you are right - I probably forgot that I was making PEQ as a configurable equalizer... But something tells me that in the end I managed to confuse REW and he agreed to create a 31-lane GENERAL-style PEQ. The problem is that I do not know which filters to choose from the possible ones in the Configurable mode so that they match what the DSP implies (the EQ-APO in this case). And there's no one to ask :( . Only the @JohnPM ... I would even dare to beg to loosen the REW restrictions on the number of PEQ's lanes (20 and 31)...
1745597454441.png
 
there is a registry tweak to fix this issue so that windows will deprioritise network traffic.
Tell me, please this tweak!
I would also like a way to increase the priority of the EAPO DSP, but I can't find
it in the list of processes...
 
You need to understand what simple I/O is. I don't mind you giving them money and listening what ever "masters" (of the universe) you want. If you are saying you will hear same master difference when played at what ever including DSD256 to 96 KHz, you won't nothing over 20 K per baby ear more realistically 16 K as mid age adult. When you learn music theory you realise it's really not critical over 12.5 KHz.
I've read that people are able to distinguish between DSD and PCM sounds of similar frequencies. with good DACs, I can also hear the difference between 11 MHz and the corresponding PCM. Of course you understand. that's not just a matter of frequency (and sampling rate), but also of implementation.

44 and 192/96 PCM playback is also different. you can hear this too by yourself, there are sets of such examples.
 
for 2 reasons. ... Two, I will be challenged by some people that I haven’t done ABX test therefore what I say is rubbish or hallucinations.
I'm MORE interested in what they say and think about the effectiveness of homeopathy than what they think or have told you about you - I'm not at all interested in what someone who overhears our private conversation will say. Therefore, please tell me, which works more steadily and efficiently in your "intuitive and unproven" opinion? Thank you in advance: the chain with foobar is long and it will demand me for long learning
 
I've read that people are able to distinguish between DSD and PCM sounds of similar frequencies. with good DACs, I can also hear the difference between 11 MHz and the corresponding PCM. Of course you understand. that's not just a matter of frequency (and sampling rate), but also of implementation.

44 and 192/96 PCM playback is also different. you can hear this too by yourself, there are sets of such examples.
I mastered, whose in touch and participated in studis when I was a bit younger. Nothing to do with sample rate but sample time (SINAD) and analytical listening on rather loud and above average SPL and even so subtle 24 bit to 16 bit without dithering and with a good one not even that. Difference between master's on other hand is very, very real! Good, great and broken. Great is when something extraordinary happens during recording and it simple turns out like that, good and technically done excellent even with gear shortcomings and pretty much neither done good or correct and simply broken (and that's a lot of music).
 
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