To Be Determined

An Exploration Into Digital Audio File Compression

Issue 143



I first got into digital file streaming about eight years ago at the insistence of my brother, who wouldn’t shut up about it until I got somewhat involved. Of course, at that point in time, we called it “computer audio,” because, frankly, I don’t think any of us really knew what “streaming” was at all about. As I’ve mentioned here before, the late John Sunier offered me the opportunity about five years ago to review the Auralic Aries first-generation streamer, and mainly because he didn’t know what it was or what it did. I didn’t know either, so I also passed. If only we could climb into a time machine!

Just for clarity here, I’m not talking about compression as in, “the remastering of this title is totally too compressed!” As in, the dynamic range (and any of the life in the music) has been totally sucked out of it. I’m talking about file conversion algorithms that employ a selectable level of compression (or not!), and if (or how much) your files may or may not have been compressed at the point when you ripped your CDs to FLAC or whatever. Most file conversion applications employ some level of compression mainly to save file space on your music server. Remember, we’re only talking about PCM files here!

Ripping with Winamp and Exact Audio Copy

I currently work on Windows and Linux-based computers at home, and on Windows and Macs at my day job, but at the time when I first became involved in “computer audio,” it was strictly Windows – as in, Windows 7. Meaning, very simplistic. I had no concept of “ripping programs,” and started out with Winamp, which is probably less horrible than you might think, and especially at the time. It wasn’t overly complicated, and I ripped everything to FLAC – mainly because my brother told me to; I didn’t really know anything else. In my early ripping experiences, I still mostly thought that all digital sources (including the compact disc and other digital discs), weren’t particularly great-sounding, at least in their then-present incarnations (with the exception of SACD, of course). I considered digital inferior to analog sources, so it probably wasn’t worthy of “overthinking” in terms of preparation for audio playback. It would take a miracle to get the CD format sounding any better than it did, so how could a rip of a CD be expected to sound much better?

I have no idea if Winamp offered any level of selectable compression of the output file; I wasn’t savvy enough at the time for anything like that to be on my radar. And besides, the Windows 7 experience was coming to an end for me – my brother’s two sons, who were computer science students at Georgia Tech, had come home with a copy of a new bootleg program that would allow you to load any version of Windows then currently available without a license. Actually, it had a “loader” application that would generate the correct product key for whatever version you chose to install. Yeah, I know, I’ll plead the fifth on this one. I ended up installing Windows 10 Pro, skipping Windows 8 completely in the process (my current setup is completely legit; yes, Bill Gates thanks me for my support). And at that point, I moved up to Exact Audio Copy (EAC), where I actually started paying some attention to things like ripping an “exact copy” of the CD to FLAC. Compression was still off my radar at this point, however.



In my initial experiences with building a digital library, I 1) ripped CDs from my own personal library that I listened to with some frequency; 2) borrowed from my brother’s (and friends’) libraries for CDs that I didn’t happen to own but wanted; and 3) borrowed from the public libraries around me for CDs that were uncommon in my usual circles or travels. Yes, I know all about the “copyright issues,” moral implications, fair use, etc….once again, I’ll plead the fifth. I wasn’t reselling anything, it was all strictly for my own personal use, so at the time, I didn’t have any issues with it. My habits have changed significantly since then.

Moving to dBpoweramp

So, basically, what I’m getting at is that I probably had hundreds upon hundreds of CD titles that had been ripped with either Winamp or EAC, and with little concern for overall file quality and, especially, any level of compression that might have been employed. The EAC era lasted a couple of years, but when I moved into a new house four years ago and suddenly found myself with a significantly higher level of digital equipment to play with, I took the next step and bought licenses for dBpoweramp. Something else happened at this point: hard drive storage (especially solid-state drive) prices began to drop significantly, and I started reading online about “no longer needing to compress FLAC files when ripping, because with the now really low prices of storage, who cares about saving a bit of space?” So, listeners could now rip all their CDs, using dBpoweramp, as uncompressed FLACs and without a care concerning the file size. No fuss, no muss, no need to budget drive space!


At this point, I probably had about 500 ripped files in total. When I started using dBpoweramp, I suddenly decided to re-rip all my previously-ripped personal library files using the “uncompressed” settings to replace the previously compressed rips. So, over the four-year period that I’ve been living in the new house, I’ve ripped about 3,000 CDs and 400-plus SACDs and DVDs. I’ve also explored some of the dBpoweramp plug-ins – for example, the HDCD plug-in – where you can rip all your library HDCDs as 24-bit versus 16-bit versions. Now that’s a really useful algorithm, and especially in my current “disc-less” setup, which allows me to experience all my HDCDs without a disc player.

But that still leaves several hundred of my “original rips” that came from CDs, libraries, or friends, that I no longer have access to. When we moved, we changed counties, and the current county I live in doesn’t even have any music available for lending at any library locations. What’s this world coming to? I have often thought over recent years that the sound quality of some of those early rips (especially from Winamp) is probably somewhat substandard compared to rips made with dBpoweramp.

Commonly-Available File Formats

In my gigs as an audio and music reviewer, I often have access to digital files of varying formats. Usually, when I get a review assignment that involves a CD release or reissue, advance files are offered. They usually are available as downloads as WAV or higher-resolution MP3 files (320 kbps). One of the available plug-ins with dBpoweramp is a file format converter that works to absolute perfection converting WAVs to uncompressed FLACs. And I always convert WAVs to FLACs; otherwise, you have no access to any of the files’ metadata. I also have the same process with LP review releases, where digital files are also planned for release on the various streaming services. And I happen to believe that with highest-resolution (320 kbps) MP3s, there’s very little going on in terms of file content getting trimmed from the final files; it’s just getting really compressed. When I convert 320 kbps MP3s to uncompressed FLACs, I’d absolutely dare anyone to be able to double-blind tell the difference between the converted files and the actual CD files.

There was a fairly popular thread on many of the digital audio sites a few years ago where the author claimed that if you ripped your CD to 320 kbps MP3 using the LAME codec, then reconverted it to uncompressed CD quality, you would experience a significant uptick in overall sound quality. A couple of writers at Stereophile even chimed in on this. I never dove too deeply into it, but did experiment a bit, and could find no difference between the original and the LAME-processed, then reconverted MP3.

So what is all this rambling getting to? I’ve had frequent conversations with Dalibor Kasac, who’s one of the principals at Euphony Audio (manufacturer of my streaming setup). And he believes that, whenever possible, zero compression should be used. Regardless of how great the conversion algorithms are that are currently in use with most streaming software applications, there will always be losses when compression is applied. (He also believes – unflinchingly so – that DoP [DSD over PCM] is terrible; only native DSD will give you the entire picture.) I have to agree with him here; since I’ve switched to an I²S playback setup, I honestly do believe that DSD (and perhaps even PCM) sounds remarkably better when played without conversion through my PS Audio GainCell DAC. YMMV, but I stand by what my ears tell me.

An Epiphany Last Night

Like I said, I’m frequently converting WAV or MP3 files to uncompressed FLAC for playback evaluation – especially when I’m reviewing an LP of material that I’m totally unfamiliar with. I’ll convert the files for playback on my home digital system, or perhaps in my car; my process is that I can just listen to the files for an extended period of time, just to get a feel for the music, then evaluate the actual LP. I usually get LPs with a protracted period of time before the actual release date, so I generally have lots of time to play about with them before writing a review. And, surprisingly, the digital files usually sound pretty darn great over my system – in fact, I’m usually quite stunned by how very good they sound!

So, last night, I was converting some files for an upcoming LP reissue, and I suddenly had this thought – using dBpoweramp’s file converter works so effortlessly and with such good results, why not use it to convert the oldest of the old FLACs in my library to uncompressed FLACs? I didn’t have any idea what level of compression (if any) was employed by Winamp or EAC, so I started looking at files in my library – I actually opened Roon, and started looking at the album covers to help remind me of the older files in my system. It amazed me how very many I recognized as coming from sources that weren’t in my CD (or other disc) library, and here’s the real kicker: when I clicked on the FLAC files on the storage drive in Windows Explorer, and checked the Audio Properties tab, many of the earliest files were compressed as much as 35 to 55 percent!!



I currently have tons of solid-state drive (SSD) storage space available, so I made a copy of the first album I came across, Kraftwerk’s Minimum-Maximum, which is a live album I got from the Chicago Public Library. It showed the Winamp compression level as 35 percent, so I used dBpoweramp’s converter to change the compression level to zero percent. I then went downstairs for a listen; nary a tick or hiccup, and no artifacts of any kind, and the sound to me was definitely better than before, so I then went on an hours-long binge, converting everything from waaaay back that I could easily identify. Yeah, I know, you probably think I’ve totally lost it here, but it’s at the very least been really liberating to get all these early files on pretty much equal footing with more recent rips.

More Previously Unidentified Compression

After converting all the early Winamp and EAC rips, I had another thought: I have at least a hundred or more digital downloads in my system that have either been promos, gifted, or that I purchased. Could they possibly have any level of compression? A reminder – we’re only talking about PCM files here; DSD files do not have any level of compression applied.



I started looking at all my 24/96 and 24/192 files I’ve downloaded from HDtracks and Acoustic Sounds; sure enough, all of them had compression levels that averaged between 30 to 50 percent. I had downloaded the recent Beatles Abbey Road 24/96 Giles Martin remix, and it was compressed at about 40 percent. Early on, I’d downloaded Joni Mitchell’s Ladies of the Canyon, in 24/192 resolution. I always found this download very puzzling, as I expected such a high-level-resolution file to be also quite a large file. It wasn’t; it was actually smaller than many of my 16-bit files. Imagine my surprise when I checked its compression level to discover that it was a monstrous 81 percent – that explained everything! Of course, after conversion the file size literally quadrupled, but who cares? Again, I experimented, and finding no trace of any artifacts, converted all of these downloads to zero compression levels. But I have to believe that compressing a file by 81 percent – which would definitely make it much easier to download – would have to negatively impact the sound quality. This was true time and time again with every HDtracks download compression level I checked – with Rush’s classic 24/96 download of Moving Pictures, there was a whopping 68 percent compression. Of all the digital download tracks I checked in my library, those from HD Tracks consistently had the highest levels of compresssion.



Then there were the Qobuz downloads; they all had an average of 30-plus percent compression. 24/96 and 24/192 downloads from the 2L and Channel Classics labels had compression levels averaging between 40 to 50 percent. One of my favorites, Channel Classics’ Guardian Angel by Baroque violinist extraordinaire Rachel Podger was compressed for delivery by 52 percent. Again, this greatly increased the ease and speed of downloading, but seriously, 52 percent compression from an audiophile label? I’ve downloaded a number of titles from High Definition Tape Transfers (HDTT), which is one of my favorite download sites for outstanding archival transcriptions. One of my favorite titles, the Spanish music classic Espana featuring conductor Ataulfo Argenta, was 49 percent compressed. Amazing, and incredibly disappointing.



Sitting around, racking my brain to confirm that I hadn’t forgotten anything, I suddenly realized that I had the Steven Wilson Remix Blu-ray and DVD-Audio discs on hand that I’d bought of Yes and King Crimson titles that I’d recently converted to FLAC. I also realized that in the conversion setup menus, I didn’t recall seeing anything that allowed for adjusting the level of compression during the conversions. I found this really interesting: when converting from Blu-ray discs, zero compression was applied, but when converting from DVD-Audio discs, usually somewhere around 50 percent compression was applied. Again, I test-listened to everything, and felt the results were definitely an improvement.


Yeah, I know – I’m waiting for the flamethrowers to come out the day this article drops. All music player applications are designed to decompress a compressed file in order to achieve correct playback, and the transfer process is supposed to be completely seamless – but we’re talking about some serious digital processing going on to achieve that. There’s some level of compression in most digital file formats, and the transfer process between compressed file and uncompressed playback isn’t supposed to have any impact on sound quality. But I can’t believe that the transfer process involved in decompressing a file that’s been heavily compressed (like 80 percent!?!) will ultimately result in absolute fidelity. As I mentioned earlier, with the current low cost of storage – why not just maintain an entirely uncompressed digital library?

Header image from the dBpoweramp website. All other images courtesy of the author and the respective entities.

10 comments on “An Exploration Into Digital Audio File Compression”

  1. Despite a handful of naysayers, FLAC has been “proven” time and again to be “lossless” in the sense that one can perform endless conversions from FLAC > WAV > FLAC > WAV and still recover the identical PCM music (some metadata is inevitably lost due to the limitations of WAV). It is identical through any comparison one might wish to make; AudioDiffMaker to compare the audio data or various computer file comparison algorithms for the bit-to-bit data. So ultimately, as long as the file doesn’t go through a lossy conversion step one can always get the original PCM information back. No matter which format one chooses from HDTracks, QoBuz, Prostudiomasters, etc, the information is always sent as compressed FLAC; either the downloader program converts to the purchased format, or the purchaser can convert him/herself after download. Regardless, the PCM information will be the same.

    All that doesn’t address how the file actually plays; PCM (WAV), PCM (AIFF), FLAC (uncompressed) and FLAC/ALC (compressed). Searching the Internet will find proponents of any of the above sounding best (on their individual system, of course, but the posters will almost always try to generalize their opinions to anyone’s system).

    Personally, in my system (which also has a Euphony Summus connected via fiber to my MSB Discrete DAC, and also a modified Oppo 205 connected by RCA to the same MSB DAC; all LPS power, no SMPS in sight) I have not been able to discern a consistent preference for (or even a difference between) any of the above, but I have not yet performed *exhaustive* comparisons. Still, to date I have spent a few hours carefully comparing a number of albums. I wouldn’t attempt to say that my findings are generally applicable, though.

  2. My recommendation and long story short is, not to rip everything to uncompressed Flac, but to AIF.

    It’s also uncompressed, it also takes meta data and it sounds clearly better than uncompressed Flac and only a tiny bit weaker in bass than wav. Can’t explain why but that was what I heard when initially comparing before I decided for a format (fortunately roughly confirmed by Paul at the time when I began streaming from his tests)

    I would be interested in an expert opinion on the native DSD/DOP topic, as I think there’s a lot more (and different, or at least more differentiated) to say, but that needs someone like Ted 😉

  3. i have no experience with windows, having used macs since the late 1980s. so my findings are only based on my experience with macOS – and to be taken with a huuuge grain of salt.

    my favorite file format for music is ALAC (apple’s version of FLAC). and my favorite software to play the files in my iTunes library is channel D’s PureVinyl/PureMusic which i’ve been using for at least a decade now. as iTunes can’t handle FLAC, i use PureMusic’s built in FLAC > ALAC converter.

    as long as i use PureMusic as-is, it makes no audible difference whether i use PV or iT – simply because they both decode the losslessly compressed data in real time. so far so good (or not). so why do i use PV at all? because i use PV’s “memory play” function: it doesn’t play right away, but first decodes the data and loads it into RAM. and only then starts the playback. and that makes a huuuge difference. all the decoding (uncompressing, actually putting a lot of warm air back into the music by inflating the losslessly compressed data with … zeroes where zeroes were to begin with.

    i’ve asked many CPU-savvy nerds why it makes a difference whether the data is inflated in real time with zeroes or before the playback starts from RAM. it’s not the actual computing (which isn’t terribly demanding), but the load on the power supply and the perturbations it creates. audio playback is a real time process so ideally the computer wouldn’t do anything else – absolutely nothing else! – besides converting the data to information an external DAC can convert into analog signals again. memory play was an even bigger asset back when we still had to use rotating hard disks instead of SSD or PCIe flash memory. but even today, with insanely large storage volume (as in tens of TB) being available in SSD format, it still makes a difference whether i play a file directly from the SSD or via the 96 GB of RAM currently installed in my mac pro.

    on his website, rob robinson (the genius behind PureMusic/PureVinyl) explains why it takes him so long to convert his two 32–bit apps into 64 bit. and surprisingly, the time consuming part is not the code, but to make the apps as lean and mean as possible in terms of system load. this may be one of the reasons, why on the fly-decompressing makes FLAC/ALAC sound different than WAV/AIF – and why they do no longer sound different if played back from RAM. there is a lenghty piece of information on that i took the liberty to copy there and paste here.

    here it goes:

    “Most notably, critical sections of UI code have been rewritten from scratch with an eye towards reducing the CPU footprint on both the M1 and on older, Intel processor computers. Of course, reducing and levelling the CPU footprint has always been a key design philosophy, since it correlates directly with playback quality. “Bits are bits,” yes, except for real-time processes like audio playback which involve another factor: time. Namely, the orderly and carefully regulated delivery of the audio stream to the DAC.*

    CPU activity directly affects computer power demand. This can be easily verified with a consumer watt meter such as the “Kill-A-Watt.” Unless computer power demand remains low and as steady as possible, fluctuations in CPU / computer power demand can induce noise on the electrical utility line. This noise affects the stability of the sample clock low/high transition comparator, causing uncertainty (clock jitter) in transition timing, causing “haze” and a loss of definition in musical presentation.

    Reducing CPU footprint requires painstaking and time-consuming code optimization; it can take many days (and nights, literally “dreaming in code”) of testing, optimization, and re-testing even for seemingly small, but critical, sections of code. Then, improvements require extensive testing inside the whole software ensemble to insure there are no unanticipated side effects. Rather than performing a quick “port” to the new OS, we’ve taken the opportunity to reconstruct the underlying foundation. There are dramatic improvements coming in our updates, unlike products which seem to have other priorities than CPU footprint minimization, even requiring purchasing a new computer to support their bloated CPU requirements. This illustrates a critical and important difference in design philosophies. The CPU requirements of our coming updates will actually be decreased – not increased; by doing more with less. […]

    *Note: as used here, “DAC” refers to the circuit-board and chip level final destination of the digital audio stream before conversion to analog. That is, the internal sub-components comprised of the sample clock managing the transfer of data plus the delta-sigma converter chip (or resistor array of a ladder DAC) which make up the heart of a digital-to-analog audio “DAC” component connected to your computer / audio system. This also includes all-in-one, turn-key audio servers, which are subject to the same kinds of issues. The troublesome noise is imposed on the circuit-board-level power supply and ground reference for the sample clock / DAC chip, and can be common-mode (imposed on power supply and ground reference together) or differential (imposed on one or the other); not including noise in connections between the computer and the DAC component or analog signals, though these also can contribute jitter-inducing noise.”

    so this very long post may at least in part explain why different file formats of uncompressed and losslessly compressed music may sound different in playback. it’s not the data that’s different – it’s the load on the CPU and whether the data is decompressed in RAM or on a rotating HDD.

    to my ears, a CD ripped with iTunes or with a separate bit-perfect ripping app as EAD (on mac) sounds the same provided iTunes is set to read a section repeatedly if it detects errors. i haven’t yet compared (= null test) one and the same piece of music ripped with iTunes and EAD. but i’m pretty sure, what ever difference a null test will come up with has more to do with having shifted the two versions by one sample or two than with the data being really different.

    and controlling your DAC with a *very* good word clock makes more difference to my ears than being too anal with ripping and file formats. jitter you can hear. and jitter is induced thru system load and how your external DAC (or streamer) copes with a slightly fluctuating data stream from the computer.

    (thanks for your patience!)

  4. Perhaps, someone on this list, can help me get some peace out of digital confusion here.
    I use Wave Pad to record from Appkle Music via my iPad connected to the line level input of my sound card, via a cable from my iPad’s headphone jack.
    I can get up as high as 392KHZ in Wave Pad in to PCM 32bites.
    But also, I use Audacity to add equalization and other sound afects.
    But the wave files come out as PCM, 64bit float in to 392KHZ.
    I guess it’s the way my ears hear things.
    But those 64bit floating files sound a tad bit better then 32bit files.
    Can anyone explain this to me?
    Thinks in advance!

    1. if i understand you correctly, you are using a line input, i.e. you feed an analog signal to your computer’s/soundcard’s A/D converter. i’m not familiar with windows, but on mac, the highest sampling rate of the ADC is 192 kHz. depending on the format, word length is either 24 bit or 32 bit float. internally, the calculations are handled with 64 bit accuracy. in terms of resolution, 24 bit fix and 32 bit float are pretty much the same. 32 float is not common as data format, but is convenient as interchange between recording studios because it preserves more headroom.

      so what you are doing is basically upsampling the 192khz (max) PCM data from your computer’s/soundcard’s ADC to 392 kHz. considering that even today’s advanced 24 bit A/D converters don’t reach the full 24 bits for multiple reasons (the laws of physics being one of them…), you don’t gain more information by handling them with 32 bit float.

      but the main issue is the fact that the Apple Music data is 16 bits to start with. and you are doing an unneccessary D/A > A/D conversion which very likely will *not* improve the integrity of your data. by re-digitizing the formerly 16 bit information with 24 bit, you end up with more data, but not more information… (it is, however, a good idea to A/D with 24 bit just because of the added headroom).

      your iPad should put out digital data over its Lightning connection. i don’t know what digital input connectors your computer or soundcard offers (USB, TOSlink, S/PDIF, 3.5mm jack) but i’m sure there is an adapter from Lightning to whatever. this would keep the signal in the digital domain.

      to answer your question: since the signal has/had only 16 bits of information, you can’t hear more by listening to them with 24 fix or 32 float. and i don’t think audacity (or any other software) will put out 64 bit data – quite simply because there is no D>A converter hat can cope with 64 bit signals…

      any audible differences are caused by either wave pad or audacity ADDING something to the music. (btw. why not record directly to audacity? or use the Audio Unit (AU) plugins that every mac has in its OS? you can use them with both, wave pad and audacity. that way you don’t have to use two apps to do the same.)

      1. FYI: while all files (aside from digital downloads from music sites) currently in my digital music library were, in fact, ripped with dBpoweramp on a Windows-based computer, that Windows computer isn’t involved in the playback process at all. My music server, file library, and streamer (all from Euphony Audio) are network-attached, and any music files streaming to my digital-to-analog converter are all lossless uncompressed FLACs that are either 16-, 24-, or 32-bits (along with over 400 native DSD files). The Euphony Summus and Enpoint systems are highly modified Linux machines — there are no Apple or iDevices involved in the process. The Euphony Endpoint device has just been upgraded to include an internal I2S card, such that the digital files are streamed unconverted from the server directly to the I2S input of my PS Audio DAC. There are no analog cables or conversions involved.

        The rather rambling portion of your post regarding Macs, iDevices, Audacity, WAVpads, etc, does not in any way apply to my particular situation. Nothing in my library has been upsampled, only decompressed from as much as 80% compressed to ZERO compression.

        My point, is that during playback, there’s a transfer function happening whenever a compressed file has to be decompressed prior to playback. With an entire library of uncompressed files, no transfer function takes place that can possibly degrade playback.

        Thanks for reading,


      2. Good evening Siiarr!
        There are two reasons why I don’t use Audacity to do everything.
        #1. that is a chore and a half for me a blind man that has to rely on screen reading text to speech to perform.
        Sure, you can make all the recordings you want.
        But the hard part with Audacity, is the editing part.
        There are a lot of things inside of Audacity that JAWS just won’t read out loud to me.
        #2. It’s easier to use Wave Pad to record and edit recordings for me.
        I just import them in to Audacity to give them the kind of sonic treatment I want the files to have.
        But when I get threw with them, they are exported out in to 64 bit floating PCM wave files.
        Sometimes, all depending on what kind of a mood I happen to be in, I convert them to DSD.
        And I’m doing all of this on a HP Pro3500 series desktop computer.
        And it’s running Windows10Pro build 2004.
        But sense you asked about my input and output jacks, I only have line out, line in, and mic in.
        Sure if I had a DAC that uses a USB input, then I could get digital audio out of this computer.
        But sense I don’t, then the only way I can connect my iPad to this computer, is via my iPad’s headphone jack in to my line in jack on my integrated motherboard sound card, until I upgrade to either a better computer and or, a better sound card.
        But for right now, I’m making do with what I have until then.
        But as for today’s Mack systems, I don’t know the ins and outs about them as of yet.

        1. john, apple makes a “Lightning to USB 3 Camera Adapter” that gets digital audio out of your iPad via USB to your computer. buy the original apple version, it costs $39.00 and it’s guaranteed to work. unlike the cheap knock-offs on ebay…

          two caveats:
          #1 it’s important to get the newer USB3 adapter, not the older lightning to USB2 because:
          #2 only the USB3 adapter has two connectors–a USB A and a lightning. connect the USB to any USB port on your PC and the lightning to your iPad charger. the bus power on the USB ports of your computer doesn’t deliver enough current to power the camera adapter, failing to supply additional power will result in the iPad delivering an error message…

          i’m not familiar with audio drivers on windows. but i’m sure there is one available that will accept the USB audio from an iPad

          (in case you have an older iPad with the 30-pin dock connector: apple makes a 30-pin to USB adapter. as far as i know, the USB will also supply data.)

          you could also try to connect your iPad to the PC via Bluetooth. if you succeed, you will have the digital audio of your iPad on your PC.

          re: wave pad / audacity. i don’t know what kind of “sonic treatment” you apply to your files. you can do exactely the same in wave pad – it has plenty of sound effects built in. and there is a button to use any VST plugin. i agree with you that audacity’s user interface is a nightmare. way too many functions and it saves files by default in a proprietary format that no other app can open. (sarcasm on) but hey, it’s free, so be glad you get so many functions for nothing… (sarcasm off)

          but i repeat what i wrote in my earlier post: you don’t “improve” the music of a digital audio file by saving it at a higher sampling or bit rate. all this does, it makes the file bigger…

      3. Siiarr,

        Okay, my apologies — virtually without fail, when a comment is posted on one of my articles, I’m notified via email. That didn’t happen with John Price’s comment that was posted on August 29 — the notification system skipped him. When I pulled up the article after being notified of your new comment, it went directly to your comment in the listing, and I didn’t see that you were in fact responding to a comment made by John Price.

        I found your remarks very confusing, but after reading through the latest comment from John Price (which I got a notification for), I scrolled back up and saw everything and figured it out.

        Sorry about that!


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