Hi,
I think the question misses the main point, and that is, that none of both mediums are capable of transporting the quality of the master. If you've ever heard a master tape, it becomes a bit irrelevant to make that comparison CD vs. LP vs. MC vs. whatever.
I can't help believing that it has to be that way -- imagine the pirating possibilities with perfect copies....
Rüdiger
I think the question misses the main point, and that is, that none of both mediums are capable of transporting the quality of the master. If you've ever heard a master tape, it becomes a bit irrelevant to make that comparison CD vs. LP vs. MC vs. whatever.
I can't help believing that it has to be that way -- imagine the pirating possibilities with perfect copies....
Rüdiger
Rüdiger.
Unfortunately, nobody has access to master tapes.
I am well aware of the mostly better quality they provide.
IMO the only way for the labels to protect their properties, to protect their treasures. 😀
Do you believe we would ever see a digital 32bit master copy
for download somewhere, somtime in the future - forget it. This wil never happen, not in my audio-life span.
Let's discuss things which are available.
Cheers
Unfortunately, nobody has access to master tapes.
I am well aware of the mostly better quality they provide.
IMO the only way for the labels to protect their properties, to protect their treasures. 😀
Do you believe we would ever see a digital 32bit master copy
for download somewhere, somtime in the future - forget it. This wil never happen, not in my audio-life span.
Let's discuss things which are available.
Cheers
Why would we need 32 bits? A true 24-bit DAC would be ultra hi-rez. Heck, the 16-bit Redbook standard is pretty much hi-rez. Does anyone even know how a true 16-bit system sounds like? Most CDs don't even have 14 bits!
Netlist said:
Short answer: I can't. It was a general response to the mad number's game.
I got it from the Wadia site. Since I had read that 24-bit oversampling DAC chips have 19 real-world bits, it kind of made sense. At least it made it easier to understand why the budget 12-13 bit TDA1543 can sound as good as it does.
The current benchmark for oversampling DAC chips is 21 bits. Maybe the same is true for CDs as well.
Maybe the gurus here can enlighten us.
out on a limb
I have a feeling that what you're talking about is the usable range vs the total ability of the devices. If you have a 24 foot ladder and you put in a five foot deep hole the usable range (to move you higher than you already are) is 19 ft. It is much the same with digital devices, since as they create dynamic range, most of that is extended towards the noise floor. If your recording has a high noise quotient, then although the parts in the middle may have shorter "steps" much of the difference is "below ground" or at very low levels that you may or may not be able to hear depending on your source and playback device.
I can however say without a doubt, that barring already recorded program, where a conversion of bitdepths would have little effect, and recording live (such as a record which theoretically has an infinite amount of data since it's analog) or better yet a mic, that upsampling makes a huge difference in what you hear. The higher the bitdepth and the faster the sample rate, the more "realistic" the result.
I tend to think of this in terms of slew rate as an analog analogy, the higher the slew rate of the device, the faster it responds, and the more realistic the sound, which is why tubes are still considered (by some) the best sounding audio devices, and that better chips (with higher slew rates) make better sounding audio.
I would be interested in a link to the site or sites so that I could peruse the data of which you speak, and will be looking around today to see if I got this right.
never be afraid to express and opinion even if it's wrong.
I have a feeling that what you're talking about is the usable range vs the total ability of the devices. If you have a 24 foot ladder and you put in a five foot deep hole the usable range (to move you higher than you already are) is 19 ft. It is much the same with digital devices, since as they create dynamic range, most of that is extended towards the noise floor. If your recording has a high noise quotient, then although the parts in the middle may have shorter "steps" much of the difference is "below ground" or at very low levels that you may or may not be able to hear depending on your source and playback device.
I can however say without a doubt, that barring already recorded program, where a conversion of bitdepths would have little effect, and recording live (such as a record which theoretically has an infinite amount of data since it's analog) or better yet a mic, that upsampling makes a huge difference in what you hear. The higher the bitdepth and the faster the sample rate, the more "realistic" the result.
I tend to think of this in terms of slew rate as an analog analogy, the higher the slew rate of the device, the faster it responds, and the more realistic the sound, which is why tubes are still considered (by some) the best sounding audio devices, and that better chips (with higher slew rates) make better sounding audio.
I would be interested in a link to the site or sites so that I could peruse the data of which you speak, and will be looking around today to see if I got this right.
never be afraid to express and opinion even if it's wrong.
just a bit more
most recordings today use huge amounts of compression, especially CD's since the idea is to make it as loud as possible, at a sacrifice of dynamic range. Many of the "modern" cd's I have looked at use less than 20db of dynamic range since the signal has been so compressed to increase the "apparent" loudness. (think tv commercials)
This would explain why a cheap dac could sound as good as an expensive one, since most of the information is towards the upper ends of the scale. It may also be some of the reasoning behind the amounts of compression used, since it will sound about the same on any device that it plays on, regardless of the quality of the device.
Many older cd's used more of the dynamic range, but suffered from very "quiet" playback levels, ie. Dire Straits "private investigations" which uses the entire dynamic range available, but is mostly inaudible unless you have a fantastic playback rig. By compressing the drum and loud, crashing parts of the track, you can actually make out the vocals. If you were to play another tune (say anything recent from madonna) that used compression after adjusting the volume to be able to hear the vocals, you would be deafened, or blow up your stuff. This is the difficulty with audiophile reproduction today, the lowest common denominator sets the stage for the rest.
If you download audacity or any other recording program and examine and compare the waveforms of two of the kinds of tracks that I describe, you'll see the difference. You can always increase gain and compression on the softer track, but unfortunately you can't go the other way with the louder one.
wise men often learn from their own mistakes, but the truly wise learn from the mistakes of others.
most recordings today use huge amounts of compression, especially CD's since the idea is to make it as loud as possible, at a sacrifice of dynamic range. Many of the "modern" cd's I have looked at use less than 20db of dynamic range since the signal has been so compressed to increase the "apparent" loudness. (think tv commercials)
This would explain why a cheap dac could sound as good as an expensive one, since most of the information is towards the upper ends of the scale. It may also be some of the reasoning behind the amounts of compression used, since it will sound about the same on any device that it plays on, regardless of the quality of the device.
Many older cd's used more of the dynamic range, but suffered from very "quiet" playback levels, ie. Dire Straits "private investigations" which uses the entire dynamic range available, but is mostly inaudible unless you have a fantastic playback rig. By compressing the drum and loud, crashing parts of the track, you can actually make out the vocals. If you were to play another tune (say anything recent from madonna) that used compression after adjusting the volume to be able to hear the vocals, you would be deafened, or blow up your stuff. This is the difficulty with audiophile reproduction today, the lowest common denominator sets the stage for the rest.
If you download audacity or any other recording program and examine and compare the waveforms of two of the kinds of tracks that I describe, you'll see the difference. You can always increase gain and compression on the softer track, but unfortunately you can't go the other way with the louder one.
wise men often learn from their own mistakes, but the truly wise learn from the mistakes of others.
I never shut up
went to the wadia site, interesting stuff, but I can't really listen to the results of their efforts. In theory they are adding bits which may or may not be where they would be if they were recorded in other words, they're taking an educated guess at where additional information would be. All of this in an effort to use a higher bitdepth device to reproduce the sound.
I have to chuckle at the "direct output" device or whatever they called it to replace your "analog circuitry" since it's an analog device anyway (it has to be), what it does do is makes you switch between your devices by moving wires. You could probably drive your amp with the output of a cd player or tapedeck or ipod anyway, unless it's a piece of crap or your amp is 20 ft away from the deck.
In the end, audio is subjective to some degree, although there is alot that experts can agree on. Resampling of cd's doesn't make much sense to me, unless you are compressing, increasing gain or changing the original waveform in some way that will make it an easier/better playback experience. 16 bit resolution will always contain the same amount of data, no matter what. There are programs for instance to "upsample" mp3's to a wavefile (cdex), which would allow you to play them on your cd player. But they're still mp3's.
If it was me, and I wasn't happy with the sound of my playback rig, I would invest in the basics, beginning with an audiophile amp, then speakers, then some combination of dac/adc soundcard or cd player. Personally I believe in the computer method of things, since I can upgrade the software, and have many more options available for playback. (the downside is the noise of the computer, which can make a distinct hum or worse.) If you're using vinyl, a tube preamp and riaa curve device directly into the dac/adc soundcard would be the purest method. You could actually connect the phono pre directly to the amp, and you should, in order to see what your sound card/dac is doing.
In most cases simpler is better, and less is more. Buying a piece at a time can be frustrating, because with every increase in quality in one area, the shortcomings of the others are highlighted. Unless you're an engineer, redesigning your equipment usually is more work and expense than buying a new piece. I know this is DIY, but I believe that most are better off buying a kit, or building something from plans (like an phono pre, amp or speakers) than trying to "fix" a commercial digital device, which is several orders of magnitude more complicated, and "good" ones have become relatively cheap.
And of course invest in the highest sampled version available of the material, which in most cases would be a record.
went to the wadia site, interesting stuff, but I can't really listen to the results of their efforts. In theory they are adding bits which may or may not be where they would be if they were recorded in other words, they're taking an educated guess at where additional information would be. All of this in an effort to use a higher bitdepth device to reproduce the sound.
I have to chuckle at the "direct output" device or whatever they called it to replace your "analog circuitry" since it's an analog device anyway (it has to be), what it does do is makes you switch between your devices by moving wires. You could probably drive your amp with the output of a cd player or tapedeck or ipod anyway, unless it's a piece of crap or your amp is 20 ft away from the deck.
In the end, audio is subjective to some degree, although there is alot that experts can agree on. Resampling of cd's doesn't make much sense to me, unless you are compressing, increasing gain or changing the original waveform in some way that will make it an easier/better playback experience. 16 bit resolution will always contain the same amount of data, no matter what. There are programs for instance to "upsample" mp3's to a wavefile (cdex), which would allow you to play them on your cd player. But they're still mp3's.
If it was me, and I wasn't happy with the sound of my playback rig, I would invest in the basics, beginning with an audiophile amp, then speakers, then some combination of dac/adc soundcard or cd player. Personally I believe in the computer method of things, since I can upgrade the software, and have many more options available for playback. (the downside is the noise of the computer, which can make a distinct hum or worse.) If you're using vinyl, a tube preamp and riaa curve device directly into the dac/adc soundcard would be the purest method. You could actually connect the phono pre directly to the amp, and you should, in order to see what your sound card/dac is doing.
In most cases simpler is better, and less is more. Buying a piece at a time can be frustrating, because with every increase in quality in one area, the shortcomings of the others are highlighted. Unless you're an engineer, redesigning your equipment usually is more work and expense than buying a new piece. I know this is DIY, but I believe that most are better off buying a kit, or building something from plans (like an phono pre, amp or speakers) than trying to "fix" a commercial digital device, which is several orders of magnitude more complicated, and "good" ones have become relatively cheap.
And of course invest in the highest sampled version available of the material, which in most cases would be a record.
With real-world bits I meant "useful bits." Not that it matters. The nomenclature eludes me. I mix up up- and oversampling all the time. I don't even quite understand them or the difference. It's largely irrelevant in my case.
Maybe an elaboration on my post is in place. The CD format is a crappy format. It's built on antiquated technology, like the fixed bit-rate. BUT I don't see the 16/44.1 as the big problem. That doesn't mean I don't think higher isn't better. But there's that but again. It gets increasing more complex. And the question is, does this complexity make things sound better or just measure better? To sort of follow up on pesky's Wadia observations. Oversampling (or is it upsampling?) may be something of a requirement for a digital volume pot to be useful (it reduces sound by throwing away bits of data), what price are we willing to pay for the convenience of a digital volume pot?
Again, not that any of that matters. The CD is what we have and are going to have for a long time. In fact, I think the only hope for digital sound reproduction is for the silvery discs to go the way of the dinosaurs. The SACD and DVD-A are history. iTunes-type distribution is the last hope. Be the format mp4 or whatever. Compression will not be a factor. The day "everybody" has 100Mb connections and can download a 100MB file in roughly 16 sec and tiny flash memory based mp3 (or cell phone) players can store 10 times as much as they do today, file size will become a no-issue. Till then nothing will happen in digital sound reproduction.
It may still mean the end for DIY, like the SACD and DVD-A did. The 21st century looks to be the best of times and worst of times. In worst case scenario, 1997-2000 will represent the pinnacle of digital sound reproduction.
Maybe an elaboration on my post is in place. The CD format is a crappy format. It's built on antiquated technology, like the fixed bit-rate. BUT I don't see the 16/44.1 as the big problem. That doesn't mean I don't think higher isn't better. But there's that but again. It gets increasing more complex. And the question is, does this complexity make things sound better or just measure better? To sort of follow up on pesky's Wadia observations. Oversampling (or is it upsampling?) may be something of a requirement for a digital volume pot to be useful (it reduces sound by throwing away bits of data), what price are we willing to pay for the convenience of a digital volume pot?
Again, not that any of that matters. The CD is what we have and are going to have for a long time. In fact, I think the only hope for digital sound reproduction is for the silvery discs to go the way of the dinosaurs. The SACD and DVD-A are history. iTunes-type distribution is the last hope. Be the format mp4 or whatever. Compression will not be a factor. The day "everybody" has 100Mb connections and can download a 100MB file in roughly 16 sec and tiny flash memory based mp3 (or cell phone) players can store 10 times as much as they do today, file size will become a no-issue. Till then nothing will happen in digital sound reproduction.
It may still mean the end for DIY, like the SACD and DVD-A did. The 21st century looks to be the best of times and worst of times. In worst case scenario, 1997-2000 will represent the pinnacle of digital sound reproduction.
oversampling vs upsampling
oversampling has to do with the frequency that the material is "checked" for the data it needs, for the next byte. The increased frequency allows the actual data byte to fall exactly on the clock pulse, since the one or zero is very long in relation to the sample time, allowing for reduced jitter (as long as the clock is stable) since even if the byte isn't exactly positioned on the cd, it is already obvious to the cd player what the information is. At the precise moment of the clock pulse it uses the stored bits to create the analog stream. The legacy of 44.1 may have a lot to do with the amount of oversampling available at the time of the creation of the standard, as devices were slower then, I believe when the standard started 4X was considered pretty good, today I believe it's around 40x, so the accuracy of the final signal has improved greatly.
upsampling has to do with changing lower bit depth material (or lower sample rate material) to a higher standard. A cd is 16 bit, 44.1 khz,(or 1411.2 kbits/sec) mp3's can vary from 80 to 320 kbts/sec, with the higher rates sounding better. If I wanted to create a cd from an mp3 which was (for the sake of example here) sampled at 141kbits/sec , the cd would contain groups of identical 10 byte lengths of information (141 X 10=1410) to fill out the space in between the "real" bytes.
In the case of your volume control, material is "upsampled" in realtime by reading 2 bytes of data (16 bit words, actually 4 since it's stereo, but we'll talk about one channel here) from the cd or source material and average them to create data in between the two in order to make a smoother transistion. Its a little more complex than that, as the volume control throws away least significant bytes (those under the noise floor of the material) and fits its added "material" to the parts of the dynamic range that you are hearing the loudest, which is why the actual rate is something less than 24 bits. It just has to do with the amount of dynamic range available. You could also look at it like they are creating 3 extra bits of information, about the limit so that you don't hear any difference in the program, but enough to do something they can crow about. I haven't heard the device, so really can't comment on how successful they are.
one more thing bits are individual ones and zeros, bytes are groups of bits, usually 8, 16, 24, or 32.
happy listening
pesky
oversampling has to do with the frequency that the material is "checked" for the data it needs, for the next byte. The increased frequency allows the actual data byte to fall exactly on the clock pulse, since the one or zero is very long in relation to the sample time, allowing for reduced jitter (as long as the clock is stable) since even if the byte isn't exactly positioned on the cd, it is already obvious to the cd player what the information is. At the precise moment of the clock pulse it uses the stored bits to create the analog stream. The legacy of 44.1 may have a lot to do with the amount of oversampling available at the time of the creation of the standard, as devices were slower then, I believe when the standard started 4X was considered pretty good, today I believe it's around 40x, so the accuracy of the final signal has improved greatly.
upsampling has to do with changing lower bit depth material (or lower sample rate material) to a higher standard. A cd is 16 bit, 44.1 khz,(or 1411.2 kbits/sec) mp3's can vary from 80 to 320 kbts/sec, with the higher rates sounding better. If I wanted to create a cd from an mp3 which was (for the sake of example here) sampled at 141kbits/sec , the cd would contain groups of identical 10 byte lengths of information (141 X 10=1410) to fill out the space in between the "real" bytes.
In the case of your volume control, material is "upsampled" in realtime by reading 2 bytes of data (16 bit words, actually 4 since it's stereo, but we'll talk about one channel here) from the cd or source material and average them to create data in between the two in order to make a smoother transistion. Its a little more complex than that, as the volume control throws away least significant bytes (those under the noise floor of the material) and fits its added "material" to the parts of the dynamic range that you are hearing the loudest, which is why the actual rate is something less than 24 bits. It just has to do with the amount of dynamic range available. You could also look at it like they are creating 3 extra bits of information, about the limit so that you don't hear any difference in the program, but enough to do something they can crow about. I haven't heard the device, so really can't comment on how successful they are.
one more thing bits are individual ones and zeros, bytes are groups of bits, usually 8, 16, 24, or 32.
happy listening
pesky
Pesky,
much to my regret I have to say that you have this pretty much all wrong. Even more to my regret I don't have the time right now to elaborate on this. Maybe someone else can point you to the correct information?
much to my regret I have to say that you have this pretty much all wrong. Even more to my regret I don't have the time right now to elaborate on this. Maybe someone else can point you to the correct information?
not according to wikipedia
I may have simplified the working of the whole thing to some degree, but would appreciate some input on what I said that was wrong. Yea I know I suck and all, but at least I spent the time to "make a fool of myself"
correcting the mistakes of others is noble, telling someone they're wrong and walking away is an act of cowardice.
I may have simplified the working of the whole thing to some degree, but would appreciate some input on what I said that was wrong. Yea I know I suck and all, but at least I spent the time to "make a fool of myself"
correcting the mistakes of others is noble, telling someone they're wrong and walking away is an act of cowardice.
There's no need to shoot at me, I sincerely apologized in previous posting.
Still don't have much time, but here goes ...
1) upsampling = oversampling, in the context of digital audio playback
2) oversampling, in the context of digital audio playback,
is increasing the data rate from a lower sampling frequency to a hgher sampling frequency, by zero-padding, interpolation, whatever (the method is not very relevant), followed by steep low-pass filtering, in the digital domain, above half of the initial sampling frequency.
That's all there is, really.
As a side effect of this, the data word length is extended.
I wrote 'in the context of' as these terms have not been formally defined and may take on slightly different meanings when you talk about audio playback, audio recording, or signal theory in general.
Any further confusion in this has its origins in the creative copywriting of marketing depts trying to sell consumers 'new' technology and in the mostly abysmal understanding of digital audio and signal theory by the consumer audio press.
Still don't have much time, but here goes ...
1) upsampling = oversampling, in the context of digital audio playback
2) oversampling, in the context of digital audio playback,
is increasing the data rate from a lower sampling frequency to a hgher sampling frequency, by zero-padding, interpolation, whatever (the method is not very relevant), followed by steep low-pass filtering, in the digital domain, above half of the initial sampling frequency.
That's all there is, really.
As a side effect of this, the data word length is extended.
I wrote 'in the context of' as these terms have not been formally defined and may take on slightly different meanings when you talk about audio playback, audio recording, or signal theory in general.
Any further confusion in this has its origins in the creative copywriting of marketing depts trying to sell consumers 'new' technology and in the mostly abysmal understanding of digital audio and signal theory by the consumer audio press.
With the risk of messing things up again. 🙂
While I agree with Werner that there is no technical difference between oversampling and upsampling, I think the terms tend to be used for two different purposes also in audio playback. Correct me if I am wrong.
Oversampling usually refers to increasing the sampling frequency by an integer factor (usually on the form 2^k for some k). This can either be because the DAC has fewer than 16 bits and the purpose is then to "move precision" from the digital value domain to the time domain. For instance, most early CDPs had only 14 bit DACs, since 16 bit DACs were still very expensive. However, fast 14 bit DACs was still reasonably priced, so it was common to oversample by a factor 4 in an attempt to move the precision of the two least significant bits into the time domain. Nowadays oversampling is either used because 1-bit DACs are used or to move the signal up to a higher frequency band to allow for more effective digital filters.
Upsampling (and correspondingly downsampling) seems to be used when there is simply a need to convert a signal from one sampling frequency to another. For instance, most computer soundcards work with 48 kHz only (or a multiple of that, like 96 kHz). Hence, all 44.1 kHz signals, like wave files with ripped CDs must be converted to 48 kHz, which is done automatically by the soundcard and/or the soundcard drivers. Similarily, if recording a signal with a soundcard and storing the data as a 44.1 kHz wave file, the signal is usually AD converted in 48 kHz (or 96 or 192 kHz) and then downsampled to 44.1 kHz. That is, up-/downsampling is not commonly seen in CD players, although I think it existed in some DAT players to make digital recording of CDs possible (something that was probably considered economically infeasible when the DAT standard was conceived and thus considered a sufficient means of copy protection at the time).
While I agree with Werner that there is no technical difference between oversampling and upsampling, I think the terms tend to be used for two different purposes also in audio playback. Correct me if I am wrong.
Oversampling usually refers to increasing the sampling frequency by an integer factor (usually on the form 2^k for some k). This can either be because the DAC has fewer than 16 bits and the purpose is then to "move precision" from the digital value domain to the time domain. For instance, most early CDPs had only 14 bit DACs, since 16 bit DACs were still very expensive. However, fast 14 bit DACs was still reasonably priced, so it was common to oversample by a factor 4 in an attempt to move the precision of the two least significant bits into the time domain. Nowadays oversampling is either used because 1-bit DACs are used or to move the signal up to a higher frequency band to allow for more effective digital filters.
Upsampling (and correspondingly downsampling) seems to be used when there is simply a need to convert a signal from one sampling frequency to another. For instance, most computer soundcards work with 48 kHz only (or a multiple of that, like 96 kHz). Hence, all 44.1 kHz signals, like wave files with ripped CDs must be converted to 48 kHz, which is done automatically by the soundcard and/or the soundcard drivers. Similarily, if recording a signal with a soundcard and storing the data as a 44.1 kHz wave file, the signal is usually AD converted in 48 kHz (or 96 or 192 kHz) and then downsampled to 44.1 kHz. That is, up-/downsampling is not commonly seen in CD players, although I think it existed in some DAT players to make digital recording of CDs possible (something that was probably considered economically infeasible when the DAT standard was conceived and thus considered a sufficient means of copy protection at the time).
thanks for clairification
to werner sorry, when you said that I was COMPLETELY wrong I assumed (when you assume you make as *** of u and me) that you referred to my entire post. Since I do check myself (with the wikipedia, since it's easy) and it was a long series of posts, I was a little "miffed" that you didn't say what you were referring to. Apologies, and nice to "meet" you 🙂
This does however raise an interesting point, since you can oversample any digital audio signal (in order to perform some sort of function, whatever it is) but oversampling does not necessarily change anything, it just means that you are reading the signal at a faster rate than it was recorded at.
upsampling implies a change in one of the parameters of the signal, whether it's bitdepth or frequency, so there is some sort of mathematical operation performed as well.
any time there is a change in the signal, errors are present, and these add up to artifacts at some point (or not) depending on the strategy and actual method of the conversion.
the free exchange of ideas (and emotions) should never be impeded.
pesky
to werner sorry, when you said that I was COMPLETELY wrong I assumed (when you assume you make as *** of u and me) that you referred to my entire post. Since I do check myself (with the wikipedia, since it's easy) and it was a long series of posts, I was a little "miffed" that you didn't say what you were referring to. Apologies, and nice to "meet" you 🙂
This does however raise an interesting point, since you can oversample any digital audio signal (in order to perform some sort of function, whatever it is) but oversampling does not necessarily change anything, it just means that you are reading the signal at a faster rate than it was recorded at.
upsampling implies a change in one of the parameters of the signal, whether it's bitdepth or frequency, so there is some sort of mathematical operation performed as well.
any time there is a change in the signal, errors are present, and these add up to artifacts at some point (or not) depending on the strategy and actual method of the conversion.
the free exchange of ideas (and emotions) should never be impeded.
pesky
Re: thanks for clairification
No, in both cases you must add new samples, since you are going up to a higher sampling frequency. One way to do this is to compute new samples by some form of interpolation, which I think is necessary if the new sampling frequency is not an integer multiple of the source frequency. In the case it is an integer multiple, you can either compute the new samples by interpolation or let all the new samples have the value zero. In neither case are you adding any new information. The difference, I think, is that if you insert zeroes the digital filter has to do also the work corresponding to the interpolation you do in the other case.
pesky said:
No, in both cases you must add new samples, since you are going up to a higher sampling frequency. One way to do this is to compute new samples by some form of interpolation, which I think is necessary if the new sampling frequency is not an integer multiple of the source frequency. In the case it is an integer multiple, you can either compute the new samples by interpolation or let all the new samples have the value zero. In neither case are you adding any new information. The difference, I think, is that if you insert zeroes the digital filter has to do also the work corresponding to the interpolation you do in the other case.
Earlier in this thread we were refered to a 6 moons artical on burning CD's with EAC software. The artical said that when ripping and burning with EAC,best results would be gotten by using a dedicated SCSI drive and an outboard SCSI burner that is well damped.
Why is that??...SCSI stuff is hard to find...what if I just used a dedicated IDE hard drive and a good Burner?? SCSI is faster than IDE...I wonder if it would be worth all the effort to use SCSI components...
Why is that??...SCSI stuff is hard to find...what if I just used a dedicated IDE hard drive and a good Burner?? SCSI is faster than IDE...I wonder if it would be worth all the effort to use SCSI components...
I read the article at 6Moons briefly. IIRC they forgot to tell that it is important to match EAC with your burner to obtain the most exact results. As burning at low speeds is recommended, I believe SCSI is not that important.
http://www.exactaudiocopy.de/eac3.html
Look under Offset Technology.
/Hugo
http://www.exactaudiocopy.de/eac3.html
Look under Offset Technology.
/Hugo
This is a facinating thread that seems to be taking on a life of its own, clearly there is some need for some serious threads just devoted to the conversion of LPs to digital and subsequent post processing.
I am fairly ignorant on the maths involved in audio upsampling etc but as a digital imaging industry guy I can see the parrallels. With images in 8 bit file heavy editing causes posterization (tone break up) I won't go into the numbers etc, but converting a file to a higher bit depth can create amazing improvments.....but...
Only if the image is then appropriatley edited/altered after the change.
For example changing the bit depth to 16 leaves the file looking exactly the same, but if you apply blur filters at low levels, colour blending, add noise at low levels, re-sharpen etc it is possible to get a result that is very very nice and quite similar to film based images, and this is especailly the case if the pixels are interpolated to much larger file sizes.
My point?
I think there is porbably great scope for recreating LP like qualities with digital processing via computers but there is a real need to look at sympathetically combining software processing solutions with hardware systems and I not sure that there is a great deal of agreement on how best to do this yet.......but to me the future looks exciting for computer based music reproduction.
I feel so much in audio is based on throwing money at problems rather than method and technique, with computer based systems we can move more towards the technique and away from the costly hardware solutions, which is exciting really.
Just think, not only can we digitize out LPs but we can optimise them for playback on our gear in our specific environment to meet our ears, and its won't send you broke doing it!
Cheers all
I am fairly ignorant on the maths involved in audio upsampling etc but as a digital imaging industry guy I can see the parrallels. With images in 8 bit file heavy editing causes posterization (tone break up) I won't go into the numbers etc, but converting a file to a higher bit depth can create amazing improvments.....but...
Only if the image is then appropriatley edited/altered after the change.
For example changing the bit depth to 16 leaves the file looking exactly the same, but if you apply blur filters at low levels, colour blending, add noise at low levels, re-sharpen etc it is possible to get a result that is very very nice and quite similar to film based images, and this is especailly the case if the pixels are interpolated to much larger file sizes.
My point?
I think there is porbably great scope for recreating LP like qualities with digital processing via computers but there is a real need to look at sympathetically combining software processing solutions with hardware systems and I not sure that there is a great deal of agreement on how best to do this yet.......but to me the future looks exciting for computer based music reproduction.
I feel so much in audio is based on throwing money at problems rather than method and technique, with computer based systems we can move more towards the technique and away from the costly hardware solutions, which is exciting really.
Just think, not only can we digitize out LPs but we can optimise them for playback on our gear in our specific environment to meet our ears, and its won't send you broke doing it!
Cheers all
I don't know the exact ins and outs, as stated earlier.
Much of my info comes from the Wadia site. Maybe they mean different things in this regard (as in oversampling=upsampling), but you can call it interpolation, dithering or random noise. In that regard there's no difference between digital sound reproduction and image editing. To put it another way, increasing the sampling rate from 44.1k to 88.2 doesn't really do anything. It just repeats the same sample twice. More or less, anyway. Of course, so does a film projector. Actually, every frame is projected trice. But there's a difference. 24 fps flicker. (Even 50 and 60Hz flicker!) 72 fps don't. 44.1k is outside of the human hearing range. It's already 72 fps.
Much of my info comes from the Wadia site. Maybe they mean different things in this regard (as in oversampling=upsampling), but you can call it interpolation, dithering or random noise. In that regard there's no difference between digital sound reproduction and image editing. To put it another way, increasing the sampling rate from 44.1k to 88.2 doesn't really do anything. It just repeats the same sample twice. More or less, anyway. Of course, so does a film projector. Actually, every frame is projected trice. But there's a difference. 24 fps flicker. (Even 50 and 60Hz flicker!) 72 fps don't. 44.1k is outside of the human hearing range. It's already 72 fps.
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