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A toast to PGGB, a heady brew of math and magic


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33 minutes ago, dmance said:

ZB…PGGB has to be integrated into playback software (please!) for automatic conversion and file management. In the interim, I will work on my own workflow components for my continued testing.

 

Combining both PGGB and Taiko Audio Server should be a marriage made in heaven

 

https://www.facebook.com/SuncoastAudio/posts/3961771397212542

 

However, Taiko Audio might be an Extreme-exclusive unless they're changing their mind at some point.

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29 minutes ago, basillus said:

Hi, does the Chord Qutest, also have the “bad” Amanero usb chipset? 

 

Let's check all of their products listed on the official site

 

https://chordelectronics.co.uk/product/dave/

https://chordelectronics.co.uk/product/mojo/

https://chordelectronics.co.uk/product/qutest/

https://chordelectronics.co.uk/product/hugo-2/

https://chordelectronics.co.uk/product/hugott2/

https://chordelectronics.co.uk/product/hugo-mscaler/

 

All of them were pointing to the same Windows-10-768KHz-driver.zip driver and obviously they're sharing the same hardware from Amanero.

  

29 minutes ago, basillus said:

If not, will it’s “newer” usb chipset, also benefit to get bypassed with the

AudioWise SRC-DX bridge?

 

It might be difficult to predict, maybe that would depend on what's actually sitting in front of Qutest?

 

https://www.whatsbestforum.com/threads/taiko-audio-sgm-extreme-the-crème-de-la-crème.27433/page-371#post-719062

Quote

Here's a speculative comment regarding this AudioWise USB to BNC bridge for Chord DAVE owners. I'm not convinced that bypassing Amanero on the DAVE has the same impact if you don't own an ultra low latency music server like the Extreme. For example, having owned the InnuOS Zenith SE Mk2 once upon a time, I would not consider that server to be ultra low latency. This applies to any other server that utilizes a low power CPU including an Aurender W20SE which I also had here. I thought my best DIY server effort was pretty low latency (as it outclassed both the InnuOS and the Aurender) until my Extreme arrived and the Extreme made that server sound slow and smeared and sleepy. So anything with higher latency than the Extreme that is introduced into the path after the Extreme will be a bottleneck to the Extreme and Amanero is clearly a bottleneck. But with other music servers, one may find that the Amanero in DAVE isn't really a bottleneck and so in these instances, the AudioWise bridge may not offer much. This is why I said YMMV.

 

Maybe not THAT many motherboards were "proven" to be a good fit for an ultra low latency music server except for these guys?

 

Asus WS C621E SAGE

https://www.asus.com/Commercial-Servers-Workstations/WS-C621E-SAGE/

 

Supermicro X10DRU-XLL

https://www.supermicro.com/en/products/motherboard/X10DRU-XLL

 

Supermicro X11DPU-XLL

https://www.supermicro.com/en/products/motherboard/X11DPU-XLL

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https://www.head-fi.org/threads/cx-ex-k-s-antipodes-owners-unite.932942/page-46#post-16466560

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You will need to use an MPD (not upnp) controller such as Rigelian to control direct play MPD. For PGGB files stored on your server it is the best sonics I have found. Beats out HQPlayer and Squeeze imho.

 

https://www.head-fi.org/threads/cx-ex-k-s-antipodes-owners-unite.932942/page-46#post-16466931

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Thanks for that. Easy as heck to get this running. I already had 8player on my iPhone so it was easy to get music playing. Bloody hell this sounds great. Yeah I know @Triode User told me so, but I can occasionally be stubborn.

 

Well, I guess that's gotta be convincing enough for some of us to jump ship then?

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IMHO 4GB of RAM should be a non-starter for PGGB, even their $60 plan would provide just under 30GB of RAM.

 

So we're looking for something with plenty of RAM on top of unmetered + uncapped + unthrottled traffic while the price is right, aren't we asking for too much?

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Could 4FS via I2S output be a bottleneck of some sort?

 

https://6moons.com/audioreviews2/pinkfaun/1.html

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The associated challenges coincided nicely with Pink Faun's R&D to have Mattijs design an I²S interface for a personal computer. This bridge as Mattijs calls it is a plug-in board built around the CMedia CM8888 multi-channel audio processor.

 

https://www.cmedia.com.tw/products/PCI_PCIe_AUDIO/CM8888

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  • Six pairs of I2S serial audio output interfaces (12-ch out)
  • Integrated 192K/176.4K/96K/88.2K/48K/44.1K, and 16/24/32-bit S/PDIF transmitter with a 2-source selector/mux (from playback DMA digital mixing, S/PDIF input), including WMA-Pro output support

 

So far most happy campers seemed to be enjoying PGGB with their systems that are capable of reaching 8FS / 16FS / 32FS.

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I2S in and of itself ain't a problem at all since it's able to reach 32FS for sure, though something else was definitely holding it back

 

https://audiophilestyle.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-computer-audio-streaming/page/574/?tab=comments#comment-967495

On 6/22/2019 at 12:14 PM, numlog said:

The CM8888 PCIe to I2S chip only supports 192kHz.

 

 

Metrum Amethyst is only capable of accepting 8FS via USB while CM8888 inside Pink Faun I2S Bridge is limited to 4FS, did you realize what's going on behind the scenes?

 

Upsampling to 8FS (PGGB)

Downsampling to 4FS (CM8888 inside Pink Faun I2S Bridge)

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While I fully understand the fact that PGGB-RT could only handle individual tracks (i.e. actual lossless music format) that are stored on local drives / NAS, reading these threads by @Cebolla below provided some hints for bridging foobar2000 to LMS

 

https://forum.vb-audio.com/viewtopic.php?t=747

https://www.avforums.com/threads/streaming-mqa-using-tidal.2076126/page-2#post-24721823

https://audiophilestyle.com/forums/topic/29124-how-to-route-signals-to-acourate-convolver-using-asio4all-vb-cable/#comment-576854

 

Someone wrote this Wave Input plugin for LMS as follows

 

https://sourceforge.net/projects/bpaplugins/files/OtherPlugins/WaveInputWin-v105.ZIP

https://forums.slimdevices.com/showthread.php?35718-Announce-Wave-Input-plugin-v1-00

 

Basically we could do something similar to this by replacing HQ Player with LMS

 

https://www.superbestaudiofriends.org/index.php?threads/the-hqplayer-thread.11376/

 

foobar2000 → PGGB-RT → HIFI-Cable & ASIO Bridge → Wave Input → LMS & C-3PO → Squeezelite → USB DAC

 

Though not sure if that were able to go above 96KHz

 

https://forum.vb-audio.com/viewtopic.php?t=397

 

Then there's also Reaper and Synchronous Audio Router

 

http://sar.audio

https://www.reaper.fm/download.php

https://www.bilibili.com/read/cv7364768

https://github.com/eiz/SynchronousAudioRouter

https://groupdiy.com/threads/768-khz.74336/#post-941017

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Reaper handles 768 kHz just fine. I suspect some other DAWs do too.

 

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https://audiophilestyle.com/profile/12166-kelvinwsy/?tab=field_core_pfield_3

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iFi iDSD BL to 300B tube monobloc amp output to resistance box to HD800 modified with analixus foam mods/Hifiman HE560/AudioTechnica AD1000/Sonorous VI Headphones

 

https://audiophilestyle.com/forums/topic/19715-hq-player/page/922/#comment-1174168

19 minutes ago, kelvinwsy said:

I ran conparisions of 705.6 PGGB fikes(44.1 khz orig) with zero filters and dither.. The PGGB tracks sounded vague and stereo location of instruments not dead center was off .. sort of vague location.. PCM upsampling to ..705.6 gave far suoerior cleaning up of the sound of the original 44.1 khz source track.. 

This was done with Hqpkayer 4.15.1!!

DSD512/EC7v2 wiped the floor vs the PGGB tracks.. 

I had done same comparisions with Hwplayer  4.13.1 doing 705.6 khz PCM upsampling.. Results were not so conclusive to my ears! 

Hqplayer 4.15 has made great SQ improvemts even for PCM upsampling..

 

Very bold statements IMHO, maybe results really are DAC-dependent (i.e. no such thing as NOS from iFi) while we might also need either MPD or LMS + Squeezelite to keep PGGB tracks "untouched" so to speak?

 

https://ifi-audio.com/products/micro-idsd-bl/

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PCM 768/705.6/384/352.8/192/176.4/96/88.2/48/44.1kHz

All native decoding, no internal hardware conversion

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On 7/3/2021 at 1:25 PM, romaz said:

Where PGGB has the potential to be a leveler is with DACs, especially DACs incapable of high sample rates.  For example, I found the $1,500 Gustard X26 Pro + PGGB to outperform (at least to my ears) another much more expensive DAC without PGGB.

 

PGGB (especially PGGB EQ) has the potential to also be a leveler with amplifiers.  I am still seriously astounded by how much better my bass has gotten with PGGB EQ.  I feel like I have seriously upgraded my amplifiers.

 

https://www.head-fi.org/threads/cx-ex-k-s-antipodes-owners-unite.932942/page-71#post-16597377

https://www.head-fi.org/threads/cx-ex-k-s-antipodes-owners-unite.932942/page-74#post-16621566

https://www.head-fi.org/threads/cx-ex-k-s-antipodes-owners-unite.932942/page-92#post-16672302

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The Gustard was a significant step up from the TT2, which is astonishing given the price difference. PGGB is a factor in this as I run the Gustard in NOS mode. It‘s not truly non-oversampling but it lets the magic come through. Likewise for scaling with HQPlayer for everything else.

 

https://audiophilestyle.com/forums/topic/63585-article-denafrips-terminator-ii-review-and-comparison/#comment-1150886

On 8/4/2021 at 3:32 AM, Fourlegs said:

As to the Holo May, I owned a L2 one for three weeks and just couldn’t like it no matter how hard I tried. But then I was comparing it to a Dave powered by a Sean Jacobs DC4 LPS. I had hoped that the Holo May would shine when playing PGGB 16FS or even 32FS files in NOS mode but it was not to be.

 

Most likely Gustard should be one of the best choices for PGGB, its I²S input is also compatible with Titans Audio Lab Helen

 

https://www.head-fi.org/threads/review-titans-audio-lab-helen-jitter-reducer-signal-enhancer.895007/

 

So we could do something like Gustard S16 → Titans Audio Lab Helen → Gustard X26 Pro or music server → Gustard U18 → Titans Audio Lab Helen → Gustard X26 Pro etc.

 

Those 4K HDMI cables from Nordost / 8K HDMI cables from AudioQuest are great for I²S connections

 

https://forum.psaudio.com/t/which-hdmi-cable-are-you-getting-for-your-new-perfectwave-sacd-transport/21409

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On 1/10/2022 at 2:12 AM, Zaphod Beeblebrox said:

Offline processing at higher rates does come with a storage cost and it may not be for everyone. There is also the option for close to real-time processing using PGGB, for now it is available as a foobar-plugin and hopefully we could expand to other players when time allows.

 

FYI - this was posted before the recent turmoil kicked in (January 6th / 7th depending on the time zone) and foobar2000 turned out to be the winner once we've got the total number of processes under Win10XPE *WAY* down to 12 or so

 

https://jplay.eu/forum/index.php?/topic/4410-windows-11-pe-audiophile-creation-guide/?p=58068

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Now I can say without any doubt that F2K is on Win 10-11 XPE, well above both HQP (3x and 4x versions) and JRMC 26 (the last one I have registered) and WTFPlay 0.75.
 It is a shame to see that, after spending a few €, 2 totally free applications are far superior to the paid ones.
 
Right now F2K and Infinity Blade are far superior. Comparing HQP (WinServCore2019 RAMDISK) against HQP (Win10-11XPE) the latter is clearly superior, the same goes for JRMC26, it sounds much better on Win10-11XPE, but both clearly worse than F2K.

 

Here's yet another thread that should be started back in 2009 and once again foobar2000 was doing well under Windows PE 

 

https://www.google.com/search?q=8705+PE+site%3Aerji.net

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根据感人程度,SAW理论上可能会更自然和贴近原声。 看来SAW确实有一套。当然,还没有在WINPE下比较过,因为PE下FOOBAR会改善不少。

 

And then don't forget about giving this a try

 

https://sourceforge.net/projects/foobar2000-wasap2-output/files/

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An alternative to the official wasapi and Asio foobar2000 output component:
WASAPI output Support 3.2.3
ASIO output Support 2.1.2
with a special effort to improve audio quality

 

Obviously foobar2000 already sounded terrific despite the fact that PGGB-RT wasn't even involved at all, let's just imagine what we could achieve with both of them. Sometimes we don't have to write something off just because it's completely free IMHO.

 

BTW, we could find quite a few interesting foobar2000 components here

 

https://www.foobar2000.org/components/author/Peter

 

Here's a quote from Rajiv

 

https://audiophilestyle.com/forums/topic/62699-a-toast-to-pggb-a-heady-brew-of-math-and-magic/page/11/#comment-1130123  

On 5/6/2021 at 4:15 AM, austinpop said:

Listening Impressions

I so very much wanted the wavpack files to sound the same as wav! But alas, I do hear a difference. Is it huge? No, and it still preserves most of the goodness PGGB brings. But if ultimate SQ is what you're going for, stick with WAV. If, on the other hand, storage is holding you back, then wavpack (and the already supported FLAC) could be a very good option.

 

For those of us who already converted lots of tracks into that WavPack container with PGGB Offline, both "Command-Line Decoder Wrapper" and "FFmpeg Decoder Wrapper" seemed to have some potential in terms of decoding to temporary files

 

https://wiki.hydrogenaudio.org/index.php?title=Foobar2000:Components_0.9/foo_input_exe

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When decoding to temporary files (%d): Because the whole file is decoded in advance, this component stalls for a while before playback of each file can begin. To avoid playback interruptions when changing songs, you might want to increase your playback buffer length (in Preferences / Playback / Output).

 

Quote

For any format supported by FFmpeg, decode whole file first (slower): ffmpeg -i %s %d

 

Yet another one that's also useful for playlists, maybe we could add plenty of RAM and disconnect the network connection to NAS then?

 

https://www.foobar2000.org/components/view/foo_ramdisk

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Creates temporary copies of a group of tracks - for an example, your playlist - in application's memory. Temporary copies exist only during the application lifetime.

Usage: right-click a group of tracks, use "Send to RAM-Disk" or "Add to RAM-Disk" commands from the "Utilities" sub-menu.

 

So there we have it, Win10XPE is free. ISO images of Windows 11 also don't cost anything at all. Same deal with foobar2000, while the trial mode of PGGB-RT could go up to 2 million taps.

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FYI - WavPack support was added to MPD since December 2008 (Christmas gift?)

 

https://raw.githubusercontent.com/MusicPlayerDaemon/MPD/master/NEWS

ver 0.14 (2008/12/25)
* decoders:
  - ffmpeg: new plugin
  - wavpack: new plugin

 


 

BTW, here's the feedback from another user of PGGB and I'm also enjoying HiBy RS6 (R2R DAC with NOS mode) as my portable rig

 

https://www.head-fi.org/threads/hiby-rs6-next-generation-hi-res-android-music-player.958878/page-85#post-16659926

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RS6 supports 2xDXD, which is crazy! I did not try 32bit DXD, and all files were 24 bit, but if you really have a favourite album and you want to hear what it is capable of, try upsampling it and feeding it to your RS6, you never know what you might hear!

I also tried DSD256 vs 2xDXD, I like 2xDXD more, treble was much better presented in DXD file, compared to DSD256 which sounded muffled.

It is fair to say this is the first time I've heard 2xDXD on a portable setup, colour me impressed!

 

Not exactly related to PGGB but it's just really awesome to have so much fun with R2R on the road

 

https://www.head-fi.org/threads/hiby-rs6-next-generation-hi-res-android-music-player.958878/page-122#post-16758501

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Paired with a dynamic and neutral monitor like IEM like the Softears RSV it can shake some pants. It feels like a massive club PA speaker system, punching and growling in an all encompassing soundfield.

My RME ADI 2 can't match its massive wall of sound.

Embrace the boogie. It reminds me of my dad's old school marantz system with celestion Ditton 44 speakers that used to rattle our house windows.

 

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Totally fantastic news, way to go

 

https://www.head-fi.org/threads/chord-mojo-2-speculation-thread.885405/page-73#post-16786562

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Of course one cannot judge by taps alone. But if everything else is kept constant, doubling taps will result in a more lifelike sound. It will be more revealing too, which isn't always a good thing based on the other components in the chain and one's musical preferences.

Really fascinating to me was to hear the difference between 2 billion and 3 billion taps when using PGGB. You would think that 2 billion would be enough and there couldn't be further improvements beyond that, but that turned out to not be the case. It wasn't difficult to hear the improvements.

I believe the theory tells us that we need an infinite number of taps to perfectly reconstruct the signal. So as long as everything else is kept equal, increasing the number of taps will always take us in the direction of better reconstructing the signal.

 


 

https://www.head-fi.org/threads/chord-mojo-2-speculation-thread.885405/page-75#post-16786923

Quote

Mojo 2 is reviewed in the latest issue of Hi-Fi World, which isn’t officially available until January 31st, but my local shop had a copy. They list the price as £449.00.

 

Quote

I haven’t finished reading yet, but it’s described as outstanding

 

Quote

I’ve just got in from work. Here’s the summary at the bottom of the HFW review.

vSPmApK.jpg

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Assuming that all tests will be converted to DSD 64 by HQP, are you expecting each candidate to sound any different from the others?

  1. 705.6kHz @ 32bit
  2. 705.6kHz @ 64bit
  3. 1411.2kHz @ 32bit
  4. 1411.2kHz @ 64bit
  5. 2822.4kHz @ 32bit
  6. 2822.4kHz @ 64bit

Just trying to find out if the law of diminishing returns were still applicable in this case, storage devices do get more affordable down the road but we might also be interested in knowing where the sweet spot is IMHO.

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10 minutes ago, ASRMichael said:

The above then converts to HQP 64bit.

 

Please correct me if I were mistaken, did you actually mean that HQP was always configured to output DSD 64 @ 1bit for every single test?

 

In other words, HQP should be essentially downconverting each PCM track from a relatively high sampling rate to a (much) lower one @ DSD 64

 

https://aurender.zendesk.com/hc/en-us/articles/360026675554-Why-does-the-sample-rate-of-a-DSD-file-show-176-4kHz-or-352-8kHz-on-my-Aurender-

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Aurender uses DoP (DSD over PCM) to send DSD files to DoP-compatible digital-to-analog convertors. Since DoP uses a PCM data stream, a DSD64 file will display at 176.4kHz, and a DSD128 file will display 352.8kHz on the Aurender Conductor app readout. Note that the receiving DoP-compatible DAC still reads the DSD file natively, as the PCM stream is simply acting as a transport for the original data.

 

DSD tracks are supposed to be a single-bit format all the way from DSD 64 to DSD 2048 while 64 simply meant the multiples of 44,100

esRoKtP.jpeg

UffNH0G.jpg

 

BTW, I wasn't expecting anything close to a critical analysis of comparing stuff like 1411.2kHz versus 2822.4kHz or 32bit versus 64bit etc.

 

IMHO it's more like whether we're merely talking about subtle differences (i.e. maybe we'll just go as far as converting our favorite tracks) if we're going all the way up to the highest sampling rate available, or a reasonably significant delta that's actually worth all that effort (i.e. let's start redoing the entire music library from now on) to dive right in.

 

Many thanks for sharing the results of your experiments with everyone, this project should be meant to benefit those DACs with “NOS” options the most but now maybe we're finding new ways to achieve even greater gains for other options out there.

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4 hours ago, Gavin1977 said:

Chord DACs do 32 / 768 no problems in Windows, but the X26 doesn't have an ASIO driver set by the looks of it.

 

Bypassing Windows audio mixer

https://www.audialonline.com/articles/bypassing-windows-audio-mixer/

https://addictedtoaudio.co.nz/blogs/what-we-think/in-search-of-768khz

Quote

I used JRiver Media Center to send it to the Topping E30 DAC, and it duly displayed “768.0 PCM” on its front panel display (see the photo up top). That worked using the ASIO driver, the WASAPI driver and even Windows Kernel Streaming. It would not work using Direct Sound.

 

Assuming that you're download ASIO drivers from this particular page

 

http://www.gustard.cn/?page_id=8956

 

Even V4.86.0 of Thesycon drivers supported 768 as shown below

 

https://dl.khadas.com/products/tone2/driver/Khadas_Tone_Driver_V4.86.0_201118.zip

https://forum.khadas.com/t/tone-2-pro-768khz-pcm-option/13055/3

Quote

You need a player like said above, I tried it with Foobar in ASIO and WASAPI Exclusive mode, and both work (go directly to the Tone2 Pro and avoid the Windows mixer).
The Windows setting can be what you want, it doesn’t change.

i6HIz7R.png

705.6 also worked just fine

 

https://community.roonlabs.com/t/wasapi-in-windows-10-21h1-update/168248/2

zxcWVQl.png

 

Maybe get rid of any unused devices and see how it goes?

 

https://www.uwe-sieber.de/files/DeviceCleanup.zip

https://www.thewindowsclub.com/device-cleanup-tool-windows

 

More importantly, what audio output options (namely ASIO since that could actually reach either 705.6 or 768) did you actually see once you go to the settings of your software player? Maybe capture that particular screen with relevant settings so that we could take a look.

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https://www.dbpoweramp.com/Version-Changes-DMC.htm

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FLAC 1.4.1 - Encoding and decoding of 32-bit PCM is now possible, 1MHz sample rates

 

https://www.signalyst.eu

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FLAC update to support higher sampling rates and up to 32-bit word length.

 

https://hydrogenaud.io/index.php?msg=1011789

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This limit has been raised to 1,048,575Hz in April, which is the true limit of the format. The previous 655,350Hz was the 'streamable' limit of the format, for some reason the reference encoder (libFLAC) didn't accept anything with a higher samplerate than that.

 

https://github.com/xiph/flac/commit/24629435bb265d740bf5ff12b2e48a7fbc319977

    /* sample rate can be up to 1048575 Hz, and thus use 20 bits, so we do the multiply&divide by hand */
    FLAC__ASSERT(FLAC__MAX_SAMPLE_RATE <= 1048575);

 


 

Compression ratios of WavPack versus FLAC seemed to be pretty darn close, too

 

https://hydrogenaud.io/index.php?msg=1011846

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I've posted some results already for the free file - now including -5 and -8p from NetRanger's build.
 

969 129 022 Let The Good Times Roll - Carmen Gomes Inc - Free 768kHz.wav
649 644 648 Let The Good Times Roll - Carmen Gomes Inc - Free 768kHz.-5.flac
642 946 998 Let The Good Times Roll - Carmen Gomes Inc - Free 768kHz.-8p.flac
641 225 096 Let The Good Times Roll - Carmen Gomes Inc - Free 768kHz.wv-hhx6.=37min=.wv
640 902 644 Let The Good Times Roll - Carmen Gomes Inc - Free 768kHz.wv480-hhx6.=40min=.wv

 

 


 

The encoding side should be a piece of cake since FLAC could be updated to version 1.4.1 (or above) on our own, though we've gotta ask the developers of software players to add support for the latest versions of FLAC.

 

Many of us should be able to ditch WavPack at some point once everyone is getting on board with FLAC @ 16FS, while others could stick with WavPack @ 32FS if our (R2R) DACs were able to handle that.

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For TAS and / or XDMS, hopefully updating FLAC or libFLAC to version 1.4.X wouldn't be all that tricky so that could be approved by Emile.

 


 

When it comes to Roon Server, someone else already started a new thread last month

 

https://community.roonlabs.com/t/flac-1-4-2-support/219843

 

Please feel free to cast your vote by hitting that "Vote" button over there.

 


 

For those music servers from Antipodes Audio, Logitech Media Server (no updates since March 2022) still came with FLAC version 1.3.4

 

https://github.com/Logitech/slimserver/blob/public/8.4/Bin/x86_64-linux/flac

Quote

Fix #773 - Add flac 1.3.4 for macOS/Linux

 

*IF* a simple drop-in replacement were actually good to go, we might consider making a backup of this file first

 

opt/logitechmediaserver/Bin/x86_64-linux/flac

 

Then grab a copy of FLAC version 1.4.2 and let's take the Arch Linux as an example

 

https://mirrors.edge.kernel.org/archlinux/extra/os/x86_64/flac-1.4.2-1-x86_64.pkg.tar.zst

 

We'll extract both files listed below and then copy them to opt/logitechmediaserver/Bin/x86_64-linux accordingly

 

https://archlinux.org/packages/extra/x86_64/flac/files/

usr/bin/flac
usr/bin/metaflac

 

If that were no go, revert all changes and then maybe ask Mark what to do about updating FLAC for LMS since the official version didn't get any updates for quite a few months already.

 

Other than that, maybe open a new issue here

 

https://github.com/Logitech/slimserver/issues

 

Simply mention a previous issue linked below and then Michael Herger should know what to do next

 

https://github.com/Logitech/slimserver/issues/773

 


 

HQP / NAA already got that handled, same deal with the latest releases of foobar2000

 

https://www.foobar2000.org/changelog

Quote

libFLAC 1.4.1, made possible to decode 32 bits per sample FLAC file.

 


 

For converting WavPack to FLAC safely, foobar2000 could be one of those viable options

 

https://hydrogenaud.io/index.php/topic,119929.msg988566#msg988566

Quote

So, using foobar2000, I'd do the following steps (lots of overnight tasks here):

  1. foo_verifier, verify all files. (Re-encoding will produce "valid" files; if something is corrupted, I want to know, not to destroy evidence.
  2. foobar2000 to convert to identical directory structure
  3. foobar2000 with foo_bitcompare. (But first, check that you have the same number of files, same number of samples, etc.)
  4. Delete old. Again, using foobar2000
  5. Search old structure for WavPack files to make sure there are no-one left that weren't caught and indexed
  6. If you didn't copy "other" files, now it is time to merge.

 

foo_verifier
https://www.foobar2000.org/components/view/foo_verifier

 

foo_bitcompare
https://www.foobar2000.org/components/view/foo_bitcompare

 

Foobar2000:Converter
https://wiki.hydrogenaud.io/index.php?title=Foobar2000:Converter

 

Free Encoder Pack

https://www.foobar2000.org/encoderpack

flac - Command-line FLAC encoder/decoder version 1.4.2

 

How to convert WavPack Files to FLAC
https://hydrogenaud.io/index.php?topic=120174.0

 

Foobar2000 Conversion Setup
http://www.vgmpf.com/Wiki/index.php/Foobar2000_Conversion_Setup

 

How to convert FLAC files to MP3 using foobar2000
https://captainrookie.com/how-to-convert-flac-files-to-mp3-using-foobar2000/

 


 

For Mac and Linux users, the author of flacon didn't state if FLAC / libFLAC were updated to version 1.4.X or otherwise

 

https://flacon.github.io

Quote
  • Supported input formats: WAV, FLAC, APE, WavPack, True Audio (TTA)
  • Supported out formats: FLAC, WAV, WavPack, AAC, OGG or MP3

 

Latest releases

 

https://github.com/flacon/flacon/releases/latest

https://flathub.org/apps/details/com.github.Flacon

 

We might start a new issue on GitHub and then find out if those features from version 1.4.X of FLAC were actually included

 

https://github.com/flacon/flacon/issues

 


 

There's always XLD as follows

 

https://sourceforge.net/projects/xld/files/

 

However, only one file was updated in September 2022 and the others seemed to be pretty old

 

https://sourceforge.net/p/xld/code/HEAD/tree/trunk/XLDFlacOutput/

 


 

Alternatively, Linux users might try foobar2000 via WINE instead

 

https://snapcraft.io/foobar2000

 

As always filenames and tags in plain English should be no biggie once we've converted WavPack into FLAC. Things are gonna get more complicated whenever we're dealing with double-byte characters as well as accented characters that are involved in other languages.

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