So how many people actually purchase mp3's?
Re: So how many people actually purchase mp3's?
In my case RAID 0 was the only option because there was not enough space otherwise. I have had months of enjoyment from my RAID 0. My backup will not be "on the shelf" for 2 years and it is not a similar drive. I plan to invest in a terabyte drive in the near future as well. Don't listen to astro people, always backup. DVD media is pennies per gigabyte.
Re: So how many people actually purchase mp3's?
hey Brain, I did not write '...don't ever backup...' but ...your backup may be less reliable than you think... 
cheers, Tom
ps: one for your personal amusement
You can synchronize your digital pictures (via iTunes) with the iphone and carry quite an amount with you...
people considered this an additional level of 'safety' (which it obviously is - different medium, different location)
Some of those folks in fact lost their harddisks, and happily synced back from the mobile phone...
Only to realize what iTunes really had meant with 'optimizing' ... reducing resolution down to 25% and blowing up data volume by converting jpgs to .bmp
cheers, Tom
ps: one for your personal amusement
You can synchronize your digital pictures (via iTunes) with the iphone and carry quite an amount with you...
people considered this an additional level of 'safety' (which it obviously is - different medium, different location)
Some of those folks in fact lost their harddisks, and happily synced back from the mobile phone...
Only to realize what iTunes really had meant with 'optimizing' ... reducing resolution down to 25% and blowing up data volume by converting jpgs to .bmp
Re: So how many people actually purchase mp3's?
braincell, the man who has perfected doublethink. the man who denied cloud computing months ago but now loves it. you'll like it even better once the neural interface is in place and you can live totally in your head. who cares what plans they have for your body? 
Re: So how many people actually purchase mp3's?
You should back them up so you don't have to worry. I think it's affecting your mood.
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irrelevance
Re: So how many people actually purchase mp3's?
Just checking out Beatport and I would love someone to explain what a WAV handling charge is? What exactly is there to "handle" and why would it cost and extra £1 
This is like the now infamous bank admin fee which most got wise to in the uk a few years back and as a consequence demanded these charges back through the courts. Worked for me back then anyhow 
Oh I've actually purchased a few mp3's via Klicktrack with no fuss as I couldn't find wav anywhere.
What exactly is there to "handle" and why would it cost and extra £1 This is like the now infamous bank admin fee which most got wise to in the uk a few years back and as a consequence demanded these charges back through the courts. Worked for me back then anyhow Oh I've actually purchased a few mp3's via Klicktrack with no fuss as I couldn't find wav anywhere.
Re: So how many people actually purchase mp3's?
What would be funny is if Beatport was converting the mp3's back to WAV...hence the 'handling fee'. Hopefully not! It took SO long to talk them into 320k mp3's (they were 192k at the time) that I've given up on VBR/FLAC. It is nice they have MP4/AAC but you would think they would elect to do ALAC/FLAC instead of WAV for lossless delivery. 1/5th to 1/10th the filesize would save some bandwidth...
Also beware that beatport has 'elected' to use a newer id3v2 tagging method than either windows or iTunes/OS X support natively. Supposedly they needed to do this for compatibility with their backend system changes, but I find it interesting that they've elected to make it possible to corrupt your mp3's with applications many will have installed by default. Especially given their 4 hour download window they provide, with no window to redownload. I've had Serato & Traktor occasionally mess up their mp3's too, though never foobar2000 or tagging tools like mp3tag.
Also beware that beatport has 'elected' to use a newer id3v2 tagging method than either windows or iTunes/OS X support natively. Supposedly they needed to do this for compatibility with their backend system changes, but I find it interesting that they've elected to make it possible to corrupt your mp3's with applications many will have installed by default. Especially given their 4 hour download window they provide, with no window to redownload. I've had Serato & Traktor occasionally mess up their mp3's too, though never foobar2000 or tagging tools like mp3tag.
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irrelevance
Re: So how many people actually purchase mp3's?
It took SO long to talk them into 320k mp3's (they were 192k at the time) that I've given up on VBR/FLAC. It is nice they have MP4/AAC but you would think they would elect to do ALAC/FLAC instead of WAV for lossless delivery. 1/5th to 1/10th the filesize would save some bandwidth...valis wrote:What would be funny is if Beatport was converting the mp3's back to WAV...hence the 'handling fee'. Hopefully not!
No that would not be funny!!!
[/quote]
I think I will send an enquiry as to why this fee exists and see what wild answer they come up with. Your point about flac and bandwidth is spot on and bandwidth is the only reason I can think why they may charge more. However there is an arguement that converting to flac would cost just as much. Beatport does offer mp4 but at 192k not 320. I have read that mp4 performs better at lower bit rates compared to mp3 but I have tested this myself but could this be the reasoning for beatports offering?
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irrelevance
Re: So how many people actually purchase mp3's?
Well I didn't make it past the first few posts in that winamp thread, but this is my take:
With Lossless codecs (FLAC, ALAC, SLS, ALS, Musepack, monkey audio etc) you'll get out what you put in. So obviously these have the capacity for the 'best' audio quality.
Mpeg4 AAC tends to sound 'as good as' mp3 at about half the bitrate of mp3, by virtue of using significantly better technology (which was too computationally intensive to be considered useful when--mp3Mpeg1 Layer3--specs were being developed.)
Mp3 is a compromise that's gained acceptance due to being the first format in wide-use. As the encoders have evolved they've improved their ability to mask & suppress some of the downsides of the mp3 compression scheme (more on this below). Using either a modern fraunhofer codec or current Lame VBR modes you should be able to achieve acceptable quality at higher bitrates & settings.
Ogg Vorbis comes in below both of these in my book, which is where I'll start...
With Lossless codecs (FLAC, ALAC, SLS, ALS, Musepack, monkey audio etc) you'll get out what you put in. So obviously these have the capacity for the 'best' audio quality.
Mpeg4 AAC tends to sound 'as good as' mp3 at about half the bitrate of mp3, by virtue of using significantly better technology (which was too computationally intensive to be considered useful when--mp3Mpeg1 Layer3--specs were being developed.)
Mp3 is a compromise that's gained acceptance due to being the first format in wide-use. As the encoders have evolved they've improved their ability to mask & suppress some of the downsides of the mp3 compression scheme (more on this below). Using either a modern fraunhofer codec or current Lame VBR modes you should be able to achieve acceptable quality at higher bitrates & settings.
Ogg Vorbis comes in below both of these in my book, which is where I'll start...
Re: So how many people actually purchase mp3's?
I'll go over more formats in a minute, but first let's start with Ogg Vorbis (Ogg is the 'container', vorbis is the audio codec standard.) Vorbis is nice in that it's unencumbered by patents (Open Source) but in my tests it's only on par with mp3's when using older mp3 codecs or using a codec for formats for which it isn't optimized (Fraunhofer is optimized for 320k CBR & higher pre-Mp3Pro, LAME--post 3.97.xx--is optimized for VBR formats with V0 to V2 being the 'standards' as per the presets). A few people I know at the BBC deride the Vorbis codec, but you'll find many Open Source supporters prefer it.
Now onto other codecs...
Most of the lossless codecs support up to 32bit files as source (INT or FLT depends on codec), though ALS/SLS (mpeg4 lossless & apple's derivative) seems to only support 24bit. Some encoders & decoders have a hard time with even 24bit files and higher samplerates, but most current codecs support up to at least 192khz (FLAC for instance supports linear sample rates from 1Hz - 655350Hz in 1Hz increments.) So lossless can have the best & worst aspects of whatever the source was...as it should decode to an identical bitstream.
When dealing with Mpeg4 audio codecs specifically, there are several types of 'default' audio encodings specified in the Mpeg4 spec. AAC (mpeg4 'part3' lossy audio compression) is the most commonly known Mpeg4 audio codec, but is actually 5 different 'flavors' or 'guidelines' encoding. I think that Sony may have also bifurcated the AAC format with its own variant as well.
AAC supports up to 192khz with most encoders & decoders, and the files do not have a 'bit-depth' but instead it's a block-based codec where each block decodes to 1024 time-domain samples but is actually stored in a transform (and unlike MP3 below, each frame stands alone and doesn't depend on other frames before it.) So AAC files can be decoded to any bit-depth, and will depend entirely on the decoder and output device (more on this later.) In fact it seems to follow that AAC files can be encoded from more than 16bits as well. I would assume most encodings are 16bit/44.1khz unless otherwise stated, though when combined with h.264 & surround formats you may have higher samplerates & bit-depths (and may also have other audio encoding layers for the surround codecs.)
Technically speaking, Mpeg4 is a set of standard 'methods' (ie, standards) that define how to package various types of media together (video, audio etc) but you also get support for things like VRML, XML and other object data. One of those methods is MPEG-4 Part 14, otherwise known as "MP4" or the ".mp4" file extensions. Mpeg4 part14 aka. ".mp4" is a 'wrapper' (containers) for whatever you choose to include inside (or alongside if you're just containing the reference.) Normally an MP4 will contain mpeg4 data from the other defined 'methods' within mpeg4, but it can contain whatever the maker chooses (including mp3 or even vorbis-encoded audio data). For 'mp4' and 'm4a' type audio files you'll typically find one of the AAC variants is used.
Apple uses a different extension for their 'Mpeg4 audio' files: M4A or ".m4a". M4A files are identical to ".mp4" files, though typically ".m4a" indicates that the file is non-protected, whether it's from Apple/iTunes or someone else (ie, from iTunes Plus). The older FairPlay DRM encoded iTunes format is M4P or ".m4p" and is always 'protected' (encumbered by DRM). Apple also has a lossless format ALAC, which I believe uses linear prediction encoding methods similar to FLAC.
Mpeg4 also has the lossless codec called ALS (".als" file extension for Mpeg4 Lossless). ALS, the mpeg4 lossless layer, uses an integer format with 32bits of data (first bit is assumed 1 and thus used to specify block type, either zero block or normal block aka. contains data.) Output formats seem to be typically anywhere from 16bit to 32bit depending on decoder based on what I've read, and I would assume the input format can be whatever the encoder allows. I've never used ALS so I don't know for sure.
ALAC (Apple lossless) format seems to be found in either 16bit or 24bit format, but I'm not an iTunes user so can't comment more. I also don't author iTunes content (at least right now) so can't say much there either. One would presume at least 24bit input is available, especially considering Apple's marketing as a cutting edge content production platform, but I would assume most encodings are 16bit/44.1khz unless otherwise stated.
Additionally MP3 or ".mp3" files are "MPEG1 audio Layer III" files, defined in the Mpeg1 ISO spec. By the original standard Mp3 files can use 32khz, 44.1khz & 48khz samplerates and bitrates between 32kbit/s up to 385kbits/s. Also the MP3 format is NOT 16bit (or 24bit etc)...
Mpeg1 audio Layer III defines 32bit frame headers but bit depth is NOT stored in the frame header. The output from an mp3 is generated from transform coefficient in the mp3, and the output bit-depth is entirely up to the decoder you use. 24bit decoders are available that implement replay-gain properly so you get full 24bit output scaled to your intended output gain. Many decoders that implement 24bit output also allow 32bit output, int or flt, depending on implementation (int would simply pad with 0's). And interestingly enough you can feed some mp3 encoders with a 24bit file (or even 32bit linear, but it will be converted to 32bit FLT) for improved encoding dynamics, especially percussion (this will depend on encoding options & encoder tuning!)
Mp3's major sound quality compromises come not from the just the lossy heuristic algorithm (derived from ASPEC) but also from the bitstream compatibility with Mpeg1 Layer 2. The short answer is that the mp3 frames are adapted to fit into the mpeg1/layer2 standard's transport and this results in the aliasing, smearing & pre-echo when decoded (a result of the hybrid time domain (filter bank) / frequency domain (MDCT) model). Hydrogenaudio forums has more info on this than I could ever absorb myself, so feel free to defer there for more info.
Now that's what can be encoded & decoded, but you also have to consider the output stream (from media player to soundcard)...
ASIO will depend on soundcard maker, and most should be relatively familiar with it here.
Directsound output will usually not take more than 16bits within the AC97 spec (16bits up to 48khz) and is usually fixed @ 48khz with a SRC algorithm always-on to handle lower samplerate data. This means when using directsound drivers (or WAV drivers that do not bypass the OS's directaudio/directsound stack) you are ALWAYS hearing the SRC even with material where the source is the same as the output format (ie, 48khz>48khz). This results in a 'clouding'/smearing of the audio material & noticeable shift in the phase response (ripple in the frequency domain). The newer "HDAudio" spec allows up to 24bits/192khz (96khz is more typical) and I don't know all the details on the sound stack in Vista & Win7. Something tells me that they probably do 24bit/96khz 'all the time' (inline SRC again) to account for modern HD video formats, but I haven't coded directsound recently to know what is actually exposed. Most soundcards that support "HDAudio" tend to be consumer affairs, with horribly cheap converters & poor frequency response/time response.
Obviously with WIndows ASIO is usually your best bet (note that my RME's WAV drivers seem to bypass the directsound stack even on XP and are actually bit-transparent, and I suspect that Scope's may be the same.)
Core audio on mac is something that's supposed to be bit transparent & pain-free for the end user and professional alike, though I could probably dig on Apple's site a bit then Hydrogenaudio forums and find a few niggles here & there. I would assume the biggest concern here is when using a Mac's onboard soundcard (and for 3rd party soundcards defer to their literature.) I've never noticed any issues with my RME under OSX, files are bit-transparent as long as I don't adjust gain on the soundcard's routing interface from nominal (0dBfs change.)
And portable media players? There are some that fare relatively well in listening tests (especially older/larger devices) but iPods and the like are NOT for critical listening, they're for convenience. Use whatever formats & compression settings sound transparent 'enough' to you when listening with these players, which is usually lower than what you would choose for studio monitors & headphones (which means you can get by with smaller files if you encode for your portable player directly.)
Now onto other codecs...
Most of the lossless codecs support up to 32bit files as source (INT or FLT depends on codec), though ALS/SLS (mpeg4 lossless & apple's derivative) seems to only support 24bit. Some encoders & decoders have a hard time with even 24bit files and higher samplerates, but most current codecs support up to at least 192khz (FLAC for instance supports linear sample rates from 1Hz - 655350Hz in 1Hz increments.) So lossless can have the best & worst aspects of whatever the source was...as it should decode to an identical bitstream.
When dealing with Mpeg4 audio codecs specifically, there are several types of 'default' audio encodings specified in the Mpeg4 spec. AAC (mpeg4 'part3' lossy audio compression) is the most commonly known Mpeg4 audio codec, but is actually 5 different 'flavors' or 'guidelines' encoding. I think that Sony may have also bifurcated the AAC format with its own variant as well.
AAC supports up to 192khz with most encoders & decoders, and the files do not have a 'bit-depth' but instead it's a block-based codec where each block decodes to 1024 time-domain samples but is actually stored in a transform (and unlike MP3 below, each frame stands alone and doesn't depend on other frames before it.) So AAC files can be decoded to any bit-depth, and will depend entirely on the decoder and output device (more on this later.) In fact it seems to follow that AAC files can be encoded from more than 16bits as well. I would assume most encodings are 16bit/44.1khz unless otherwise stated, though when combined with h.264 & surround formats you may have higher samplerates & bit-depths (and may also have other audio encoding layers for the surround codecs.)
Technically speaking, Mpeg4 is a set of standard 'methods' (ie, standards) that define how to package various types of media together (video, audio etc) but you also get support for things like VRML, XML and other object data. One of those methods is MPEG-4 Part 14, otherwise known as "MP4" or the ".mp4" file extensions. Mpeg4 part14 aka. ".mp4" is a 'wrapper' (containers) for whatever you choose to include inside (or alongside if you're just containing the reference.) Normally an MP4 will contain mpeg4 data from the other defined 'methods' within mpeg4, but it can contain whatever the maker chooses (including mp3 or even vorbis-encoded audio data). For 'mp4' and 'm4a' type audio files you'll typically find one of the AAC variants is used.
Apple uses a different extension for their 'Mpeg4 audio' files: M4A or ".m4a". M4A files are identical to ".mp4" files, though typically ".m4a" indicates that the file is non-protected, whether it's from Apple/iTunes or someone else (ie, from iTunes Plus). The older FairPlay DRM encoded iTunes format is M4P or ".m4p" and is always 'protected' (encumbered by DRM). Apple also has a lossless format ALAC, which I believe uses linear prediction encoding methods similar to FLAC.
Mpeg4 also has the lossless codec called ALS (".als" file extension for Mpeg4 Lossless). ALS, the mpeg4 lossless layer, uses an integer format with 32bits of data (first bit is assumed 1 and thus used to specify block type, either zero block or normal block aka. contains data.) Output formats seem to be typically anywhere from 16bit to 32bit depending on decoder based on what I've read, and I would assume the input format can be whatever the encoder allows. I've never used ALS so I don't know for sure.
ALAC (Apple lossless) format seems to be found in either 16bit or 24bit format, but I'm not an iTunes user so can't comment more. I also don't author iTunes content (at least right now) so can't say much there either. One would presume at least 24bit input is available, especially considering Apple's marketing as a cutting edge content production platform, but I would assume most encodings are 16bit/44.1khz unless otherwise stated.
Additionally MP3 or ".mp3" files are "MPEG1 audio Layer III" files, defined in the Mpeg1 ISO spec. By the original standard Mp3 files can use 32khz, 44.1khz & 48khz samplerates and bitrates between 32kbit/s up to 385kbits/s. Also the MP3 format is NOT 16bit (or 24bit etc)...
Mpeg1 audio Layer III defines 32bit frame headers but bit depth is NOT stored in the frame header. The output from an mp3 is generated from transform coefficient in the mp3, and the output bit-depth is entirely up to the decoder you use. 24bit decoders are available that implement replay-gain properly so you get full 24bit output scaled to your intended output gain. Many decoders that implement 24bit output also allow 32bit output, int or flt, depending on implementation (int would simply pad with 0's). And interestingly enough you can feed some mp3 encoders with a 24bit file (or even 32bit linear, but it will be converted to 32bit FLT) for improved encoding dynamics, especially percussion (this will depend on encoding options & encoder tuning!)
Mp3's major sound quality compromises come not from the just the lossy heuristic algorithm (derived from ASPEC) but also from the bitstream compatibility with Mpeg1 Layer 2. The short answer is that the mp3 frames are adapted to fit into the mpeg1/layer2 standard's transport and this results in the aliasing, smearing & pre-echo when decoded (a result of the hybrid time domain (filter bank) / frequency domain (MDCT) model). Hydrogenaudio forums has more info on this than I could ever absorb myself, so feel free to defer there for more info.
Now that's what can be encoded & decoded, but you also have to consider the output stream (from media player to soundcard)...
ASIO will depend on soundcard maker, and most should be relatively familiar with it here.
Directsound output will usually not take more than 16bits within the AC97 spec (16bits up to 48khz) and is usually fixed @ 48khz with a SRC algorithm always-on to handle lower samplerate data. This means when using directsound drivers (or WAV drivers that do not bypass the OS's directaudio/directsound stack) you are ALWAYS hearing the SRC even with material where the source is the same as the output format (ie, 48khz>48khz). This results in a 'clouding'/smearing of the audio material & noticeable shift in the phase response (ripple in the frequency domain). The newer "HDAudio" spec allows up to 24bits/192khz (96khz is more typical) and I don't know all the details on the sound stack in Vista & Win7. Something tells me that they probably do 24bit/96khz 'all the time' (inline SRC again) to account for modern HD video formats, but I haven't coded directsound recently to know what is actually exposed. Most soundcards that support "HDAudio" tend to be consumer affairs, with horribly cheap converters & poor frequency response/time response.
Obviously with WIndows ASIO is usually your best bet (note that my RME's WAV drivers seem to bypass the directsound stack even on XP and are actually bit-transparent, and I suspect that Scope's may be the same.)
Core audio on mac is something that's supposed to be bit transparent & pain-free for the end user and professional alike, though I could probably dig on Apple's site a bit then Hydrogenaudio forums and find a few niggles here & there. I would assume the biggest concern here is when using a Mac's onboard soundcard (and for 3rd party soundcards defer to their literature.) I've never noticed any issues with my RME under OSX, files are bit-transparent as long as I don't adjust gain on the soundcard's routing interface from nominal (0dBfs change.)
And portable media players? There are some that fare relatively well in listening tests (especially older/larger devices) but iPods and the like are NOT for critical listening, they're for convenience. Use whatever formats & compression settings sound transparent 'enough' to you when listening with these players, which is usually lower than what you would choose for studio monitors & headphones (which means you can get by with smaller files if you encode for your portable player directly.)
Re: So how many people actually purchase mp3's?
Sorry for killin the thread with a text-barf. 
Re: So how many people actually purchase mp3's?
did you try powercore or uad?braincell wrote:Other companies let the customer transfer keys but somehow this wasn't possible for them
when I got my first poco card it was a hassle with manual intervention by tc stuff to just transfer the card. it took several days. the tc surround plugin the box came with revealed as a beta that was never released and wasn't even available anymore, although I own the original cd (which contains... um well some kinda pre-installer not including the real plugin).
you could point me to the fact that I wasn't charged for that process, but then... I'm still curious whether you can transfer plugins on poco at all. hints, anyone?
afaik you cannot transfer uad plugins as well. once you bought them they are bound to your card or group of cards. you can sell the card holding the plugins and be left off with a card having stock plugins (if you own more than one card) but you can only "get rid of" all your extra plugins or none. that's the way it is. no charge for it, of course.
btw, afair creamware never intended plugin transfer. that was just their policy back in thoise days, and everybody not being ok with that could just refrain from buying it. they made plugin transfer possible after numerous requests, and they decided to charge for it after it was used more frequently. they might make some people feel uncomfortable or even being ripped off, but they make more people feel like their investions were well protected by this kinda drm or whatever you feel like calling it.
apart from that I enjoy buying cds (and hope my house won't burn down
), ripping them into mp3 to carry around with my brand new personal stereo (and having a mobile backup ), using raid1 data storage and external disks for backup. I gave up the idea of backup to tape the moment my amount of data was bigger than 3 qic80 tapes and because it took all night to complete. I still put my digital photos to 5.25" MO every now and then, for extra protection. yeah, I think I didn't forget anything... -greetings, markus-
ps. sorry for not replying in just-in-time. I was on vacation.
--
I'm sorry, but my karma just ran over your dogma.
I'm sorry, but my karma just ran over your dogma.
Re: So how many people actually purchase mp3's?
Hopefully downloads will lead to a higher quality standard for audio masters. I don't like 16 bit 44.1 sample rate. This is a very poor level of fidelity and a low dynamic range. It was designed decades ago. Of course the Luddites will always disagree. The CD is a bad bad format.
Re: So how many people actually purchase mp3's?
well, the CD format specs cover way more than what's needed for the sh*t most top 40 producers release.
... from their 32bit 192 khz $$$ studios
cheers, Tom
... from their 32bit 192 khz $$$ studios
cheers, Tom
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netguyjoel
- Posts: 1228
- Joined: Sat Apr 11, 2009 9:34 am
- Location: The Land of Cheese, Beer & Fat Chicks
Re: So how many people actually purchase mp3's?
Not to mention...it ALL gets knocked down to 16 bit 44.1... the human ear (in most cases) is not sophisticated or trained to even know the difference...
the human ear (in most cases) is not sophisticated or trained to even know the difference... Joel
Re: So how many people actually purchase mp3's?
astroman wrote:well, the CD format specs cover way more than what's needed for the sh*t most top 40 producers release.
... from their 32bit 192 khz $$$ studios
cheers, Tom
Re: So how many people actually purchase mp3's?
Good website for high resolution audio.The stuff on it is free so give it a try.I was surprised when i heard it,quite a difference.
http://www.2l.no/hires/index.html
Edit: The required codec in on the site too.
http://www.2l.no/hires/index.html
Edit: The required codec in on the site too.
Re: So how many people actually purchase mp3's?
Nice link fluxpod