Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
Do you have it, or are you just making a "wtf" statement at a quote?Morpheus wrote:
Ok. TBH..... not impressed by the DACMagic...Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
At the whole "Upsampling" technology.Zimmer wrote:
Do you have it, or are you just making a "wtf" statement at a quote?Morpheus wrote:
Ok. TBH..... not impressed by the DACMagic...Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
EMU is totally entry-level.Jenspm wrote:
Oh, ok then. Budget is flexible, I just thought it would be less than £100. Used the EMU0404 as a reference, which I thought was relatively up-the-scale.
So would this EMU be enough, then? If I have to hop to a DACMagic (ie £200+), I think I'll have to reconsider the whole thing (e.g. step down on the speakers) as I'm not prepared to spend £550 on audio equipment at this time.
Last edited by Uzique (2011-01-02 17:10:52)
The only thing you're adding by changing the sample rate is distortion. It is to be avoided, not endorsed. Changing the bit depth simply does nothing, as there is no data in the extra eight bits.Morpheus wrote:
At the whole "Upsampling" technology.Zimmer wrote:
Do you have it, or are you just making a "wtf" statement at a quote?Morpheus wrote:
Ok. TBH..... not impressed by the DACMagic...
I don't understand the point of adding shit that's not there to make it 'sound good'.
but the marketing for DACMagic says thta it'll up-convert to 24/196... thereby, making the statement that it'll take any shitty mp3 off youtube, and make it sound like it's in a recording studio. And I'm calling bullshit.Freezer7Pro wrote:
The only thing you're adding by changing the sample rate is distortion. It is to be avoided, not endorsed. Changing the bit depth simply does nothing, as there is no data in the extra eight bits.Morpheus wrote:
At the whole "Upsampling" technology.Zimmer wrote:
Do you have it, or are you just making a "wtf" statement at a quote?
I don't understand the point of adding shit that's not there to make it 'sound good'.
No, it says that it'll translate a 16/44 PCM signal and turn it into some 192/24 bit signal somewhere on the inside. There probably isn't too much loss in the conversion, but it's pointless for all intents and purposes other than translating for a chip that can only handle 192/24.Morpheus wrote:
but the marketing for DACMagic says thta it'll up-convert to 24/196... thereby, making the statement that it'll take any shitty mp3 off youtube, and make it sound like it's in a recording studio. And I'm calling bullshit.Freezer7Pro wrote:
The only thing you're adding by changing the sample rate is distortion. It is to be avoided, not endorsed. Changing the bit depth simply does nothing, as there is no data in the extra eight bits.Morpheus wrote:
At the whole "Upsampling" technology.
I don't understand the point of adding shit that's not there to make it 'sound good'.
I must have missed that point, but my view still stands - you can't get something from nothingFreezer7Pro wrote:
No, it says that it'll translate a 16/44 PCM signal and turn it into some 192/24 bit signal somewhere on the inside. There probably isn't too much loss in the conversion, but it's pointless for all intents and purposes other than translating for a chip that can only handle 192/24Morpheus wrote:
but the marketing for DACMagic says thta it'll up-convert to 24/196... thereby, making the statement that it'll take any shitty mp3 off youtube, and make it sound like it's in a recording studio. And I'm calling bullshit.Freezer7Pro wrote:
The only thing you're adding by changing the sample rate is distortion. It is to be avoided, not endorsed. Changing the bit depth simply does nothing, as there is no data in the extra eight bits.
Yea.................... So, basically, this thing is only useful if all your digital music is already at24bit/192kHz...
- Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
- Adaptive Time Filtering (ATF™) asynchronous upsampling technology converts 16-24 bit audio (at any standard sampling frequency between 32-96kHz) to 24 bit/192kHz
- Digital filter: Texas Instruments TMS 320VC5501 DSP upsampling to 24bit 192kHz
- Audio output up-sampling: Fixed 24bit 192kHz
No, it doesn't make anything up, there simply is nothing there. It just increases the resolution; it turns data "0001011100011001" into "000101110001100100000000"Morpheus wrote:
I must have missed that point, but my view still stands - you can't get something from nothingFreezer7Pro wrote:
No, it says that it'll translate a 16/44 PCM signal and turn it into some 192/24 bit signal somewhere on the inside. There probably isn't too much loss in the conversion, but it's pointless for all intents and purposes other than translating for a chip that can only handle 192/24Morpheus wrote:
but the marketing for DACMagic says thta it'll up-convert to 24/196... thereby, making the statement that it'll take any shitty mp3 off youtube, and make it sound like it's in a recording studio. And I'm calling bullshit.Yea.................... So, basically, this thing is only useful if all your digital music is already at24bit/192kHz...
- Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
- Adaptive Time Filtering (ATF™) asynchronous upsampling technology converts 16-24 bit audio (at any standard sampling frequency between 32-96kHz) to 24 bit/192kHz
- Digital filter: Texas Instruments TMS 320VC5501 DSP upsampling to 24bit 192kHz
- Audio output up-sampling: Fixed 24bit 192kHz
"Changing the bit depth does nothing" actually does something - "as there is no data in the extra... bits" exactly. It has to make up something to fill in those extra bits. essentially, adding digital noise.
Surely they work off a waveform plot of the sound. You wouldn't get nothing added - certainly you wouldn't get 0s on the end like you've shown, because that is a different number (unless it's a little endian representation). You just have more sample points of that waveform expressed in a higher resolution. It'll just be a case if artificially filling in gaps between existing sample points - which should make for a better quality output, depending on how well it is done.Freezer7Pro wrote:
No, it doesn't make anything up, there simply is nothing there. It just increases the resolution; it turns data "0001011100011001" into "000101110001100100000000"Morpheus wrote:
I must have missed that point, but my view still stands - you can't get something from nothingFreezer7Pro wrote:
No, it says that it'll translate a 16/44 PCM signal and turn it into some 192/24 bit signal somewhere on the inside. There probably isn't too much loss in the conversion, but it's pointless for all intents and purposes other than translating for a chip that can only handle 192/24Yea.................... So, basically, this thing is only useful if all your digital music is already at24bit/192kHz...
- Essentially it converts 16-24 bit audio to 24 bit/192kHz so you hear so much more from compressed material.
- Adaptive Time Filtering (ATF™) asynchronous upsampling technology converts 16-24 bit audio (at any standard sampling frequency between 32-96kHz) to 24 bit/192kHz
- Digital filter: Texas Instruments TMS 320VC5501 DSP upsampling to 24bit 192kHz
- Audio output up-sampling: Fixed 24bit 192kHz
"Changing the bit depth does nothing" actually does something - "as there is no data in the extra... bits" exactly. It has to make up something to fill in those extra bits. essentially, adding digital noise.
Yes, just saying that it's adding zeroes at the end isn't quite accurate, but I think it demonstrates the point quite well.Bertster7 wrote:
Surely they work off a waveform plot of the sound. You wouldn't get nothing added - certainly you wouldn't get 0s on the end like you've shown, because that is a different number (unless it's a little endian representation). You just have more sample points of that waveform expressed in a higher resolution. It'll just be a case if artificially filling in gaps between existing sample points - which should make for a better quality output, depending on how well it is done.Freezer7Pro wrote:
No, it doesn't make anything up, there simply is nothing there. It just increases the resolution; it turns data "0001011100011001" into "000101110001100100000000"Morpheus wrote:
I must have missed that point, but my view still stands - you can't get something from nothingFreezer7Pro wrote:
No, it says that it'll translate a 16/44 PCM signal and turn it into some 192/24 bit signal somewhere on the inside. There probably isn't too much loss in the conversion, but it's pointless for all intents and purposes other than translating for a chip that can only handle 192/24
Yea.................... So, basically, this thing is only useful if all your digital music is already at24bit/192kHz...
"Changing the bit depth does nothing" actually does something - "as there is no data in the extra... bits" exactly. It has to make up something to fill in those extra bits. essentially, adding digital noise.
analogue doesn't have to fill in any gaps...Freezer7Pro wrote:
Yes, just saying that it's adding zeroes at the end isn't quite accurate, but I think it demonstrates the point quite well.Bertster7 wrote:
Surely they work off a waveform plot of the sound. You wouldn't get nothing added - certainly you wouldn't get 0s on the end like you've shown, because that is a different number (unless it's a little endian representation). You just have more sample points of that waveform expressed in a higher resolution. It'll just be a case if artificially filling in gaps between existing sample points - which should make for a better quality output, depending on how well it is done.
Regarding the sample rate conversion, I can't quite see how any digital signal processing can fill in the gaps better than the filter on the analogue output.
That's not actually hard.. using spline functions.Morpheus wrote:
I edited the previous image to not have the area, or original line.... go ahead, and try to add it in. Better yet, add in more blue dots, then try to re-approximate the line
This is more likely.mcminty wrote:
^ You got it backwards.....
Sample rate - 44.1/48/96/192/etc kHz - is the number of times a sample is taken per second. A hint to this fact is that sample rate is in Hertz, or cycles per second.
Bit depth - 16/24/etc bit - is the amount of information contained in each sample.
herp derp. Someone explained it to me wrong, then... (Oh, hm, yea, i see that, makes sense now. Duh. I think I was posting at, like, 3AM.....)mcminty wrote:
^ You got it backwards.....
Sample rate - 44.1/48/96/192/etc kHz - is the number of times a sample is taken per second. A hint to this fact is that sample rate is in Hertz, or cycles per second.
Bit depth - 16/24/etc bit - is the amount of information contained in each sample.That's not actually hard.. using spline functions.Morpheus wrote:
I edited the previous image to not have the area, or original line.... go ahead, and try to add it in. Better yet, add in more blue dots, then try to re-approximate the line
i got them within 3 weeks from bugden audio, i think you just have to keep looking back for when theyre on sale. you might be able to pre order as wellLucien wrote:
can I actually buy the fischer dba-02 anywhere? what's a good alternative?
Because they work in completely different ways doing different things. Analogue filters just chop bits off the signal, they don't fill in gaps.Freezer7Pro wrote:
Yes, just saying that it's adding zeroes at the end isn't quite accurate, but I think it demonstrates the point quite well.Bertster7 wrote:
Surely they work off a waveform plot of the sound. You wouldn't get nothing added - certainly you wouldn't get 0s on the end like you've shown, because that is a different number (unless it's a little endian representation). You just have more sample points of that waveform expressed in a higher resolution. It'll just be a case if artificially filling in gaps between existing sample points - which should make for a better quality output, depending on how well it is done.Freezer7Pro wrote:
No, it doesn't make anything up, there simply is nothing there. It just increases the resolution; it turns data "0001011100011001" into "000101110001100100000000"
Regarding the sample rate conversion, I can't quite see how any digital signal processing can fill in the gaps better than the filter on the analogue output.
It's a lot easier when you have several thousand samples per second. Think anti aliasing.... But there's no point to that because you can't get a better image from information that isn't there....Morpheus wrote:
I edited the previous image to not have the area, or original line.... go ahead, and try to add it in. Better yet, add in more blue dots, then try to re-approximate the line:
http://static.bf2s.com/files/user/31499 … vebits.png
Oh, but they do. A low-pass filter will turn the output of a cheap 8-bit resistor-ladder DAC into a nice sine wave for you, no problem.Bertster7 wrote:
Because they work in completely different ways doing different things. Analogue filters just chop bits off the signal, they don't fill in gaps.Freezer7Pro wrote:
Yes, just saying that it's adding zeroes at the end isn't quite accurate, but I think it demonstrates the point quite well.Bertster7 wrote:
Surely they work off a waveform plot of the sound. You wouldn't get nothing added - certainly you wouldn't get 0s on the end like you've shown, because that is a different number (unless it's a little endian representation). You just have more sample points of that waveform expressed in a higher resolution. It'll just be a case if artificially filling in gaps between existing sample points - which should make for a better quality output, depending on how well it is done.
Regarding the sample rate conversion, I can't quite see how any digital signal processing can fill in the gaps better than the filter on the analogue output.
