HD audio converter file format and resolution (software)

audiventory · « **on:** 12 Sep 2014, 04:49 am »

Hello friends and colleagues.

My name is Yuri Korzunov. I’m founder and developer of Audiophile Inventory (created in 2007).

Our software product

AuI ConverteR 48x44 - HD audio converter software file format and resolution [Windows, Mac OS]. The software intended for music production and audiophiles.

http://samplerateconverter.com

Supported formats

You can convert now (September 2014):
1. PCM to PCM
2. PCM to DSF
3. DSF/DFF to PCM/DSF
4. ISO to PCM/DSF
5. CD audio to PCM/DSF (metadata [track name, album, year, …] auto obtained from Internet)

Here:
1. PCM is WAV, FLAC, AIFF, ALAC, wma, mp3, aac,…
2. DSF and DFF is 1-bit/D64/D128 audio file formats
3. ISO is 1-bit HD audio disk format

You can connect to AuI own command line codecs for extension of input/output formats range.

Why AuI ConverteR?

You get with AuI ConverteR 48x44:
1. Assurance in sound quality
2. Simple user interface and control
3. Fast and reliable technical support

AuI ConverteR 48x44 has:
1. on the average 1 … 2 update per month,
2. permanent researching for sound improving, supporting new formats and functionality extension.

Technical details

Own Audiophile Inventory's audio processing algorithms:
AuI alphaC - sample rate and bit-depth conversion
AuI alphaD - dithering
AuI alphaS - [PCM to 1-bit/D64/D128] and [1-bit/D64/D128 to PCM] conversion

All internal math in 64-bit float (double precision).

Opinions

Opinions of owners of AuI ConverteR and users of FREE version you can read here:
http://samplerateconverter.com#opinions

Check FREE version with full audio performance

http://samplerateconverter.com

Contact me http://samplerateconverter.com/contact

Best regards,
Yuri Korzunov

audiventory · « **Reply #1 on:** 16 Sep 2014, 06:59 pm »

Tips and tricks: How Convert ISO to DSF (1-bit high resolution audio)

http://youtu.be/MYv29q0jZm0

Phil A · « **Reply #2 on:** 16 Sep 2014, 07:46 pm »

Thanks for the post (saved the site). I'll have to give it a look. Over the winter, I have some last ripping stuff planned (some DVD-As mainly) and at least some conversions (a few DFF to DSF files and depending on what I do with LMS in a secondary system with a Teac DAC on a Squeezebox Touch perhaps some DSF to PCM, unless I do beta version 7.9 or do EDO which I have not got to work yet). I probably can use some DSF to PCM for an Oppo BDP-93 which is in the master bedroom and feeds the outdoor speakers too. I did some via dbPoweramp's converter but the output on the Squeezebox Touch is low (via the coax out to the Teac DAC) and it also does them at 24/96 vs. 24/88.2.

audiventory · « **Reply #3 on:** 17 Sep 2014, 09:39 am »

Phil,

I have some questions:
1) What is version 7.9 (it's AuI v. 3.7.9)?
2) What is LMS?

Thank you,
Yuri

JohnR · « **Reply #4 on:** 17 Sep 2014, 09:42 am »

Hi Yuri, LMS is Logitech Media Server.

audiventory · « **Reply #5 on:** 17 Sep 2014, 09:43 am »

Thank you, John.

audiventory · « **Reply #6 on:** 25 Sep 2014, 09:13 am »

Now released new AuI ConverteR 48x44 v.3.8.7

1. Added volume control -40 ... +40 dB and 0.1 dB step (for version PROduce-R and PROduce-RD).

2. Fixed some bugs with BWF WAV files, generated by Pro Tools.

Read more http://samplerateconverter.com/content/new-aui-converter-48x44-v387-released

steve in jersey · « **Reply #7 on:** 25 Sep 2014, 03:23 pm »

Yuri, Hello & welcome to the AudioCircle !

I've been Oversampling my CDs for some time as I find that this really suits the highly dense dynamic contrast Orchestral (& other
acoustic based) music sound quite a bit different than it does at 16bit/44.1 Khz . I realize that there is'nt any "new" information being added, but that the sound improvement may be due to "a reduction of various forms of filter-induced phase distortions". I am actually
not knowledgeable enough to come to this conclusion on my own, but it makes extreme sense to me from what I'm hearing. No one
can actually hear even 44.1 khz so the lack of musical clarity has to be due to something other then just frequency retrieval.

I'm sorry to bore you with information that I'm sure you're already aware of, as the real reason I've written is that I'm having
some difficulties with your test conversion program. I'm not quite sure how to select the output path for the converted files as I need to
place them on an external drive . I have been using JRMC for my conversions only as I use a seperate Digital File Player for playing my
files

Thank you & once again welcome to AudioCircle,
Steve

barrows · « **Reply #8 on:** 25 Sep 2014, 03:47 pm »

Audiventory: Would you please share some technical details of the digital filters incorporated with your oversampling algorthms? For example, when oversampling 44.1 files, is the digital filter applied a symmetrical linear phase type, or some variation on a minimum phase type with less/no pre-rining? Is the filter used a straight forward steep brickwall type with no alias products or do incorporate a less steep filter allowing some aliasing but reducing ringing?
When converting PCM-DSD, same questions as above, and, what is the process: for example, do you oversample 44.1 to 176.4 or 352.8, and then go to DSD (or DSDx2 is that an option), and if you go to 176.4 or 352.8 first, what type of filters are applied (as above).
Thanks for any details you care to share.

Phil A · « **Reply #9 on:** 25 Sep 2014, 03:58 pm »

The above would be of interest to me as well if I need to go that route

steve in jersey · « **Reply #10 on:** 25 Sep 2014, 05:31 pm »

Once I've finished my lunch, I hope to go home & "play around" with the test converter program

If I have any success I will post my impressions of how this appears to compare with the algorithms that JRMC
uses. (My JRMC converted files have spoiled me to the point that I have no interest in playing CDs anymore,
unless I'm having a network problem & am unable to start my Auraliti PK100 which uses MPaD to control it.)

24bit/ 176.4 khz really appears to be the "sweet spot" of my DAC ,so that's what I try to "feed it".

steve in jersey · « **Reply #11 on:** 25 Sep 2014, 07:00 pm »

I'm sorry to say that I was unable to figure out the correct procedural steps to operate the program.

Luckily, I've been pretty satisfied with my results with using JRMC so I'll continue to use it for my file converts.

(I was thinking how uncanny "reducing various phase-induced distortions to gain more clarity" could be a description of my life; Yeah, we'd all be on the path to "Uptopia" if we were able to "see" those distortions early enough. Well, I can at least make my music more enjoyable)

Have a Great Day

jimtranr · « **Reply #12 on:** 25 Sep 2014, 09:50 pm »

Quote from: steve in jersey on 25 Sep 2014, 07:00 pm

I'm sorry to say that I was unable to figure out the correct procedural steps to operate the program.

"Open files" (select the file you want to convert)
Select "Output" (Format, Bit depth, Sampling rate)
Press "Start"

Worked on the first try converting a 16/44 to DSD64 DSF. You'll find the converted file in the ConvertResult subfolder of the folder containing the original file you converted.

Plays and displays in my JRMC20 and my TEAC DAC as a DSD64 file.

audiventory · « **Reply #13 on:** 26 Sep 2014, 05:16 am »

Hi Steve,

> I've been Oversampling my CDs for some time as I find that this really suits the highly dense dynamic contrast Orchestral (& other
acoustic based) music sound quite a bit different than it does at 16bit/44.1 Khz.

Analog filter does restoring signal from 16/44. This process very like to oversampling. Thus we oversample signal we do work of analog filter.
Demands for filter on oversampled sampling rate is less hard in this case.

> I realize that there isn't any "new" information being added, but that the sound improvement may be due to "a reduction of various forms of filter-induced phase distortions".
AuI don’t add info. It simply carefully restore it from 16/44. AuI ConverteR 48x44's conversion algorithm alphaC helps to analog filter.

> I'm not quite sure how to select the output path for the converted files as I need to
place them on an external drive.

You can select 2 types of paths:
1. Relative
Example:
Input path: c:\music
Path into field «Directory output files» of main AuI’s window: Album
Thus output fule: c:\music\Album
2. Absolute
Example:
Input path: c:\music
Path into field «Directory output files» of main AuI’s window: c:\Album
Thus output fule: c:\Album

You can recreate full directory's structure of input files into path from «Directory output files». Need turn ON middle directory's button.
If you want recreate only directory containing files, need turn ON most right directory button too.

As output directory you select your external drive's folder as usual folder. In select path window you can create folder. Or input it manually into «Directory output files» (will created automatically).

Best regards,
Yuri

audiventory · « **Reply #14 on:** 26 Sep 2014, 05:17 am »

Hi barrows,

> when oversampling 44.1 files, is the digital filter applied a symmetrical linear phase type, or some variation on a minimum phase type with less/no pre-rining?

You can select preferable filter into Settings window: turn ON/OFF "Use minimum phase filter…» (available into FREE, Audiophil-E, PROduce-R, PROduce-RD)

>Is the filter used a straight forward steep brickwall type with no alias products or do incorporate a less steep filter allowing some aliasing but reducing ringing?

General rule: provide maximal suppressing of aliases with minimal loss of useful band.
It manually optimized filter for each pair input/output sampling rates

For linear phase filter:
1. maximal suppression of aliases - minimum ringing.

For minimal phase filter:
1. maximal suppression of aliases - minimum ringing
2. maximal linearity of phase response - minimal pre-ringing and post-ringing.

> When converting PCM-DSD, same questions as above, and, what is the process: for example, do you oversample 44.1 to 176.4 or 352.8, and then go to DSD (or DSDx2 is that an option), and if you go to 176.4 or 352.8 first, what type of filters are applied (as above).

44/48/88/…/352/384 kHz oversampled to 2.8/5.6 MHz and applied delta sigma modulation.
Filters same. You can use as linear phase filter as minimal phase filter.

Best regards,
Yuri

audiventory · « **Reply #15 on:** 28 Sep 2014, 05:34 pm »

Here described how connect dff2dsf utility (as input decoder) to AuI ConverteR 48x44 under Mac OS X and Windows.
Now AuI ConverteR read intermediate DSF (converted by dff2dsf from input DFF) without conversion to PCM.

Need download last AuI's update 3.8.8 or later.

http://samplerateconverter.com/content/conversion-dff-dsf-3-ways-mac-os-x-windows-and-linux

audiventory · « **Reply #16 on:** 4 Oct 2014, 05:50 pm »

New AuI ConverteR 48x44 3.8.11. Improving DFF and DSF conversion

http://samplerateconverter.com/content/new-aui-converter-48x44-3811-improving-dff-and-dsf-conversion

audiventory · « **Reply #17 on:** 11 Oct 2014, 06:37 pm »

New AuI ConverteR 48x44 3.8.17. Fixed some audible clicks for DFF and DSF conversion
http://samplerateconverter.com/content/new-aui-converter-48x44-3817-fixed-some-audible-clicks-dff-and-dsf-conversion

audiventory · « **Reply #18 on:** 17 Oct 2014, 06:26 pm »

HD Audio Converter AuI ConverteR 48x44 Tested under Mac OS X Yosemite http://samplerateconverter.com/content/hd-audio-converter-aui-converter-48x44-tested-under-mac-os-x-yosemite

audiventory · « **Reply #19 on:** 14 Nov 2014, 02:11 pm »

Here described how simply remove noise for high resolution audio files (appear during playback) with audio converter

http://www.youtube.com/watch?v=67_3Qmbq8Y4