MP3 and other HiRes formats

1. Introduction

Hey babes... today I´ll be talking about high resolution formats / codecs. I will show that even lossy codecs like MP3, AAC or WMA are perfectly able of encoding music in high resolution with spectacular results. I will show you listening tests, graphs and measurement results. I´ll also talk about the necessary requirements so that all of these codecs perform at their very best. I´ll tell you how these codec really sound (minus the usual bullshit). And finally, I´ll give advice how to unleash the best possible quality while using lossy codecs. I won´t talk about FLAC, APE, TAK, ALAC or WavPack... those are all lossless, a.k.a. encode music without taking something away. This article is divided into seperate paragraphs which I´ll mention now so that you might jump to the one interesting you the most:

1. Introduction
2. A bit of lossy history
3. How lossy codecs work
4. Transparency
5. Fairytales - or why MP3 & Co. are HiRes capable
6. Seeing is believing
7. Listening to differences
8. And now... the problem
9. The solution
10. How to make the best MP3s (& Co.) ever!
11. This really is high resolution? Some disadvantages
12. REAL "lossyless" high resolution
13. The sound of noise
14. Transcoding horror
15. Conclusion

2. A bit of lossy history

Fig. I: Responsible for all the 'mess': Karlheinz Brandenburg (copyright: Wikipedia)

The guy above did it all: Karlheinz Brandenburg. He was part of a group of scientists at the University of Erlangen-Nuremberg and in 1989 he described in his dissertation several techniques necessary for lossy codecs. The principles he wrote about are the foundations for any lossy codec and they have been in use ever since. Herr Brandenburg is called by many people the 'father' of MP3. In fact, after his dissertation he further developed this codec in cooperation with other scientists at the Fraunhofer Society. Since then, MP3 has become the most dominant codec to store music with. Around the same time (1992) Sony unleashed ATRAC necessary for the MiniDisc, by that catapulting lossy coding into mainstream consciousness. In 1999, Microsoft released WMA in order to have its own MP3-alternative which then could be licensed to partners for loads of money (or so they wished). In the same year, the Fraunhofer Society standardized what was to be the successor to MP3: AAC or MP4 and just like WMA and MP3 the use of this codec costs money. The Xiph.Org foundation followed suit in 2000 with a completely open and free alternative called Vorbis, commonly referred to as OGG. But as I said above, MP3 still is the most used lossy codec even though it´s old and technically inferior to the other codecs (ATRAC is worse).

3. How lossy codecs work

To make it short: they make music smaller while trying to keep sound quality on par with the original they were encoded from. They remove what our ears cannot hear... well, that´s not very precise. They remove parts of the music our ears AND our brain are unable to perceive. 'Perceive' is important as lossy codecs remove information our brain would ignore anyway. Because of this ear/brain combination those codecs are called 'psychoacoustic'; us humans never listen with our ears only, our brain is indeed the biggest part of our hearing. I won´t go into detail describing how lossy codecs are able to shrink filesizes... but have you ever thought about the description 'they remove parts of the music'? In fact, they don´t literally remove those parts; what they do is dynamically decreasing bit depth for certain parts, frequencies or information they deem to be inaudible. For example: if a louder part masks a softer part, lossy codecs decrease bit depth for the soft part to, say, 1 bit (see Fig. II). When you decrease bit depth you create a noisy residue, called quantization noise. The ability of a codec to hide this noise partly determines how transparent it sounds to us.

Fig. II: audio masking and subsequent bit depth decreasing (copyright: Wikipedia)

4. Transparency

If MP3 & Co. are only removing what we cannot perceive anyway, why do we sometimes hear more or less horrible compression artifacts? Well, the sound of a lossy codec basically depends on how efficiently it encodes, on the available bitrate (measured in kBit/s), encoding speed (fast isn´t always best) and if the codec is maintained well (a.k.a. continuously developed). MP3 for example has been in constant development for 20 years, first and foremost in its LAME variant. It is now so good that it reaches transparency for many people with a bitrate as low as 128 kBit/s. 'Transparency' means that the lossy encoding cannot be distinguished from the lossless original it was derived from. In other words, for many people a 128 kBit/s MP3 sounds the same as the original, 11 times bigger 1.411 kBit/s (bitrate of the CD) lossless source. In my opinion that fact not only proves how un-trained the ears (and brains) of casual listeners are, it also serves to point out how good lossy codecs have become. Wouldn´t it be transparent, you´d hear artifacts like flanging, pre-echo, smeared transients, problems with virtual stage, distortions (quantization noise), etc. These artifacts are a problem... for decades now MP3 has "enjoyed" a reputation as bad as the always-bad-stepmother in fairytales. Audiophiles and casual listeners alike constantly claim that MP3 (or any other lossy codec of their choice) sounds cold, lifeless, digital and... yes, compressed; lossy codecs are too often accused of "dumbing down" the sound.

5. Fairytales - or why MP3 & Co. are HiRes-capable

You see, all of this is was the truth. 15-20 years ago. In 2013 it´s indeed like the bad stepmother; a fairytale. Most lossy codecs are far, far better than their reputation. To you this may be surprising, dear constant reader, but they are indeed able to encode high resolution material like 24/44.1, 24/48 or even 24/96.
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Don´t believe it? Here´s why... all lossy codecs share a common trait: they encode any audio material not with static integer but with floating point values. You may remember that I talked for hours and hours about the ability of the MiniDisc to encode audio signals with a quality surpassing that of CDs. It´s exactly the same with MP3 & Co.: since they all employ floating point precision, the bit depth fed to the encoder's input is irrelevant, they will encode anything, whether it´s 16 bit, 24 bit or 32 bit.

What´s the difference between integer and floating point? Integer handles values like this: 23, 45, 156, 001, etc. Floating point is like this: 7,89654367, 674,342167, 55,236548955214587, etc.... you get the idea. It is able to work with higher precision since it allows for much more possible values than integer (or fixed point). The results are a gigantic dynamic range and a spectacular signal-to-noise ratio. For 24 bit integer you have a (quantization) noisefloor as low as -144 dB, for 32 bit floating point it´s at a stunning -202 dB.

6. Seeing is believing

The first graph below presents a tiny sine at 1.000 Hz with a level of -90 dB on an original 24/48 wave file (-> lossless, see Fig. III). This is so low-level that you won´t ever hear it. But you can see that the sine still looks like a perfect sine. 24 bit obviously allows for enough possible values for any signal at -90 dB to be properly represented, no wonder considering a signal-to-noise ratio of -144 dB. The next graph (Fig. IV) shows the same signal, though this time encoded with 16 bit. The situation now changes: this low-level signal gets very close to the -96 dB noisefloor limit of a common 16 bit system. In place of a sine you now have the very famous 'digital staircases' signal so often used to (mis-)represent supposed flaws of any digital system.

Fig. III: 1.000 Hz sine, -90 dB, 24/48 WAVE

Fig. IV: 1.000 Hz sine, -90 dB, 16/48 WAVE

Now let´s have a look at several codecs. Are they able to encode with high resolution so that the sine will look like Fig. III? Please be aware that I used only the latest codec versions; I also employed their highest possible bitrate. In case of MP3 (LAME) that´s 320 kBit/s, for AAC (Apple's implementation) it´s 320 kBit/s as well and for WMA Professional it´s 440 kBit/s. 

Fig. V: MP3 320 kBit/s, 1.000 Hz sine, -90 dB, 32/48

Fig. VI: AAC 320 kBit/s, 1.000 Hz sine, -90 dB, 32/48

Fig. VII: WMA Professional 440 kBit/s, 1.000 Hz sine, -90 dB, 24/48

Duh! The three graphs (Fig. V to VII) prove that we have true 24 bit resolution with every lossy codec represented here. Not one of the sines looks like Fig. IV. As I said: high resolution - despite lossy compression by a factor of 7. Wanna hear a word from the inventor of MP3, Herr Brandenburg (the guy up above), regarding this?

There are some kinds of deficiencies of standard audio equipment which cannot be found in properly designed Layer-3 and AAC codecs. They are listed here to mention the fact that it does not make sense to test for them. Most noticable are Dynamic range: MP3 and AAC both contain a global gain adjustment parameter for every block of music data. According to the word length and resolution of this parameter, the dynamic range of both MP3 and AAC is well beyond the equivalent of a 24 bit D/A resolution. In short, MP3 and AAC represent the music in a way that the dynamic range of every known audio source is perfectly retained. (Source)

High resolution again, he´s well aware of it. Of course he is, he developed it. But how will all those lossy codecs react to those plain old measurement signals he describes as unneccesary? I mean, lossy codecs are designed to work very well with music. Measurements are entirely different... those don´t fully adhere to all the psychoacoustic principles so vital to every lossy codec so effectivey they should measure horribly. To find out how they react to these difficult conditions I used RMAA and compared all encodings to the original .wav file they were derived from (a 32/48 kHz file). Therefore, the resulting, decoded test files were at 32/48 too (WMA Professional: 24/48). Look below at Fig. VIII and see for yourself how they managed to deal with this situation.

I have to repeat that lossy codecs behave worse with measurement signals, they are simply not developed for this kind of signal.

Fig. VIII

Duh... again. Think about it: roughly 90% of the test signal has been removed... gone forever... Poof! (WMA Professional: 80%) - yet they still perform so fuckin' well... and of all things with signals they aren´t even designed to encode well! But you probably glimpsed that there are quality differences between all those codecs. OGG for example ends up badly, this is due to OGG not being revised often enough. The same goes for WMA Professional; while it clearly is one of the winners in this contest it has an advantage because of its comparably high bitrate. Ironically, the oldest codec, MP3, fares best in RMAA's quality assessments. The picture changes somewhat when one looks at the graphs... to make a long story short, the best codec is AAC as used and continiously developed by Apple for iTunes. The AAC version from Nero isn´t able to hold up that well, again caused by long pauses in development.

7. Listening to differences

Measurements are one thing, listening to results another... and I´ll simply show you the residue that is produced when MP3 encodes music. I chose something you might already know, a song you´d hear on the radio: Woman's World (-> video) by Cher from her first album in 12 years, Closer to the Truth. The residue is called - see above - quantization noise, in this case mixed with imprecisions produced by the filterbanks MP3 needs to find out what it might erase. These quantization artifacts are usually hidden by the music itself (this is supposed to be that way). In order to reveal how MP3 works I simply inverted the phase of the original 32/48 .wav source file and mixed it with the decoded MP3 file. Et voilà, every little speck of dirt MP3 produced for this file is subsequently revealed. I didn´t alter the gain of this file, the level you´re hearing is the actual level of the errors within the MP3 file. Now, for those who´ll now say "Very loud... and that´s exactly why MP3 sucks big time" I can only say that you´re stupid. Just consider for a moment how our ears & brain perceive sounds and have a look at this: the RMS-level of the original file is -15 dB, the residue has an RMS-level of only -43 dB. A gain reduction of just 6 dB feels to our ear / brain half as loud. I think, you can do the math yourself.

Again: you cannot use this noise to point out how badly any lossy codec performs - it´s how it´s supposed to work, nothing else. As I said: (with a high enough bitrate) usually these artifacts are hidden and inaudible.

Sounds funny, doesn´t it? You can hear that the residue mirrors the original file closely, except that it´s stripped of bass and mids. Our ears & brain aren´t very good when it comes to high frequencies. Hence any lossy codec prefers to focus on the treble area. You can also hear that artifacts rise in level when the music gets louder and that their level decreases when the music gets softer or less complex. This is why lossy codecs are able to encode high resolution music - the additional resolution is kept. After all, high resolution is, when it comes to bit depth, nothing more than lowering the static quantization noise floor. To be fair, the artifacts left by MP3 are anything but static, they are chaotic and at all the places where MP3 removed information. BTW, the MP3 used for this example was encoded with 320 kBit/s. When the bitrate is reduced to, say, 128 kBit/s, the artifacts are considerably louder.

Update 30.03.15: Soundcloud used to host the audiofile containing the compression artifacts. But just this day, Soundcloud decided to delete everything I ever uploaded because their automated content protection system detected several breaches of copyright.

Well, of course it did! For my reviews I need to listen to music and in order to make sound differences available to you, dear Reader, I uploaded several samples, each of them - at max - 30 seconds long. Naturally, this isn´t a breach of copyright, because a) I don´t have a commercial agenda nor background for this blog and b) I don´t advertise filesharers nor do I encourage to download things illegally. I don´t even want to mention, that only 30 seconds (!) of a particular song or piece are far too short to be enjoyed properly by anyone who attempts to be an illegal asshole. Yet Soundcloud fears the lables and their paranoia of copyright breaches which in turn prompts them to be paranoid and incompetent ninnies themselves.

I hate paranoia, I don´t want to have anything to do with stupid people / companies and everything was deleted anyway... so I decided to delete my Soundcloud account. Sorry for that, dear Reader.

8. And now... the problem

All of this would be marvellous... if all those nifty lossy codecs would be decoded properly. Have you ever heard about a digital audio player or a smartphone that decodes lossy codecs with 32 bit floating point precision? See, neither have I. And that´s where the beast rears its ugly head: if they aren´t decoded with full precision they produce strong, additional quantization distortions NOT contained inside the signal itself. Software players for the PC usually don´t suffer from this malady, foobar2000, Winamp or JRiver decode MP3 & Co. with full floating point precision for simple playback. This makes perfect sense because the aforementioned softwares work internally with 32 or 64 bit floating point precision anyway (for DSPs, volume control, etc.). JRiver deemed this issue so important that they opened a thread in their forum, talking about it. But should you attempt a conversion from MP3 to WAVE, the basic problem is resurrected, meandering around again like a zombified corpse. Foobar2000 for example assumes that every lossy file was derived from CD; using the 'Auto' bit depth configuration in its converter dialogue converts everything lossy to 16 bit, whether it´s MP3, OGG, AAC or WMA (see Fig. IX). High resolution? Forget it. The same goes for the usually wonderful software dBpoweramp: floating point decoding has to be activated under advanced options (Fig. X).

Fig. IX: foobar2000 converter dialogue

Fig. X: dBpoweramp configuration, advanced dialogue

By all means, the 16 bit, quasi-standard decoding of MP3 & Co. isn´t a good thing. Imagine a CD you ripped yourself to MP3; these files were derived from a normal 16 bit source. 16 bit decoding should be enough then, right? Should, but is not. While the decoded data boasts the sources' original bit depth again, something new and eerie has been added... and I don´t mean the inlying compression errors produced by the encoder. No, this thing from the crypt is additional quantization noise produced by the decoder. Cause: truncating floating point values to integer values. These additional artifacts are produced only because the decoder works at half speed and with half of its options. Look at the graphs below:

Fig. XI: original, lossless 16 bit wavefile for comparison

Fig. XII: MP3, decoded with 32 bit floating point

Fig. XIII: MP3, decoded with 16 bit integer

Doesn´t look so bad, you´re saying? Well, then look again at the signal causing the noise, it´s a simple 1.000 Hz sine only. Fig. XI shows an original, lossless 16 bit wavefile, the quantization noise is evenly distributed across the spectrum. Fig. XII shows an MP3 file that has been decoded with floating point precision and while there are distortions, those are well below audible levels; most of them are at frequencies we cannot hear well. Fig. XIII shows the same file, this time decoded truncated to 16 bit integer. The artifacts have doubled - and only because they were decoded with 16 bits instead of 32 bit floating point. They might still be inaudible... but I´m not sure because now we have additional aliases at frequencies where us humans can hear extremely well. These distortions will be added by the stupid integer decoding, and it doesn´t matter if you have created those lossy files yourself or bought them at some online store. You´ve probably been listening to quantization artifacts all your life, errors produced by dumb decoding of portable players and stupid software. Even files encoded with WMA Professional, files that are clearly marked as being 24 bit by their data stream, suffer from erroneous decoding as most software decodes it to 16 bit only. I´m looking at you, foobar2000. Just because you´ve been programmed by people who give a shit about proprietary, 'bad' software coming from Microsoft, this still doesn´t mean that you have to behave like a silly goose.

9. The solution

I´m afraid that for the time being there is no solution. Companies producing digital audio players don´t seem to be aware of the problem or they just assume that your average-joe won´t notice it. Well, at least you can do something about it when decoding those files yourself with software. Just have a look at dBpoweramp again (Fig. X above) and configure it to decode MP3, AAC or OGG as 32 bit floating point or 24 bit. Do the same with foobar2000 in the converter dialogue (see Fig. IX) and change the output bit depth to '32' or '24', it´ll then decode lossy formats to their full potential when converting files to HDD. But there´s hope that this problem gains attention... people like Bob Katz who´s a mastering engineer and important enough to maybe excert some influence, mentioned this problem in a recent thread at the JRiver forum. Yeah, he was talking about dithering but at the same time he was fully aware that for MP3 & Co. it´s imperative that they are decoded properly and that decoding those lossy codecs with anything else than floating point will result in truncating values which in turn produces the aforementioned artifacts. He wrote:

"Did you know that all current Lame and Fraunhofer and Apple AAC and MP3 decoders run internally at 32-bit floating point? In fact, if you take a "16-bit" source AAC file and reproduce it through the AAC decoder, it produces a 32-bit float output word! If it was a very good encoding, you will lose audible depth if you reproduce it at 16-bit because more than 16-bits come out of the decoder. The output of an AAC decoder should therefore be dithered down from 32-bit float to 24-bits for best reproduction. Almost NO ONE does that, but they should, and I've heard the audible difference when I play AAC in an engine that permits that."

Thank you, Bob, exactly what I´ve been saying! At least someone acknowledges it. I won´t dither to 24 bit but each to his own (I´m not too fond of dithering to 24 bit, using a bit depth like this renders quantization-related problems moot). From now on, I will mention if digital players (portable CD players for example) are able to decode MP3 properly.

10. How to make the best MP3s (& Co.) ever!

In the meantime I´ll present some advice everyone encoding her/himself might find useful. I´ll also recommend the best sounding codec to you, based on my own personal experience. Please be aware that my suggestions will cause any encoding to take longer, if you don´t have the time to wait for the encoder, then don´t even bother. But then you´re not interested in best sound anyway, or are you?

First of all, all those nifty codecs are command line based, on a Windows PC they look like plain old DOS. Usually you can´t see this as the command line window is hidden by the software employing those decoders/encoders. But it nevertheless enables you to make everything yourself by using your keyboard... though I admit that it isn´t very convenient. So we´ll use nice, flashy software instead. The following encoding tips and setups will depend on the codec of your choice; in some cases their setup might be complicated for noobs but once you´ve done it correctly you won´t need to bother with it anymore.

MP3 (LAME)

One of the best and most versatile softwares around the net is the aforementioned dBpoweramp. It costs money but I can recommend it without reservations. It´s a powerful encoder/decoder for any format you can think of, it contains one of the best available CD rippers and its powerful talents are hidden inside an easy looking package. Even better, the CEO of Illustrate (company maintaining dBpoweramp) is a nice guy, discussing things and giving advice in his own forum and on hydrogenaudio. A free alternative would be xrecode II (shareware with a nag screen) but it´s buggy and inconvenient - use at your own risk. So, in case you want to use MP3 and in order to create the best sounding MP3s ever I recommend these encoding settings (Fig. XIV):

Fig. XIV: best encoding settings for MP3 (LAME)

If you´d like to use a different frontend for the command line based encoder/decoder instead  (xrecode II for example) I´ll now give you the commands so that you may copy and paste them:

-b 320 -q 0 --noreplaygain

The most important part is the '-q' switch, it configures the 'quality' option of the LAME MP3 encoder. The standard setting advertised by hydrogenaudio is '-q 2' but we want a choice and the best quality so we opt to set it ourselves. Why the constant bit rate (CBR) of 320 kBit/s when Hydrogenaudio recommends variable bit rate (VBR) in order to save on storage space? Think about it: 60 minutes of music occupy 137 megabytes when encoded with 320 kBit/s CBR. For VBR with an average of 240 kBit/s these 60 minutes take roughly 108 megabytes. A difference of 29 megabyte. In 2013, two to three photos on a smartphone alone consume this. We have to be realistic here: it might have been an issue 10 or even 5 years ago but nowadays with an abundance of storage space anywhere, surely we can afford bigger files. Furthermore, MP3 profits from more bitrate, no matter what the skeptics (hello, my dear hydrogenaudio-ists) are saying. More on sound issues later.

WMA Professional

Fig. XIV: best encoding settings for WMA Professional

I wouldn´t recommend WMA Standard as WMA Professional is superior in every way, it´s also the one codec officially supporting 24 bit output. Which is a fake of course, Microsoft just embedded an additional integer decoding option, internally it works with floating point just like other codecs. As a quality option, you should always use 2 pass encoding, it yields audibly superior results. Compared to MP3 above, 60 minutes of music occupy more space because of the highest 440 kBit/s setting: 189 megabytes. Should you really be concerned about storage requirements, using 384 kBit/s would work too, it´ll still sound well. None of this however hides the fact that WMA Professional enjoys close to zero hardware support. Most portable devices are able to play WMA Standard only - and I can´t recommend that one.

AAC (Apple)

Instead of using an old or proprietary codec I´ll recommend one of the most recent instead, one that also comes with ample hardware support by almost every manufacturer: AAC. Be advised that it now gets inconvenient. To make it easy, you could of course use the AAC codec from Nero (with dBpoweramp for example) but I´d advise against it; the one from Apple included with iTunes is much better (and it shames me to write this as I don´t like Apple). There´s only one way to unleash the iTunes encoder (or more precisely: the QuickTime encoder) with the best quality options: qtaacenc (get it here). You´d have to setup foobar2000 like this (Fig. XV):

Fig. XV: foobar2000 setup for qtaacenc

Just like with MP3 there´s a quality setting allowing for better-than-standard results (the standard settings are used by Apple for music they sell through the iTunes store): it´s simply called '--highest' (another source says '--high'). Don´t forget to instruct foobar2000 to use 32 bit during encoding, you now know well that AAC can handle it. Anyway, here are the commands in case you want something else besides foobar:

--cbr 320 --highest

11. This really is high resolution? Some disadvantages

MP3 and AAC (Apple) have one big disadvantage: they aren´t able to handle samplerates beyond 48 kHz. For those there´s only one codec left: WMA Professional. AAC (Nero) can handle 96 kHz too... but as I said above it´s not very good. Furthermore, devices usually able to handle AAC will react in strange ways (or not at all) when trying to play those 96 kHz AAC files. So you might want to use WMA Professional. But its main problem hasn´t changed: hardware support is extremely limited, not even software properly recognizes it.

What lossy codec to use depends on what you yourself consider to be high resolution. To me, HiRes starts with 24/48, for others it starts with 24/44.1. Strictly speaking, everything that´s not CD is high resolution. Take HDTracks: they sell even 24/44.1 as high resolution. More than one third of the music they´re offering is at 24/44.1 or 24/48. For those releases lossy codecs would be the perfect choice if you could ensure proper decoding on playback. And if you want to save on storage space, you might consider resampling your 96 kHz albums to 48 kHz and encode the result it with AAC or MP3.

12. REAL "lossyless" high resolution

But the best combination to save space and keep any file at its original resolution is... not FLAC. Have you ever heard about WavPack? Thought so. WavPack normally is, just like FLAC or APE, a completely lossless encoder/decoder that won´t ever touch the material it encodes. But it has a second, not so well known setting: WavPack lossy ('hybrid' is the correct designation). This 'lossy' mode is unlike MP3, AAC or WMA, it won´t remove anything within the music. As I said, the other lossy codecs work psychoacoustically and 'hack' into the frequency band at countless places, removing what cannot be perceived. WavPack lossy ignores psychoacoustics and does only this: reducing overall bitdepth of the file according to its level and distribution of frequencies.
Let´s assume a 24 bit file with lots of loud and soft parts. With WavPack lossy the soft parts will retain close to 24 bit resolution, loud parts will be reduced to 16-20 bit resolution. The resulting quantization noise is then moved by a very tame noiseshaper towards high frequencies where it cannot be perceived anymore. Very much like SACD. Unlike SACD though, bit depth isn´t static. WavPack lossy can be described as a 'dynamic bit depth decreaser'. In that respect it also differs from other lossy codecs; with them bitdepth changes a lot within frequencies, with WavPack lossy only from one level change to the next. It truly preserves the full bandwidth and dynamic range of high resolution material and in my experience, the added quantization noise remains completely inaudible. Remember the MP3 noise sample up above? If I would have extracted the artifacts of a WavPack lossy-encoded file, I´d have been required to raise the level by +50 dB to make them even audible!

Fig. XVI: RMAA chart comparing several lossy codec against their lossless source

Please refer to Fig. XVI above where three lossy codecs have to compete against their own source. While WMA Professional is superior to AAC Nero, it´s WavPack lossy winning the contest. It measures almost exactly like the original. Yet it´s 7 times smaller than the WAVE file it was produced from (1.200 kBit/s against 6.144 kBit/s). See Fig. XVII & Fig. XVIII for details.

Fig. XVII: total harmonic distortions - WAVE and WavPack lossy are the clear winner

Fig. XVIII: intermodulation distortions - WAVE & WavPack lossy measure the same

Because of its noiseshaping feature, WavPack lossy is ideally suited for encoding anything beyond 48 kHz. If you employ the '-x6' switch, you can exploit this further. Using this switch, I´ve found out during many hours of testing that WavPack lossy is fully transparent with 96 kHz material the moment the bitrate exceeds 1.000 kBit/s. Because of that I always use it with 1.200 kBit/s, as an additional security I also employ the '-h' switch (high quality). However, encoding takes forever. To me, this doesn´t matter; I only encode those files once and never touch them again. But with you it might be different - so you decide.

Fig. XIX: WavPack frontend with my recommended settings for best lossy quality

Fig. XIX presents my recommended settings for 96 kHz material using the WavPack frontend (for convenience I would´ve loved to recommend foobar2000 or dBpoweramp... but both won´t allow extra commandline switches - as it appears, foobar2000 now can. Very convenient, because it´ll use all your CPUs cores for additional encoding speed). The 'Extra Option' -x6 is most important to achieve the best possible quality... but using it will prolong encoding time (no kidding; while you wait, you could write a novel). The same goes for the '-h' switch and because it´ll also prolong time needed for decoding, it´s optional and not mentioned anymore. Anyway, for 44.1 / 48 kHz those 1.200 kBit/s are overkill, for 176,4 / 192 kHz they aren´t enough. That´s why the bitrate needs to be tailored to the samplerate:

44.1 / 48 kHz:

bitrate: 500-600 kB/s, switches: -x6 (optional)

88.2 / 96 kHz:

bitrate: 1.000-1.200 kB/s, switches: -x6

176,4 / 192 kHz:

bitrate: 2.000-2.400 kB/s, switches: -x6

13. The sound of noise

One of the most important questions of this article is how lossy codecs sound. And I don´t mean what the mainstream public thinks they are sounding which can be answered easily: cold, digital and lifeless (yet all use them - a mystery?). No, I mean the actual sound quality. I´ve said above that lossy codecs are far better than their reputation. BUT: you have to make sure to extract the best possible quality when using them! Please consider my encoding suggestions above again and remember that they take time but are worth every second spent on them. The best encoding isn´t the one that is the fastest (C'mon... a whole album encoded in 40 seconds... really?). To make it simple: the higher the quality, the longer it´ll take to encode. Just deal with it. Back to topic... if you used the best encoding options along with the highest bitrates and if you decoded all these files correctly (-> floating point) the sound of several lossy codecs is like this:

WavPack lossy: perfect. Using bitrates of roughly 1.200 kBit/s with 24/96 in combination with the -x6 switch it sounds exactly like the original. Always. Very limited hardware support... software support is good though. I have been archiving every bit of music with WavPack lossy since 2008 and I´ve never looked back. Not suited for portable use (for lack of hardware support), but perfectly suited for archiving and transcoding to other codecs like the ones below.

AAC (Apple): might produce instable staging, instruments occasionally seem to change size and position. Yet this happens so rarely that I could have been imagining it. Otherwise it´s completely devoid of artifacts. Sounds O.K. enough with lower bit rates. Hardware support is phenomenal. Simply the best mixture of convenience and good sound, therefore highly recommended for portable use. 

MP3: slightly 'dark', 'warm'. Somtimes sounds too dry, as if reverb has been reduced (especially audible with VBR). Smeared transients are another problem. No matter the bitrate, MP3 will always have difficulties encoding really short and tiny transients (the sample size for short blocks isn´t small enough). However, in 85-90% of all cases all of this isn´t audible at all. The danger of typical problems like metallic sizzling, flanging, etc. completely disappears if you use the highest bitrate of 320 kBit/s. Of course, using LAME avoids most problems anyway compared to other MP3 encoders. Hardware support? A 100%.

WMA Professional: on occasion creates instable staging worse than AAC (Apple). Instruments might change size and place, dimensions shrink or expand sometimes. This depends very much on the material, in many cases it´s completely inaudible. When used with 96 kHz, sound is too mellow. Otherwise it´s one of the most neutral and artifact free codecs available. Hardware support: laughable.

AAC (Nero): like WMA Professional, only (much) worse. Instruments always move around slightly, sizes vary as well, dimensions shrink and expand constantly. Furthermore, the sound feels 'blown up' at lower mids. Despite being much more recent and advanced than MP3 (LAME), it doesn´t sound remotely as good. Hardware support is - naturally - the same as for AAC (Apple).

WMA Standard: sizzles. Even with higher bitrates. On the other hand it enjoys almost the same hardware compatibility as MP3. I still can´t recommend it, it just isn´t good enough. 

OGG: was my standard choice more than 8 years ago. Shouldn´t be used nowadays. Obscures much of the virtual stage, sounds harsh (this is the only codec sounding literally 'digital'). Sounded different back in 2005: very beautiful and pleasant. Ignore it.

14. Transcoding Horror

If there´s one thing I hate, hate, hate people doing it´s transcoding from one lossy codec to another... or from 128 kBit/s to 320 kBit/s. You cannot imagine how many think that it actually improves quality - see here & here (I´ve found countless other examples, but they´re in German and my native language is a bitch for most people). The only thing it does is hurting quality immensely. To the encoder, the file that was compressed before is just a new file and it´s treated as such. Example: the really loud encoding artifacts within a 128 kBit/s MP3 are simply treated as they are, to the encoder they are just new musical information, a seemingly natural part of the music. You see, not one encoder in the world can distinguish between noise, artifacts or music... which means that artifacts are now treated as music. Back in 1999, when Microsoft tried to advertise its WMA Standard codec as superior to MP3 out of licensing greed, they tried to fool everyone to re-encode their MP3s to WMA. For this stupidity they should have been shot... so please, don´t you ever transcode from MP3 to MP3, AAC, WMA or OGG. The only allowed trancoding is one from WavPack lossy to another lossy codec (preferrably not WavPack lossy again).

15. Conclusion

My dear constant reader, I don´t know where you´re coming from, if you´re an audiophile, a skeptic or just feeling lucky to be here. But I know one thing: if MP3 & Co. would be sounding horrible no one would use lossy coding in the first place. If you´re an audiophile, do you really think that half of the earth's population fell for a ruse invented by some German scientist in order to 'dumb down' the sound of music? Then let me tell you that all the hills marbles of these 3 billion people are very much alive and there. This is no conspiracy, MP3 & Co. sound well enough that most people won´t even think about other formats. And as I´ve proven with this article, they are prepared to exceed and surpass mass-market sound with ease. Only if my suggestions are heeded, that is. Should you decide to follow them you´ll be rewarded with true high resolution sound... or high end quality, if you will. All coming from some 'dumbing down' lossy codec. I can´t stress it out often enough: take care in using the best possible quality when encoding with lossy codecs yourself. Take care in trying to make sure to employ the best decoding as well. Only then will you enjoy a sound quality you wouldn´t have expected from MP3 & Co. Happy encoding... and perfect decoding!

Last update: 29.12.2013

Vodoo continued: my final Statement on USB cables

This article here is the final sequel to "Hot Vodoo: Audio differences between USB cables - yes, they do exist" and "More USB cables - more differences". So before you read this one go ahead and scroll through the other two.

Almost one year has passed since my last article concerning USB cables and their sonic signatures was published here. Truth is that I ran out of steam: I didn´t want to do it since I was afraid that I had been measuring the effects of ground loops only. But before I start with the embarassing truth I´ll tell you about the stir those two USB articles created. The articles I refer to (linked above) were written almost a year ago and since then they have been popping up constantly in several online forums. I´ve received visits from Russians, Germans, Americans, Norwegians, Japanese or Chinese citizens to name but a few. Sometimes the participants of these forums made fun of me, sometimes they tried to find out how I was doing things, sometimes I was accused of lying or imagining things and on some occasions people actually tried to help me. It fascinates me to this day that 99 % of all those people didn´t comment here on my blog but only in their respective habitat, forums I´ve never heard of before. Since I´m the most gleefully cunning person I occasionally registered there just to take part in their discussions... and boy, it was delightful to witness how fast participants there transformed from being bullies into polite, elaborate people upon myself unexpectedly taking part in those discussions. Without me it was 'chic' to mock me - so why did people stop when I suddenly entered? I love mockery, especially when it´s about me (I tend to be too self-centered sometimes). This wasn´t restricted to one country, no, it was a trait exhibited by people coming from all around the world. For all those who were doubting me throughout the last year I assume this article will be a field day which is perfectly alright - yet it saddens me that people generally seem to ignore transparency and truthfulness. But let´s start, eh?

USB cables revisited: only the Belkin and the Audioquest were used

In the last article I compared five different USB cables, for the article you are now reading I used only two of them: the Belkin which performed the best & the Audioquest which performed the worst. Since the measurment differences between those two were the strongest I assumed that they´d be the most likely candidates for also showing differences today. Likewise, I used my Creative Labs Soundblaster X-Fi HD USB again... BTW, that was the most peculiar thing... because in those forums I mentionend above I was attacked for measuring cheap hardware 'on my kitchen table'. Well, of course I did, that was the whole point. What good would my test have been if I would measure with expensive boutique hardware (which I don´t own, thank you very much)? I wanted to show results with everyday hardware, something people would actually buy. That way my results would (hopefully) be more comprehensible. Furthermore, assuming that something more expensive is behaving in a superior way is audiophile bullshit. One word about the 'kitchen table' comment... I will ignore it since it reveals opinions much too conservative for my taste. Just something to think about. Back to topic: I had to use the X-Fi HD USB and my ASUS Xonar Essence ST again since I wanted to compare possible differences between one year ago and today. My testing methodology may have changed / improved during the last year, yet I still have to make sure to repeat testing circumstances exactly as before to obtain reliable results.

Creative Labs Soundblaster X-Fi HD USB was used again

For that reason my testing chain then and now consists of the X-Fi HD USB connected via the Audioquest King Cobra RCA cable to the Xonar Essence ST. The last time both soundcards were powered by my desktop PC and since that obviously created the problematic grounding loop I now took a slightly different approach. This time I used my nine year old laptop, an FSC Amilo 1425 which was used to power my X-Fi HD. Naturally this laptop is able to run on batteries, it completely avoids grounding loops that way. Everything else was the same though: the Creative still was connected to the ASUS with exactly the same RCA cable. The laptop uses Windows XP as its OS so I had to utilize the Kernel Streaming plugin for foobar2000 to avoid potentially harmful resampling (one year ago: WASAPI direct connection on Windows 7). Another difference to the older test is that I decided to show you the results of ten test runs (five for every USB cable). I also did five test runs for the last test but I sadly failed to disclose their results. So, let´s have a look at the facts, shall we?

For comparison: Old results, one year ago, both interfaces run from the same PC

Belkin USB cable, two different PCs, one running on batteries,
five different test runs which have been averaged for this table

Audioquest Forest USB cable, two different PCs, one running on batteries,
five different test runs which have been averaged for this table

Shit! You now can see that one year ago I indeed measured the effects of a ground loop caused by using the same PC for both interfaces. With one interface running on batteries and by that removing the ground loop I also removed the measurment differences between the two USB cables, the lower resistance of the Audioquest is not important anymore. Yes, the X-Fi HD USB now measures EXACTLY the same with two different USB cables. Which probably means that 1. differences between USB cables don´t exist or 2. that I indeed can´t measure them with my limited hardware or 3. that I need more and different measurments. In any way, two things are now crystal clear: the X-Fi HD USB is a gorgeous performer on its line-out while the Xonar Essence ST boasts an equally impressive line-in. With measurment results as fantastic as these it´s hard to imagine that more expensive soundcards can do much better; you also have to keep in mind that both cards are relatively cheap and are far away from being hardware for demanding professionals. Just take the noise levels: they have improved by roughly 12 dB - Creative was indeed telling the truth by claiming a noisefloor as low as -114 dB for the X-Fi HD. Back to the USB cables: the only differences I was able to find were so small that they are truly insiginificant:

X-Fi HD USB, Belkin USB cable

X-Fi HD USB, Audioquest Forest USB cable

Click on the upper of those two pictures, that way you can scroll between them using your mouse wheel - only then you´ll see that the X-Fi HD USB has a very, very slightly improved high frequency performance with the Audioquest. That´s it. You could now say "Hey, there is a difference!" Erm... no, there isn´t. It is way too tiny to be of significance. Maybe distortions are able to reveal something else?

X-Fi HD USB, THD & Noise, Belkin USB cable

X-Fi HD USB, THD & Noise, Audioquest USB cable

Again, click on one of those two pictures and scroll between them with your mouse wheel and you´ll see that nothing changes - again. If at all, only the noisefloor seems to reveal a slight tendency in favor of the Belkin so I won´t bore you with additional non-existing differences between two pictures. Today I´ve learned two things: ground loops are evil & USB cable differences cannot be measured when using my approach. Phew, now I´m embarrased: I wrote two articles that were basically describing the effects of a ground loop! Stupid, stupid, stupid. But wait, has the sound changed? That´s where it gets interesting because it hasn´t. The X-Fi HD USB still sounds its best with the Audioquest Forest USB and the worst with the Belkin. Let me point out that the sonic differences are indeed small, I assume that they won´t be audible to 90 % of all people. But I started this article with an emphasis on being honest and truthful (which I always am - and which is painful here) and therefore I´ll tell you that I had doubts about my ability to hear things; I was afraid I might have encountered a prime example of a placebo. So I did what I usually detest, namely a Double Blind Test, the results of which you can see below:

foo_abx 1.3.4 report
foobar2000 v1.2
2013/01/09 02:54:00
File A: RAM-Disk:Audioquest.wav
File B: RAM-Disk:Belkin.wav
02:54:00 : Test started.
02:55:31 : 01/01  50.0%
02:56:05 : 02/02  25.0%
02:56:27 : 03/03  12.5%
02:57:05 : 04/04  6.3%
02:57:18 : 05/05  3.1%
02:57:28 : 05/06  10.9%
02:57:45 : 06/07  6.3%
02:58:01 : 06/08  14.5%
02:58:19 : 07/09  9.0%
02:58:33 : 08/10  5.5%
02:58:48 : 08/11  11.3%
02:58:59 : 09/12  7.3%
02:59:17 : 10/13  4.6%
03:00:01 : 11/14  2.9%
03:00:17 : 12/15  1.8%
03:00:35 : 13/16  1.1%
03:00:51 : 14/17  0.6%
03:00:54 : Test finished.
 ----------
Total: 14/17 (0.6%)

I had occasional problems making out differences. It took me 17 trials during roughly seven minutes to find out if I was imagining things - but now I´m at least confident that my ears (and brain) are still fully functional. I´m no so-called 'Golden Ear', those are a myth. Audiophiles often claim that they are able to hear more than, say, skeptical people relying purely on measurments - which isn´t true. The only thing that´s valid to state is that there are people out there who have trained themselves to hear differently compared to others. I only hear this because I know what to listen for, nothing more. I´m also proud of my (perhaps imaginary) ability of not falling for typical placebo effects. "How can Marlene be so arrogant and full of herself?" you ask... maybe you´re right, look what being conceited brought me: grounding loops! All kidding aside, the situation of most of the time knowing what to listen for really isn´t the easiest thing to live with. I sometimes wish to have the gift of blissful ignorance because everytime I hear an album I immediately spot errors or defects & I cannot enjoy the music until they are gone. This might sound contrived to you but it prompts me to ignore bad sounding albums, even if the music is engaging. Imagine the time I could save by not being obsessed with these things. Enough with the cocky self pitying and back to topic!

Measures extremely well on its line-out: Creative Labs Soundblaster X-Fi HD USB

I´m afraid there isn´t that much to conclude. I still hear the effects of different USB cables, yet I´m unable to measure them; the centuries old dilemma of audiophiles. It all leads to you and the decisions you make, dear reader. Are you a believer or a skeptic? I hate to exploit this common prejudice but I started measuring USB cables through external soundcards as a means of diminishing troubles between those two groups - I thoroughly failed. The only thing I did was creating a lot of hot air. Shame on me. I did however learn a few things: more expensive cables have less resistance causing ground loops to leak through more easily, that I cannot measure everything, that ground loops hamper effective resolution and that the X-Fi HD USB is a great performer on its line-out. But where is the benefit for you?

Audiophile bulls**t & final thoughts about nine reviews

Hallo meine süßen Lieblinge!

During the last few weeks I´ve reviewed the following units: Kenwood DP-5090, Pioneer DV-610, Sony CDP-470, Pioneer BDP-140, iRiver IMP-550, Sony MZ-R 55, Sony MZ-R 37, Sony NW-A 1000 and the Sansa Clip+. You might have wondered where I was getting at with all these reviews; I usually have an overarching theme or reason for my articles, haven´t I? Well, I planned to feature all reviews in just one article, in the end stating my conclusions - of course, that would have enlarged that article into gross proportions. So I seperated the already written article into ten singular parts, the one you´re about to read contains my conclusion and opinion.

For as long as I can think I wanted to read a review where units from different classes are compared to each other. Why not comparing a portable player playing lossy music to an SACD player spinning high resolution media? I mean their goal is the same: playback with a-good-as-possible quality. But have you ever read a review where something like that happens? I have not and I assume peoples' misconceptions about compression, size, weight, number of parts and price have something to do with it. Ask yourself: why should a tiny, portable player be worse than a much bigger stationary player? Can less available space really be a reason for worse sound? I also wanted to know how well ancient players would hold up to more modern devices, a universal disc player for example. You only have to take a look at eBay where more than 20 year old CD players are able to reach high prices; something that also applies for units that were not even top-of-the-line back then. Yet recent universal players are derided as being far away from high fidelity sound by magazines and the public alike. People seem to believe that everything was better in the old days... a curious thing really when you think about it: assuming modern units contain more advanced parts why should they be worse? Common conceptions therefore are:

• old units have better built quality
• they contain much more parts - good for sound, isn´t it?
• ancient units are heavier - must sound better
• small units with their space restrictions must sound worse
• older technology sounds less 'digital'
• huge, conservative power supplies are vital for perfect sound
• lossy media cannot ever be competitive against high resolution material

I could extend this list to infinity but it wouldn´t make much sense in light of the contents of this article. I could for example mention the recent craze about NOS (Non OverSampling) D/A converters... there are so many things wrong with this approach that it´s very confusing that people even consider it as an option while ignoring 30 years of technical progress. Or tubes... no, I won´t start. I´d rather talk about audio magazines  seemingly confirming those conceptions about quality as stated above. You see, magazines only compare units from their respective classes, they also support the view that smaller gadgets must sound worse simply because of their size. But the biggest thing they are claiming is that High End audio sounds much better than 'normal', standard audio. For some years now I have the opinion that High End audio isn´t really better, only different. Expensive media players for example are more or less tweaked to sound a certain way, a sound pleasing to the common audiophile. Companies subsequently tweak their units towards sonic goals such as warmth, precision, punch, snap etc. Will those expensive toys sound like the original? Of course not. They will sound well with Rock or classical music and not so well with anything else - but that doesn´t matter since few audiophiles actually branch out of the musical genres they usually listen to. I have to admit though that heavy units look and feel better, you extend a considerable trust towards them by assuming that weight and complexity are good traits for players.

Let´s ignore my own and your prejudices and take a look at the actual thing. In this article you won´t have any of the prejudices or certain preferences with me because as I´ve stated many times before I don´t compare any unit to another one, I compare directly to the master used for the creation of the testing media itself. No one else does this and I believe my approach to be genuine and also to be the only methodology coming close the precision of DBTs without their disadvantages. Regarding my reviews I didn´t know what to expect or what I´d be ending up with. I thought it possible that all the conceptions above might be true while also accepting the opposite. Naturally, this article reflects my findings, with different hardware it could have turned out different. So you have to be aware that I don´t regard my conclusions as written in stone, they could change depending on experiences with other units. I also accept that you, dear reader, won´t agree on a comparison of high resolution to lossy material. But think about it: the advantages of high defintion might be swallowed by the imperfections from the device playing it back so that in the end lossy material coming from a superior unit might be the better representative.

Four stationary players: Pioneer DV-610, Sony CDP-470, Pioneer BDP-140, Kenwood DP-5090

For better comparison of all the reviewed units I have calculated the average of the points they earned, these are subsequently used for a ranking (this is a first for my blog). Be advised that those averaged numbers obscure the differences between the units, a unit like the MZ-R 37 for example might not have a sonic character suitable for you despite receiving a high number of points. For the described details and difference please have a look at the respective articles linked on the top of the page. The results were surprising on some occasions, on others however they were not:

1st place

Pioneer DV-610, SACD:

4.6 Points

2nd place

Pioneer BDP-140, CD/SACD:

4.4 Points

3rd place

Pioneer DV-610, CD:

4.0 Points

4th place

Sony NW-A 1000

3.8 Points

4th place

Sansa Clip+

3.8 Points

5th place

Sony MZ-R 55

3.6 Points

5rd place

Sony MZ-R 37

3.6 Points

6th place

Kenwood DP-5090

3.0 Points

7th place

iRiver IMP-550

2.6 Points

8th place

Sony CDP-470

1.2 Points

If you´re a believer in high end units you are probably crushed now, right? A multidisc player playing SACDs is the winner (it really sounds close to the reference) while another multidisc player follows closely. Both multidisc units are lightweight, not manufactured that well, have very few parts and cheap plastic drives. Their power supply doesn´t seem to have received much care at the design stage, they 'only' have those dreaded and efficient switching power supplies supposedly known for causing bad sound quality on PCs. Likewise, small portable players are not necessarily worse in sound quality - as proven by the two 4th places, the Sony NW-A 1000 and the Sansa Clip+. The NW-A 1000 is especially interesting since it has to use lossy material - who would´ve thought that mp3 would sound that well? Most disappointing were the results for the iRiver IMP-550, the Sony CDP-470 and the Kenwood DP-5090. The latter was surprising because it contains many parts (some of high quality), has a decent power supply, a well built CD drive and a good overall built quality. It also received favourable reviews from magazines when released in 1997. But its sonic signature favors 'shrill' and 'piercing' way too much, it´s not neutral and introduces very audible differences though all of these could be pleasing for a certain target audience. The Sony CDP-470 was supposed to have a good old school sound... it does indeed sound old school - but in a bad way. It´s a very digital and harsh sounding player, the numerous parts on the PCB and the reliable drive are not helping. The Sansa Clip+ is beloved everywhere but doesn´t hold up to its reputation, due to age related imperfections it shows enormous amounts of audible distortions. The best gadget for the price would be the Sansa Clip+; while it´s not perfectly balanced its dynamic and slightly more aggressive sound signature will appeal to many people.

What does these results tell us then? Not as much as I´d hoped but nonetheless they pose some interesting questions: when a cheap multidisc player sounds close to the reference how can a high end unit have superior sound? Answer: it can´t. It probably changes sound into something different with a more charming character, appealing more to the intended audience it was designed for. When a player using as few parts as possible can sound so well and refined why are many, high quality parts even necessary? Answer: they aren´t. Both Pioneer players achieve their goal of good audio quality with a layout that only seems to lack sophistication, yet in reality it works well. If lossy material (mp3 or MD) holds up so well against lossless or SACD, are those even necessary? Answer: Yes and No. High resolution is indeed able to extract the last little detail and crispness - the real question is if it´s audible for everyone which I´d answer with a 'No'. It certainly isn´t audible on every unit as proven by the BDP-140. Generally, sonic disadvantages of lossy audio are exaggerated as well as advantages of lossless or high resolution audio. Are portable personal players worse than their stationary counterparts? Answer: not necessarily. The NW-A 1000 or the Clip+ perform well while the two stationary players (DP-5090 and CDP-470) really fucked up. Are ancient players any better than more recent counterparts? Answer: No. The last place is occupied by the oldest player, also sounding 'old'. If you want to listen to an example of the supposedly digital and distorted sound of the CD go for that model. Can units playing back lossy material be superior to units playing back lossless quality? Answer: Yes. Three players, using mp3, AAC or MiniDisc are superior or equal to four others using lossless material. The codec therefore doesn´t decide about sound quality, the device itself does (provided that one uses well encoded material).

All of this should prompt you to think carefully about purchasing an ancient unit, it should also create skepticism regarding fantastic reviews of high end gadgets and their supposedly gorgeous sound quality. Many parts or heavy units don´t automatically guarantee perfect sound. You should also regard the recent craze about high resolution with serious doubts; it can have an advantage and better sound quality but the differences are so small that many people won´t ever hear them. The Pioneer BDP-140 comes extremely close to the sound of the Pioneer DV-610 (using SACD) with CDs only. So please, think twice before you buy something or trust someone, go out there yourself and don´t stop comparing and being skeptical.

Last update: 17.06.15