mastering & MP3's. 2 Questioins:

[greggybud]

1. As a standard procedure, do you as a mastering engineer convert each song to MP3 and listnen?

As MP3's become more popular does it warrant it's own mastering? What I mean is if you go to any major commercial download site, have those MP3's been altered in any way other than going through a standard conversion just like I would when I convert .wav to MP3 with Wavelab?

2. Over the past year it seems I'm having an increasing difficulty detecting differences in MP3 from .wav when played in my studio. Maybe it is my hearing. Maybe it is the content of the music. Maybe better conversion? A while back I read some total BS about Emu designing a system that will make an MP3 sound better than a 16 bit CD. I can read all day about this subject, but I really need to get my ears in shape especially if companies such as Emu are going to release what seems to be just a lot of marketing hype.

I am wondering if anyone has any sample music references to A/B MP3 vs .wav. that will really amplify the differences? And what should I be listening for to detect the differences? Am I loosing it??

[MASSIVE Master]

1. Absolutely not. As "decent" as some MP3's can sound through good conversion, I know of no one who actually does something "different" to wind up at a final MP3 other than converting from the PCM data -

Good sound in = good sounding MP3.

2. Yes, you're probably losing it. Just kidding - Really, MP3's in high-res (224, 256, higher) really don't sound bad. A/B them directly against the PCM data that it was created from and you'll probably hear at least some subtle differences (especially at the lower rates, and very obvious differences at "crap" rates like 128 & 160kbps).

[Timecode]

I recently did a test on the new Rattatat album between a 320 Kbps rip and the wav - I could hear the difference fairly cleary, but it seemed to me that the mp3 was slightly louder, like maybe 0.5db RMS or less. I know the encoding can give you clips by changing the transients, but does it tend to boost RMS as well? That would be a pretty nasty way of 'making up' for the loss in sound quality by an increase in sound quantity. Just wondering if this is a common thing, an exception, or me just hearing it wrong.

Thanks, cheers
Matthew

[inlinenl]

I think you're right ... mayby i'm wrong .. but when converting to Mp3 using the Lame encoding ... it always get his peaks to 0.0 dB ( like normalize to 0.0 dB ) ... even when you wav/aiif is -0.3 dB ...

Mp3 are always clipping.

is this correct .... ????

regards/wim

www.inlinemastering.com

[Andrew B.]

Many MP3 converter programs have a normalise option, which I guess may be ticked as a matter of course by some vendors.

I find the most obvious place to hear the MP3 vs WAV difference is on cymbals, especially as they decay.

Andrew

[Nordenstam]

Originally Posted by greggybud

As MP3's become more popular does it warrant it's own mastering?

Depends. The typical ways of gaining loudness in CD mastering does little good for psychoacoustic coders. So in that case, I would say yes. If the CD is done without severe limiting and is technically correct level-wise in every possible way, it'll translate fine to lossy coding.


Been testing and reading up on this a bit and the impression so far is that lossy coders doesn't like modern CD mastering. Reasons seems to be both related to the way the coders work, and the way digital signal levels are treated and measured. What I've found so far is basically:

Coders divides the spectrum into frequency bands, resembling the critical bands of our own hearing perception. Within each band and somewhat to neighboring bands, loud frequencies will mask other frequencies. The basis of audio data reduction is to remove these signals we hopefully won't hear anyway. Different parts of the spectrum is given priority depending on what the psychoacoustic models believe humans hear. The rate of success depends on the coding models and the signal being coded.

This bit of theory suggests that limiters and jam packed audio may not be the best starting point for the coding. They may be tuning them for the typical audio of today, though I have a hunch this is not the case. It seems they are set for fidelity in vocal recordings and normal instrument sounds. Audio signals have harmonics that define the character of that signal. Distortion, often from cliping and limiting the signal in mastering, will add spurious information. This may gain priority in the coding process, effectively lowering the fidelity of the more important original signal. In any case, less limiting gives 'more space' around each sound, making the guesswork of the coder easier. To me, 'breathy' music sounds better when coded than massive masters. The text book facts seems to support it, but I dunno what really goes on in the coder.


Less subjective and perhaps more important is the sample point peak levels and the intersample peak levels(link to explanatory paper).

The sample points themselves will change in the coder process. The output of the decoder is a synthesis. Although it will sound much the same, the actual waveshape is not an image copy of the original. The peak points will move around a bit and probably end up with a slightly higher level. Maxing out at zero dB flat on the CD is guaranteed to result in clip'ing in the decoder process.

How much the peaks will change depends on the nature of the signal. If sample point peak level and intersample peak levels are nearly identical, the change will only be a few tenths of a dB. If the sample point peak level are far below the real intersample peak level, the effect is more severe. It also depends a lot on the selected bitrate, with lower bitrates gives more discrepancy. The coders are strickt low pass filters, any extraneous high frequency information will be removed. The result is to act somewhat as a reconstruction to 20-20K bandwidth on the waveform.

Limiters and cliping creates higher frequencies where the peaks have been squashed. Removing high frequencies smooths the flat peaks to softer waves with larger peaks. A typically extremely loud modern master with peak clipping often exhibits several dB's higher peak level from the lossy coding process.

The way to do it is to ease up on the flat squashing, rather pushing the bottom up if massive is what the clients want. And work with strickt control of the intersample peak levels.

Combined with the absolute need for lo-fi compability and zero dynamic range of headphone listening in loud outdoor ambient noise, it's a real bugger to master!

1. As a standard procedure, do you as a mastering engineer convert each song to MP3 and listnen?

Only when mastering for Mp3. Would love to have a plugin that does coding and decoding in realtime! For checking the end result when crunched through a codec. AFAIK, there is no such thing that caters for this specific need of mastering for mp3. Going offline to convert back and forth between Mp3 and wav formats is a long way around.

2. Over the past year it seems I'm having an increasing difficulty detecting differences in MP3 from .wav when played in my studio. ... And what should I be listening for to detect the differences?

Try converting to mp3 and back to wav, line files up to do seamless A/B switching between the wave file and the mp3. Also, line the samples up so the match accurately and try phase reversing one of the channels so you only listen to the difference product between the original and chrunched file. Should give you a clear indication of what to listen for in the Mp3. Creaky highend, broken glass" artifacts, flanging/phasing sensations.

Originally Posted by Timecode

I recently did a test on the new Rattatat album between a 320 Kbps rip and the wav - I could hear the difference fairly cleary, but it seemed to me that the mp3 was slightly louder, like maybe 0.5db RMS or less. I know the encoding can give you clips by changing the transients, but does it tend to boost RMS as well?

Using the lame encoder, my experience have been the opposite. RMS tends to measure a tenth or two of a dB lower than the original, while the peaks measure higher.



Regards,

Andreas Nordenstam


Disclaimer: I'm just a learner in mastering. Am thankful if someone more proficient will correct my mistakes if needed!

[greggybud]

Originally Posted by Lupo

try phase reversing one of the channels so you only listen to the difference product between the original and chrunched file. Should give you a clear indication of what to listen for in the Mp3. Creaky highend, broken glass" artifacts, flanging/phasing sensations.

Regards,

Andreas Nordenstam

Thats a wealth of information Andreas....Thanks. It would be hard to get accurate statistics, but at some point, and maybe we have already reached it, more people are listening to MP3 than an audio CD. If that is the case, then I would guess more attention has to be focused on final MP3's regardless of the coder process?

With Wavelab you dont even have to phase reverse. Instead you can do a file comparison and then play the difference.

My next question is why I can convert a loud pop song (maybe -9dbRMS) to MP3 in Wavelab, then upload it to Myspace, and it will still sound relatively quiet. And yes I have tried a non-mastered (-14dbRMS) with poor (not loud) results.

Can anyone share the secret to obtaining a loud song on Myspace?

[PhilRanger]

You can change your encoder settings, but usually everything above 16 kHz is dropped in mp3 encoding. Subtle effects as reverb trails, rooms, etc. often get lost. Listen to hihats, cymbals and voices on mp3s. The high frequencies are wobbled and flangy...

[fak0u]

Originally Posted by greggybud

Can anyone share the secret to obtaining a loud song on Myspace?

do not use myspace own player
it is slow, buggy, and alters the sound so much !

put your own flash MP3 player that will stream directly from your home page an mp3 encoded at your will

it will sound better, look better, and work better

[greggybud]

Originally Posted by fak0u

do not use myspace own player
it is slow, buggy, and alters the sound so much !

put your own flash MP3 player that will stream directly from your home page an mp3 encoded at your will

it will sound better, look better, and work better

That may well be true. But Im noticing a lot of loud mixes on the myspace player and they usually come from well known commercial artists.

When I check the .wav loudness of their songs it's around -10dbRMS. Mine are about the same loudness. But when played on the myspace player theirs sound much louder than mine. So my conclusion is they use a different type of coder, or the settings on my coder i.e Wavelab is not set correctly.

Im not even sure Wavelab gives any coder options and Im away from home so I cant check.

[MASSIVE Master]

I can show you file after file at -12dBRMS that are *apparently louder* than file after file at -10dBRMS. The number doesn't mean all that much.

And generally great sounding mixes done by teams of proffesionals at every single stage in the game working with core sounds that are tweaked for hours - days before the "RECORD" button is ever pressed for the first time - It's not a big surprise when that kind of quality sounds better @ louder.

I'm not sure what your experience / gear / capabilities are - But the vast, vast majority of mixes will *never* have the "loudness potential" as those that can handle that type of abuse.

Personally, I wish *none* of them would... But that's for another thread.

[greggybud]

Originally Posted by MASSIVE Master

I can show you file after file at -12dBRMS that are *apparently louder* than file after file at -10dBRMS. The number doesn't mean all that much.

And generally great sounding mixes done by teams of proffesionals at every single stage in the game working with core sounds that are tweaked for hours - days before the "RECORD" button is ever pressed for the first time - It's not a big surprise when that kind of quality sounds better @ louder.

I'm not sure what your experience / gear / capabilities are - But the vast, vast majority of mixes will *never* have the "loudness potential" as those that can handle that type of abuse.

Personally, I wish *none* of them would... But that's for another thread.

Of course the number doesn't mean much. One file could be nothing but a big loud wall-of-sound organ pad with no dynamics, and the other could sound louder but have a quiet bridge in the middle of the song that will pull the average dbRMS way down.

Great sounding mixes are done by anyone who has experience applying proper tools to achieve the desired sonic objectives in any given circumstance. Sometimes it might be compression in series, parallel, and in combination with EQ and maybe M/S tweaks. Sometimes it is best achieved by surgical compression/limiting where you are just focusing on certain aspects of a track. Sometimes it is treading lightly allowing "overs" to just exist depending on number of samples. And once in a while, if you get lucky it is doing very little.

But I cant help but wonder if there is a special coder that works better with the defualt myspace player?

[Curtis Franklin]

do you think that myspace allows better quality mp3's for popular bands?

i will do some research.

[stellar]

Originally Posted by pumadrum

do you think that myspace allows better quality mp3's for popular bands?

i will do some research.

yeah you can upload whatever bitrate you want (though not sure about variable?) the problem with using higher bitrates on myspace is that the load times are really bad, and often they don't load at all. quicker load times = more plays. sucks dont it? though I would like to know about variable bit rates and how myspace handles that.

[Curtis Franklin]

wouldnt it be a situation where you audition different codecs at different qualities and choose accordingly?

does it not sound the same as the file that you upload?

if it doesnt sound how you want, why don't you just make it sound like what you want.

[treymonfauntre]

Originally Posted by stellar

yeah you can upload whatever bitrate you want (though not sure about variable?) the problem with using higher bitrates on myspace is that the load times are really bad, and often they don't load at all. quicker load times = more plays. sucks dont it? though I would like to know about variable bit rates and how myspace handles that.

are you sure? as far as i know myspace has always transcoded everything to 96kbps. in the past you were able to download the original source file, but nowadays all you can download is the 96kbps one..

[Watersound]

Andreas, any relation to Stina Nordenstam? She is a great artist.


John

[Nordenstam]

Hi!

Originally Posted by greggybud

It would be hard to get accurate statistics, but at some point, and maybe we have already reached it, more people are listening to MP3 than an audio CD. If that is the case, then I would guess more attention has to be focused on final MP3's regardless of the coder process?

Think you're right, at least if counting all the lossy coding in the radio and satellite transmissions. Let's hope it gets more attention! Most of the issues seems to be identical to the technical level problems in the CD loudness wars. Both lossy coded and CD format could easily be safeguarded against overs if general levels could drop a few dB's. With todays gear, a global ceiling of -6dB sample peak level would not hurt performance significantly. Quite the contrary in all the various digital processing that may happen to the signal, before it gets converted at the end users.

As for now the current CD mastering trends demands of the playback equipment to deliver a few dB's above zero. This problem usually gets less obvious the better the converter equipment is. An oversampled peak meter is a great tool for this purpose, along with an oversampled limiter if needed(, or, the sony limiter). Gives full control of the real intersample peaks.

Originally Posted by John

Andreas, any relation to Stina Nordenstam? She is a great artist.

Unfortunately not. She is! =)


Andreas

[lucey]

Originally Posted by greggybud

1. As a standard procedure, do you as a mastering engineer convert each song to MP3 and listnen?

As MP3's become more popular does it warrant it's own mastering? What I mean is if you go to any major commercial download site, have those MP3's been altered in any way other than going through a standard conversion just like I would when I convert .wav to MP3 with Wavelab?

The whole point of mastering is that the product sounds best on all playback systems, and even though mp3 does change the sound it's still under the heading of "all playback systems".

If you dont hear what format compression is doing you are missing something for sure, it's very audible, but it doesn't require a separate mastering pass.

The only reason to master things two ways would be for more dynamics vs. less. thumbsup

[stellar]

Originally Posted by treymonfauntre

are you sure? as far as i know myspace has always transcoded everything to 96kbps. in the past you were able to download the original source file, but nowadays all you can download is the 96kbps one..

no, i'm not sure. all I know is a couple years ago when I was uploading to myspace the high quality mp3's wouldn't load and I had to upload some crappier ones. instant success!