Re: [music-dsp] Dither video and articles
mmh, Affiliation: Meridian Audio Ltd? -Message d'origine- From: Vicki Melchior Sent: Friday, February 06, 2015 2:21 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical problems that have long been discussed, some are enumerated in the above. As far as dithering at the 24th bit, I can't disagree more with a conclusion that says it's unnecessary in data handling. Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Whether you personally hear it is likely to depend both on how extensive your data flow pathway is and how good your playback equipment is. Vicki Melchior On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote: On 6/02/2015 1:50 PM, Tom Duffy wrote: The AES report is highly controversial. Plenty of sources dispute the findings. Can you name some? Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp - Aucun virus trouve dans ce message. Analyse effectuee par AVG - www.avg.fr Version: 2015.0.5645 / Base de donnees virale: 4281/9068 - Date: 06/02/2015 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
It was just several times the same fading in/out noise at different levels, just to see if you hear quieter things than I do, I thought you'd have guessed that. https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing (0dB, -36dB, -54dB, -66dB, -72dB, -78dB) Here if I make the starting noise annoying, then I hear the first 4 parts, until 18:00. Thus, if 0dB is my threshold of annoyance, I can't hear -72dB. So you hear it at -78dB? Would be interesting to know how many can, and if it's subjective or a matter of testing environment (the variable already being the 0dB annoyance starting point) -Message d'origine- From: Andrew Simper Sent: Friday, February 06, 2015 3:21 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles Sorry, you said until, which is even more confusing. There are multiple points when I hear the noise until since it sounds like the noise is modulated in amplitude by a sine like LFO for the entire file, so the volume of the noise ramps up and down in a cyclic manner. The last ramping I hear fades out at around the 28.7 second mark when it is hard to tell if it just ramps out at that point or is just on the verge of ramping up again and then the file ends at 28.93 seconds. I have not tried to measure the LFO wavelength or any other such things, this is just going on listening alone. All the best, Andrew Simper On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote: On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote: Just out of curiosity, until which point do you hear the noise in this little test (a 32bit float wav), starting from a bearable first part? https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing I hear noise immediately in that recording, it's hard to tell exactly the time I can first hear it since there is some latency from when I press play to when the sound starts, but as far as I can tell it is straight away. Why do you ask such silly questions? All the best, Andrew Simper -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp - Aucun virus trouve dans ce message. Analyse effectuee par AVG - www.avg.fr Version: 2015.0.5645 / Base de donnees virale: 4281/9068 - Date: 06/02/2015 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Hi Michael, I know that you already understand this, and comment that this is for internal calculations, but for the sake of anyone who might misinterpret your 32-bit vs 64-bit comment, I’ll point out that this is a situation of error feedback—the resulting error is much greater than the sample sizes you’re talking about, and can result in differences far above the 24-bit level. A simple example is the ubiquitous direct form I biquad, which goes all to hell in lower audio frequencies with 24-bit storage (unless you noise shape or increase resolution). Nigel On Feb 6, 2015, at 10:24 AM, Michael Gogins michael.gog...@gmail.com wrote: Do not believe anything that is not confirmed to a high degree of statistical signifance (say, 5 standard deviations) by a double-blind test using an ABX comparator. That said, the AES study did use double-blind testing. I did not read the article, only the abstract, so cannot say more about the study. In my own work, I have verified with a double-blind ABX comparator at a high degree of statistical significance that I can hear the differences in certain selected portions of the same Csound piece rendered with 32 bit floating point samples versus 64 bit floating point samples. These are sample words used in internal calculations, not for output soundfiles. What I heard was differences in the sound of the same filter algorithm. These differences were not at all hard to hear, but they occurred in only one or two places in the piece. I have not myself been able to hear differences in audio output quality between CD audio and high-resolution audio, but when I get the time I may try again, now that I have a better idea what to listen for. Regards, Mike - Michael Gogins Irreducible Productions http://michaelgogins.tumblr.com Michael dot Gogins at gmail dot com On Fri, Feb 6, 2015 at 1:13 PM, Nigel Redmon earle...@earlevel.com wrote: Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. Quick observations: 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a volt). Hearing capabilities aside, the converter must be able to resolve this, and it must make it through the thermal (and other) noise of their equipment and move a speaker. If you’re not an electrical engineer, it may be difficult to grasp the problem that this poses. 2) I happened on a discussion in an audio forum, where a highly-acclaimed mastering engineer and voice on dither mentioned that he could hear the dither kick in when he pressed a certain button in the GUI of some beta software. The maker of the software had to inform him that he was mistaken on the function of the button, and in fact it didn’t affect the audio whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a great source of info to people and is clearly excellent at what he does, and everyone who works with audio runs into this at some point.) The mastering engineer graciously accepted his goof. 3) Mastering engineers invariably describe the differences in very subjective term. While this may be a necessity, it sure makes it difficult to pursue any kind of validation. From a mastering engineer to me, yesterday: 'To me the truncated version sounds colder, more glassy, with less richness in the bass and harmonics, and less front to back depth in the stereo field.’ 4) 24-bit audio will almost always have a far greater random noise floor than is necessary to dither, so they will be self-dithered. By “almost”, I mean that very near 100% of the time. Sure, you can create exceptions, such as synthetically generated simple tones, but it’s hard to imagine them happening in the course of normal music making. There is nothing magic about dither noise—it’s just mimicking the sort of noise that your electronics generates thermally. And when mastering engineers say they can hear truncation distortion at 24-bit, they don’t say “on this particular brief moment, this particular recording”—they seems to say it in general. It’s extremely unlikely that non-randomized truncation distortion even exists for most material at 24-bit. My point is simply that I’m not going to accept that mastering engineers can hear the 24th bit truncation just because they say they can. On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote: The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical
Re: [music-dsp] Dither video and articles
I SO agree with 4), that when it comes to recorded not synthesized (but even synthesized in some cases actually - I've made additive synths and it's a big CPU saver to avoid processing inaudible partials) audio, room noise is so much above the levels we're debating, that it's a bit silly. -Message d'origine- From: Nigel Redmon Sent: Friday, February 06, 2015 7:13 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. Quick observations: 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a volt). Hearing capabilities aside, the converter must be able to resolve this, and it must make it through the thermal (and other) noise of their equipment and move a speaker. If you’re not an electrical engineer, it may be difficult to grasp the problem that this poses. 2) I happened on a discussion in an audio forum, where a highly-acclaimed mastering engineer and voice on dither mentioned that he could hear the dither kick in when he pressed a certain button in the GUI of some beta software. The maker of the software had to inform him that he was mistaken on the function of the button, and in fact it didn’t affect the audio whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a great source of info to people and is clearly excellent at what he does, and everyone who works with audio runs into this at some point.) The mastering engineer graciously accepted his goof. 3) Mastering engineers invariably describe the differences in very subjective term. While this may be a necessity, it sure makes it difficult to pursue any kind of validation. From a mastering engineer to me, yesterday: 'To me the truncated version sounds colder, more glassy, with less richness in the bass and harmonics, and less front to back depth in the stereo field.’ 4) 24-bit audio will almost always have a far greater random noise floor than is necessary to dither, so they will be self-dithered. By “almost”, I mean that very near 100% of the time. Sure, you can create exceptions, such as synthetically generated simple tones, but it’s hard to imagine them happening in the course of normal music making. There is nothing magic about dither noise—it’s just mimicking the sort of noise that your electronics generates thermally. And when mastering engineers say they can hear truncation distortion at 24-bit, they don’t say “on this particular brief moment, this particular recording”—they seems to say it in general. It’s extremely unlikely that non-randomized truncation distortion even exists for most material at 24-bit. My point is simply that I’m not going to accept that mastering engineers can hear the 24th bit truncation just because they say they can. On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote: The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical problems that have long been discussed, some are enumerated in the above. As far as dithering at the 24th bit, I can't disagree more with a conclusion that says it's unnecessary in data handling. Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Whether you personally hear it is likely to depend both on how extensive your data flow pathway is and how good your playback equipment is. Vicki Melchior On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote: On 6/02/2015 1:50 PM, Tom Duffy wrote: The AES report is highly controversial. Plenty of sources dispute the findings. Can you name some? Ross. -- -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp - Aucun virus trouvé dans ce message. Analyse effectuée par AVG - www.avg.fr Version: 2015.0.5645 / Base de données virale: 4281/9068 - Date:
Re: [music-dsp] Dither video and articles
Yes, but note that in the case Michael is reporting, all filters have double-precision coeffs and data storage. It is only when passing samples between unit generators that the difference lies (either single or double precision is used). Still, I believe that there can be audible differences. Victor Lazzarini Dean of Arts, Celtic Studies, and Philosophy Maynooth University Ireland On 6 Feb 2015, at 18:43, Ethan Duni ethan.d...@gmail.com wrote: Thanks for the reference Vicki What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. I'm skeptical, but definitely hope that such a test gets undertaken and published. Would be interesting to have some real data either way. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Hopefully everyone agrees that the questions of what is appropriate for intermediate processing and what is appropriate for final distribution are quite different, and that substantially higher resolutions (and probably including dither) are indicated for intermediate processing. As Michael Goggins says: In my own work, I have verified with a double-blind ABX comparator at a high degree of statistical significance that I can hear the differences in certain selected portions of the same Csound piece rendered with 32 bit floating point samples versus 64 bit floating point samples. These are sample words used in internal calculations, not for output soundfiles. What I heard was differences in the sound of the same filter algorithm. These differences were not at all hard to hear, but they occurred in only one or two places in the piece. Indeed, it is not particularly difficult to cook up filter designs/algorithms that will break any given finite internal resolution. At some point those filter designs become pathological, but there are plenty of reasonable cases where 32 bit float internal precision is insufficient. Note that a 32-bit float only has 24 bits of mantissa, which is 8 bits less than is typically used in embedded fixed-point implementations (for sensitive components like filter guts, I mean). So even very standard stuff that has been around for decades in the fixed-point world will break if implemented naively in 32 bit float. E -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] ARMv8 64-bit Superpowered Audio Engine
Hi there, Just wanted to let everyone know that Superpowered's 64-bit library is, like Frankenstein's monster, ALIVE!!! and kicking ass. :) See here: http://superpowered.com/superpowered-audio-engine-64-bit-arm-android-ios/ Also, we have a write-up of some of the DSP optimization methods we made use of for switch to ARMv8 vs ARMv7. http://superpowered.com/64-bit-arm-optimization-audio-signal-processing/ Please don't hesitate to reach out if we can be of help with any of your projects or answer any questions --- he...@superpowered.com Thanks, Patrick @pv PS Happy Friday! -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Isn't it generally agreed that truncation noise is correlated with the signal? “Is correlated”? No, but it can be. First, if there is enough noise in the signal before truncation, then it’s dithered by default—no correlation. Second, if the signal is sufficiently complex, it seems, then there is no apparent correlation. See my video (https://www.youtube.com/watch?v=KCyA6LlB3As https://www.youtube.com/watch?v=KCyA6LlB3As) where I show a 32-bit float mix, truncated to 8-bit, nulled, and boosted +24 dB. There is no apparent correlation till the very end, even though the noise floor is not sufficient to self-dither. On Feb 6, 2015, at 10:42 AM, Tom Duffy tdu...@tascam.com wrote: Isn't it generally agreed that truncation noise is correlated with the signal? The human ear is excellent at picking up on correlation, so a system that introduces multiple correlated (noise) signals may reach a point where it is perceptual, even if the starting point is a 24 bit signal. I would believe this to be an explanation for why ProTools early hardware mixers were regarded as having problems - they used 24bit fixed point DSPs, coupled with fixed bit headroom management may have introduced truncation noise at a level higher than the 24 bit noise floor. Also, the dither noise source itself needs to be investigated. Studies have shown that a fixed repeated buffer of pre-generated white noise is immediately obvious (and non-pleasing) to the listener up to several hundred ms long - if that kind of source was used as a dither signal, the self correlation becomes even more problematic. Calculated a new PRDG value for each sample is expensive, which is why a pre-generated buffer is attractive to the implementor. --- Tom. On 2/6/2015 10:32 AM, Victor Lazzarini wrote: Quite. This conversation is veering down the vintage wine tasting alley. Victor Lazzarini Dean of Arts, Celtic Studies, and Philosophy Maynooth University Ireland On 6 Feb 2015, at 18:13, Nigel Redmon earle...@earlevel.com wrote: Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. Quick observations: 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a volt). Hearing capabilities aside, the converter must be able to resolve this, and it must make it through the thermal (and other) noise of their equipment and move a speaker. If you’re not an electrical engineer, it may be difficult to grasp the problem that this poses. 2) I happened on a discussion in an audio forum, where a highly-acclaimed mastering engineer and voice on dither mentioned that he could hear the dither kick in when he pressed a certain button in the GUI of some beta software. The maker of the software had to inform him that he was mistaken on the function of the button, and in fact it didn’t affect the audio whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a great source of info to people and is clearly excellent at what he does, and everyone who works with audio runs into this at some point.) The mastering engineer graciously accepted his goof. 3) Mastering engineers invariably describe the differences in very subjective term. While this may be a necessity, it sure makes it difficult to pursue any kind of validation. From a mastering engineer to me, yesterday: 'To me the truncated version sounds colder, more glassy, with less richness in the bass and harmonics, and less front to back depth in the stereo field.’ 4) 24-bit audio will almost always have a far greater random noise floor than is necessary to dither, so they will be self-dithered. By “almost”, I mean that very near 100% of the time. Sure, you can create exceptions, such as synthetically generated simple tones, but it’s hard to imagine them happening in the course of normal music making. There is nothing magic about dither noise—it’s just mimicking the sort of noise that your electronics generates thermally. And when mastering engineers say they can hear truncation distortion at 24-bit, they don’t say “on this particular brief moment, this particular recording”—they seems to say it in general. It’s extremely unlikely that non-randomized truncation distortion even exists for most material at 24-bit. My point is simply that I’m not going to accept that mastering engineers can hear the 24th bit truncation just because they say they can. On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote: The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible.
Re: [music-dsp] Dither video and articles
This was done before John ffitch (I believe it was he) changed the filter samples in even the single-precision version of Csound to use double-precision. And I think this change may have been made as a result of my report. Regards, Mike - Michael Gogins Irreducible Productions http://michaelgogins.tumblr.com Michael dot Gogins at gmail dot com On Fri, Feb 6, 2015 at 2:04 PM, Victor Lazzarini victor.lazzar...@nuim.ie wrote: Yes, but note that in the case Michael is reporting, all filters have double-precision coeffs and data storage. It is only when passing samples between unit generators that the difference lies (either single or double precision is used). Still, I believe that there can be audible differences. Victor Lazzarini Dean of Arts, Celtic Studies, and Philosophy Maynooth University Ireland On 6 Feb 2015, at 18:43, Ethan Duni ethan.d...@gmail.com wrote: Thanks for the reference Vicki What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. I'm skeptical, but definitely hope that such a test gets undertaken and published. Would be interesting to have some real data either way. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Hopefully everyone agrees that the questions of what is appropriate for intermediate processing and what is appropriate for final distribution are quite different, and that substantially higher resolutions (and probably including dither) are indicated for intermediate processing. As Michael Goggins says: In my own work, I have verified with a double-blind ABX comparator at a high degree of statistical significance that I can hear the differences in certain selected portions of the same Csound piece rendered with 32 bit floating point samples versus 64 bit floating point samples. These are sample words used in internal calculations, not for output soundfiles. What I heard was differences in the sound of the same filter algorithm. These differences were not at all hard to hear, but they occurred in only one or two places in the piece. Indeed, it is not particularly difficult to cook up filter designs/algorithms that will break any given finite internal resolution. At some point those filter designs become pathological, but there are plenty of reasonable cases where 32 bit float internal precision is insufficient. Note that a 32-bit float only has 24 bits of mantissa, which is 8 bits less than is typically used in embedded fixed-point implementations (for sensitive components like filter guts, I mean). So even very standard stuff that has been around for decades in the fixed-point world will break if implemented naively in 32 bit float. E -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
So you hear all 6 too? -Message d'origine- From: Richard Dobson Sent: Friday, February 06, 2015 4:10 PM To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] Dither video and articles On 06/02/2015 14:21, Andrew Simper wrote: Sorry, you said until, which is even more confusing. There are multiple points when I hear the noise until since it sounds like the noise is modulated in amplitude by a sine like LFO for the entire file, so the volume of the noise ramps up and down in a cyclic manner. The last ramping I hear fades out at around the 28.7 second mark when it is hard to tell if it just ramps out at that point or is just on the verge of ramping up again and then the file ends at 28.93 seconds. I have not tried to measure the LFO wavelength or any other such things, this is just going on listening alone. Its a series of six smoothly enveloped noise bursts (slowish rise/ slower decay) the first peaking at max amplitude (so you have to be ready to hear it as very loud!), then successively softer repeats until at some point it is (presumably?) too quiet to be heard. Very visible in Audacity using the Waveform (dB) display mode. So the word until is entirely appropriate. I do recommend visual inspection of waveforms in such situations to minimise guessing (or at least, to confirm the guesses or otherwise). In any case, I would expect people to hear all six, give a suitably quiet listening environment and an appropriately generous overall playback level etc. Richard Dobson -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Funny…you made me think of the event below, late last night, but I couldn’t recall what I was thinking of when I got up this morning. Oh well, it will come to me later maybe, I think...I take a little break and pick up Tape Op magazine, which I had rescued form the trunk of my car, from AES, flip it open to a quote from Brian Eno…oh yeah, now I remember my thought: Years ago, a friend invited me to a colloquy with Brian Eno. Eno talked about his first experience with CDs in some sort of show or presentation, where he had music on CD player over the sound system. It didn’t sound “right” to him, so he hooked up a cassette deck with a blank tape and mixed hiss into the sound system to improve the listening experience. Maybe it’s that “air” that people like to hear ;-) On Feb 5, 2015, at 11:32 PM, Andrew Simper a...@cytomic.com wrote: Hi Nigel, You're welcome! Thanks for spending the time and effort preparing examples so I could make some observations on. Yeah, with headphones my ears easily picked up the stereo-ness of the hiss as soon as I switched sources. If I was listening to an entire CD and all tracks had the same hiss I would have just assumed it would be part of the recording chain in making the CD, which I suppose in a sense it is, but the hiss definitely sounded quieter in headphones when it was mono. Now I'm just being lazy with the plugin, I can do it myself as a command line thing / plugin, but I just figured if you had recently compiled the plugin it would be an interesting addition to have! All the best, Andy -- cytomic -- sound music software -- On 6 February 2015 at 14:47, Nigel Redmon earle...@earlevel.com wrote: Funny, Andy, I was thinking about the merits of mono versus stereo dither a couple of nights ago while having dinner…while independent dither makes sense, in that your equipment’s background noise should be uncorrelated, there is the issue with headphones (maybe making it more obvious, more spacious?)…I didn’t think it through very far, just a thought to try out, but it’s interesting that you should bring it up... But actually, those files aren’t using my plug-in. Since the test didn’t require a constant residual level at various truncation levels (which is the best part of the plug-in—nothing like juggling a couple of gain plug-ins to manually compensate the gain in a null test, and blasting your ears off when a stray index finger mouse-scrolls bit-depth down to one or two bits with a high gain setting in place), I went with the off-the-shelf stuff, and not have a chance that someone would question whether my plug-in was doing something misleading. DP’s Quan Jr plug-in is supplying the dither. I can mod my plug-in for mono dither, though, and supply a version of that. You make an interesting observation, thanks. On Feb 5, 2015, at 6:31 PM, Andrew Simper a...@cytomic.com wrote: Hi Nigel, Can I please ask a favour? Can you please add a mono noise button to your dither plugin? In headphones the sudden onset of stereo hiss of the dither is pretty obvious and a little distracting in this example. I had a listen with a make mono plugin and the results were much less obvious between the 16-bit with dither and the float file. It would be interesting to hear a stereo source (eg the same Diva sounds but in unison) put through mono noise dithering. The differences are pretty clear to me, thanks for posting the files! My setup: (*) Switching between files randomly the three files randomly playing them back with unity gain (the float file padded -6 dB to have the same volume as the others) (*) FireFace UCX with headphone output set to -12 dB, all other gains at unity (*) Senheisser Amperior HD25 headphones My results (*) the float file is easy to spot, because of the differences when compared to the other two (*) the dithered one sounds hissy straight away when I switch to it, it is obvious that the hiss is stereo, my ears immediately hear that stereo difference, but otherwise it sounds like the original float file (*) the undithered one, right from the start, sounds like a harsher version of the float one with just a hint of noise as well, an aggressive subtle edge to the tone which just isn't in the original. When the fadeout comes then it becomes more obvious aliasing distortion that everyone is used to hearing. I also tried boosting the float version of the bass tone to -1 dB (so another 18 dB up from with the same test setup), it was loud, but not anywhere near the threshold of pain for me. I then boosted it another 12 dB on the headphone control (so 0 dB gain), so now 30 dB gain in total and my headphones were really shaking, this was a bit silly a level, but still definitely not painful to listen to. My point being that this is a very reasonable test signal to listen to, and it is clear to hear the differences even at low levels of gain. If I had to
Re: [music-dsp] 14-bit MIDI controls, how should we do Coarse and Fine?
Dealing with the order/existence of MSB/LSB is per-hardware, I'm afraid. There is only very little that's standard with MIDI, it mostly depends on the manufacturer. NRPNs aren't obscure, they're quite common out there. It's annoying to set up a parser, but once you have one working, you can deal with many devices. Sending them is of course easy. But yeah, 4 messages.. See NRPNs as a 4-message extension to MIDI to deliver a large amount of high-precision CCs. -Message d'origine- From: robert bristow-johnson Sent: Wednesday, February 04, 2015 10:51 PM To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] 14-bit MIDI controls, how should we do Coarse and Fine? On 2/4/15 4:21 PM, Didier Dambrin wrote: That's in theory, but in practice, controllers out there send 14bit values in different ways: -LSB before MSB -MSB before LSB -LSB or MSB sometimes missing (sigh..) yes, all of that. that's what the question is about. dealing with all of these possibilities. -NRPNs (much more common, with the same differences as above) well, maybe like the RPNs, they shouldn't do anything until you send 0xB0 0x62 0x7F 0xB0 0x63 0x7F or maybe i'm wrong about that. dunno shit about NRPN. anyway, i look at NRPNs a lot like an obscure SysEx. who knows what it does? about RPNs, the only RPNs i worry about are 0x – Pitch bend range 0x0001 – Fine tuning 0x0002 – Coarse tuning and maybe 0x0005 – Modulation depth range is RPN 0x0005 supposed to affect the range of the Mod Wheel like RPN 0x affects the range of the Pitch Wheel? i know how the latter is defined, but i can't figger out the former. and i still don't know how Portamento Time (MIDI Control #5) is defined w.r.t. an actual unit of time? and do they mean Time Constant (like exp(-t/tau) or maybe 5 times that (which might be settling time)? i suspect the latter. (and portamento can be done in a variety of different ways, doesn't have to be a simple LPF like in the old Moogs.) -pitch bends (common for mixers) pitch bend sends a 14 bit value in a single MIDI message. that's not a problem. In practice, there is thus no established standard, it all comes down to the controller's manual. hence my question. what have people here commonly done in practice (if they're willing to share a secret) to deal with Fine and Coarse? i'm thinking my 1 or 2 ms delay is the only way to be both flexible about the MSB/LSB order (and missing MSB or LSB as per the MIDI standard) *and* avoid the glitch going from 0x207F to 0x2080. is a 1 or 2 ms delay on a MIDI Control update considered really bad? which is worse, a 1 or 2 ms delay or a glitch lasting 1 or 2 ms? -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp - Aucun virus trouvé dans ce message. Analyse effectuée par AVG - www.avg.fr Version: 2015.0.5645 / Base de données virale: 4281/9056 - Date: 04/02/2015 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Two pole one zero biquad filter
Hi All, I'm going to show some transfer curves of what I call scp biquad v1, which is my first and simplest digital approximation of an s-domain analog 2-pole resonant lowpass filter, using the following time domain function: y[n] = a0*x[n] + a1*x[n-1] - b1*y[n-1] - b2*y[n-2] Some expressed doubt that I might be faking the transfer curves, so I actually implemented this as an audio plugin, and grabbed and merged several screenshots from a spectrum analyzer plugin. Here are the graphs: http://morpheus.spectralhead.com/img/scp-biquad.png Parameters are: q = 10 w = 0.013, 0.025, 0.05, 0.1, 0.2, 0.3, 0.4, 0.45 (573, 1100, 2205, 4410, 8820, 13230, 17640, 19845 Hz) As you see, there's still some misbehaving near Nyquist, but this is still work-in-progres. I have two ideas on how to improve this further. Your homework: -- Implement the filter with the above transfer function using 2 poles and 1 zero. In other words, implement a two pole biquadratic lowpass filter formula with the 2nd zero fixed at origin. You guys are masters of formal and symbolic computation, right? So I'm sure it's going to be child's play for you. I won't disturb your thinking with my boring explanations ;) Good luck! ;) Best regards, Peter Schoffhauzer Prof. Bitflip -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] R: Two pole one zero biquad filter
Nicely done, Peter and thanks for sharing. I was the one who suggested to be super-clear and void any doubt in us judging your results, because via email and with no paper published that should be not ever an ultimate argument in the discussion. I never thought that you could be cheating on your proof. To me, without going into any math and by just giving a thought/quick glance at the SPAN screenshot, they look like the Massberg analog-matched, or something very similar. Ciao Marco -Messaggio originale- Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp- boun...@music.columbia.edu] Per conto di Peter S Inviato: venerdì 6 febbraio 2015 23:18 A: A discussion list for music-related DSP Oggetto: [music-dsp] Two pole one zero biquad filter Hi All, I'm going to show some transfer curves of what I call scp biquad v1, which is my first and simplest digital approximation of an s-domain analog 2-pole resonant lowpass filter, using the following time domain function: y[n] = a0*x[n] + a1*x[n-1] - b1*y[n-1] - b2*y[n-2] Some expressed doubt that I might be faking the transfer curves, so I actually implemented this as an audio plugin, and grabbed and merged several screenshots from a spectrum analyzer plugin. Here are the graphs: http://morpheus.spectralhead.com/img/scp-biquad.png Parameters are: q = 10 w = 0.013, 0.025, 0.05, 0.1, 0.2, 0.3, 0.4, 0.45 (573, 1100, 2205, 4410, 8820, 13230, 17640, 19845 Hz) As you see, there's still some misbehaving near Nyquist, but this is still work-in-progres. I have two ideas on how to improve this further. Your homework: -- Implement the filter with the above transfer function using 2 poles and 1 zero. In other words, implement a two pole biquadratic lowpass filter formula with the 2nd zero fixed at origin. You guys are masters of formal and symbolic computation, right? So I'm sure it's going to be child's play for you. I won't disturb your thinking with my boring explanations ;) Good luck! ;) Best regards, Peter Schoffhauzer Prof. Bitflip -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
The self dither argument is not as obvious as it may appear. To be effective at dithering, the noise has to be at the right level of course but also should be white and temporally constant. The noise floors present in music data normally come from the self noise of the analog components used in recording and are composites of a number of noise PDFs. For example, a graph in a second paper by the same group (cited below if wanted) shows spectra of the measured noise floors from around a dozen recordings. The noise spectra are composites with the lower frequencies clearly 1/f noise and the upper frequencies summing closer to flat. Whether composite noise of this sort is both temporally continuous and white enough to be relied on for dither needs to be shown; it's been shown under at least some circumstances (not in these papers) that a truncation distortion spectrum can be produced and measured when signals are truncated to 24b. I'm not saying the self dither argument is necessarily wrong; but it needs verification as to when and where it is reliably valid. If 24b truncation turns out to be demonstrably audible in an AB/X, then the self dither idea clearly needs to be rethought. Vicki Melchior (graph mentioned is fig 8 in this paper: http://www.aes.org/e-lib/browse.cfm?elib=17501) On Feb 6, 2015, at 2:20 PM, Nigel Redmon wrote: First, if there is enough noise in the signal before truncation, then it’s dithered by default—no correlation. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Original Message Subject: Re: [music-dsp] Dither video and articles From: Vicki Melchior vmelch...@earthlink.net Date: Fri, February 6, 2015 2:23 pm To: A discussion list for music-related DSP music-dsp@music.columbia.edu -- The self dither argument is not as obvious as it may appear. To be effective at dithering, the noise has to be at the right level of course but also should be white and temporally constant. � why does it have to be white?� or why should it? -- � r b-j � � � � � � � � � r...@audioimagination.com � Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Sorry, you said until, which is even more confusing. There are multiple points when I hear the noise until since it sounds like the noise is modulated in amplitude by a sine like LFO for the entire file, so the volume of the noise ramps up and down in a cyclic manner. The last ramping I hear fades out at around the 28.7 second mark when it is hard to tell if it just ramps out at that point or is just on the verge of ramping up again and then the file ends at 28.93 seconds. I have not tried to measure the LFO wavelength or any other such things, this is just going on listening alone. All the best, Andrew Simper On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote: On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote: Just out of curiosity, until which point do you hear the noise in this little test (a 32bit float wav), starting from a bearable first part? https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing I hear noise immediately in that recording, it's hard to tell exactly the time I can first hear it since there is some latency from when I press play to when the sound starts, but as far as I can tell it is straight away. Why do you ask such silly questions? All the best, Andrew Simper -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Musical pitch detection by counting bitflips
On 04/02/2015, Alan Wolfe alan.wo...@gmail.com wrote: Do you have a write up of this anywhere? I'd love to read more and have a place to point people to for more info. Also it would be neat to see how you extend this to higher dimensions, and also your log2 calculation is quite intriguing (: Sadly, I've not yet found time to make a proper write-up yet, but here's the output of a test program for the log2 approximation, which tells a lot in numbers: http://morpheus.spectralhead.com/txt/log-test.txt In a nutshell, I compare the operand to what I call a triangular binary filterbank using various decorrelation analysis operators (the simplest being a simple bitflip counter using binary XOR operation). The interesting thing about this, is when you normalize that to 0-1 range and visually rotate the whole thing 90 degrees to the left, then the normalized graph on the right side looks effectively like the response curve of a 2 pole resonant filter, with a distinct resonant peak with a maxima of 1.25 (with interesting aliasing artifacts around the resonance). I confirmed numerically (though not formally) that (for 32 bit numbers) this filter always resonates (=has a local maxima) at the 'band' that corresponds to floor(log2(x)), which you can later use to calculate log2(x) to arbitrary high precision (in the test, I used simple linear interpolation). Of course you can do optimizations, and skip much of the calculations because you only need to find the 'peak' (local maxima) of this resonant filter, which you can do rather easily if you consider the monotonity of the transfer curve. (And the whole expensive normalization is only to make fancy graphs and formal analysis, in practice you can skip it entirely, the whole thing becomes a simple loop and some bit operations.) Now, by using a different filterbank (like, you skip every 2nd row in the analyis filter), you can approximate log2(x) to arbitrarily _low_ precision as well - if you want less precision, you just compare it against a lower number of 'bases', and find the local maxima. In that sense, it's a bit like FFT - you can arbitrarily choose the length of the filter and thus the number of 'bases' you compare the data against, giving you higher or lower precision with higher or lower cost. So, that's the log2 story in a nutshell, I've yet not seen this particular approach (though I didn't do literature research as I just found this accidentally - hope I'm not reinventing the wheel). This simple idea can be used for a myriad other things, not just log2 approximation, and I see potential uses specifically in audio processing as well (though I'll need more experiments in that area). It would be nice if later I could find time to do a more proper write-up and put it online with some example code. Best, Peter -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote: Just out of curiosity, until which point do you hear the noise in this little test (a 32bit float wav), starting from a bearable first part? https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing I hear noise immediately in that recording, it's hard to tell exactly the time I can first hear it since there is some latency from when I press play to when the sound starts, but as far as I can tell it is straight away. Why do you ask such silly questions? All the best, Andrew Simper -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Looking for plug-in developers
Hi Kevin, My name is Miguel Conradi. I'm telecom engineer specialized in DSP. I work as embedded systems developer, and I did few plugins in Juce recently. I'm not the most experienced guy in plugins development concretely, but I'm experienced in C/C++ and matlab (and obviously in DSP). Could you explain me more in detail the project please? Thanks. All the best, El jueves, 5 de febrero de 2015, Kevin Vanwulpen vanwul...@gmail.com escribió: Hi, We are looking for someone to help us in bringing a few audio plugins from concept to fruition as contractor. This is for DAW plug-ins (the usual formats AAX/AU/etc). Please contact me privately with some info about yourself and I’ll give you some more background on the project. Thanks, Kevin -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- Saludos/Regards, Miguel -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical problems that have long been discussed, some are enumerated in the above. As far as dithering at the 24th bit, I can't disagree more with a conclusion that says it's unnecessary in data handling. Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Whether you personally hear it is likely to depend both on how extensive your data flow pathway is and how good your playback equipment is. Vicki Melchior On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote: On 6/02/2015 1:50 PM, Tom Duffy wrote: The AES report is highly controversial. Plenty of sources dispute the findings. Can you name some? Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
On 06/02/2015 14:21, Andrew Simper wrote: Sorry, you said until, which is even more confusing. There are multiple points when I hear the noise until since it sounds like the noise is modulated in amplitude by a sine like LFO for the entire file, so the volume of the noise ramps up and down in a cyclic manner. The last ramping I hear fades out at around the 28.7 second mark when it is hard to tell if it just ramps out at that point or is just on the verge of ramping up again and then the file ends at 28.93 seconds. I have not tried to measure the LFO wavelength or any other such things, this is just going on listening alone. Its a series of six smoothly enveloped noise bursts (slowish rise/ slower decay) the first peaking at max amplitude (so you have to be ready to hear it as very loud!), then successively softer repeats until at some point it is (presumably?) too quiet to be heard. Very visible in Audacity using the Waveform (dB) display mode. So the word until is entirely appropriate. I do recommend visual inspection of waveforms in such situations to minimise guessing (or at least, to confirm the guesses or otherwise). In any case, I would expect people to hear all six, give a suitably quiet listening environment and an appropriately generous overall playback level etc. Richard Dobson -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. Quick observations: 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a volt). Hearing capabilities aside, the converter must be able to resolve this, and it must make it through the thermal (and other) noise of their equipment and move a speaker. If you’re not an electrical engineer, it may be difficult to grasp the problem that this poses. 2) I happened on a discussion in an audio forum, where a highly-acclaimed mastering engineer and voice on dither mentioned that he could hear the dither kick in when he pressed a certain button in the GUI of some beta software. The maker of the software had to inform him that he was mistaken on the function of the button, and in fact it didn’t affect the audio whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a great source of info to people and is clearly excellent at what he does, and everyone who works with audio runs into this at some point.) The mastering engineer graciously accepted his goof. 3) Mastering engineers invariably describe the differences in very subjective term. While this may be a necessity, it sure makes it difficult to pursue any kind of validation. From a mastering engineer to me, yesterday: 'To me the truncated version sounds colder, more glassy, with less richness in the bass and harmonics, and less front to back depth in the stereo field.’ 4) 24-bit audio will almost always have a far greater random noise floor than is necessary to dither, so they will be self-dithered. By “almost”, I mean that very near 100% of the time. Sure, you can create exceptions, such as synthetically generated simple tones, but it’s hard to imagine them happening in the course of normal music making. There is nothing magic about dither noise—it’s just mimicking the sort of noise that your electronics generates thermally. And when mastering engineers say they can hear truncation distortion at 24-bit, they don’t say “on this particular brief moment, this particular recording”—they seems to say it in general. It’s extremely unlikely that non-randomized truncation distortion even exists for most material at 24-bit. My point is simply that I’m not going to accept that mastering engineers can hear the 24th bit truncation just because they say they can. On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote: The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical problems that have long been discussed, some are enumerated in the above. As far as dithering at the 24th bit, I can't disagree more with a conclusion that says it's unnecessary in data handling. Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Whether you personally hear it is likely to depend both on how extensive your data flow pathway is and how good your playback equipment is. Vicki Melchior On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote: On 6/02/2015 1:50 PM, Tom Duffy wrote: The AES report is highly controversial. Plenty of sources dispute the findings. Can you name some? Ross. -- -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Dither video and articles
Isn't it generally agreed that truncation noise is correlated with the signal? The human ear is excellent at picking up on correlation, so a system that introduces multiple correlated (noise) signals may reach a point where it is perceptual, even if the starting point is a 24 bit signal. I would believe this to be an explanation for why ProTools early hardware mixers were regarded as having problems - they used 24bit fixed point DSPs, coupled with fixed bit headroom management may have introduced truncation noise at a level higher than the 24 bit noise floor. Also, the dither noise source itself needs to be investigated. Studies have shown that a fixed repeated buffer of pre-generated white noise is immediately obvious (and non-pleasing) to the listener up to several hundred ms long - if that kind of source was used as a dither signal, the self correlation becomes even more problematic. Calculated a new PRDG value for each sample is expensive, which is why a pre-generated buffer is attractive to the implementor. --- Tom. On 2/6/2015 10:32 AM, Victor Lazzarini wrote: Quite. This conversation is veering down the vintage wine tasting alley. Victor Lazzarini Dean of Arts, Celtic Studies, and Philosophy Maynooth University Ireland On 6 Feb 2015, at 18:13, Nigel Redmon earle...@earlevel.com wrote: Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. Quick observations: 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a volt). Hearing capabilities aside, the converter must be able to resolve this, and it must make it through the thermal (and other) noise of their equipment and move a speaker. If you’re not an electrical engineer, it may be difficult to grasp the problem that this poses. 2) I happened on a discussion in an audio forum, where a highly-acclaimed mastering engineer and voice on dither mentioned that he could hear the dither kick in when he pressed a certain button in the GUI of some beta software. The maker of the software had to inform him that he was mistaken on the function of the button, and in fact it didn’t affect the audio whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a great source of info to people and is clearly excellent at what he does, and everyone who works with audio runs into this at some point.) The mastering engineer graciously accepted his goof. 3) Mastering engineers invariably describe the differences in very subjective term. While this may be a necessity, it sure makes it difficult to pursue any kind of validation. From a mastering engineer to me, yesterday: 'To me the truncated version sounds colder, more glassy, with less richness in the bass and harmonics, and less front to back depth in the stereo field.’ 4) 24-bit audio will almost always have a far greater random noise floor than is necessary to dither, so they will be self-dithered. By “almost”, I mean that very near 100% of the time. Sure, you can create exceptions, such as synthetically generated simple tones, but it’s hard to imagine them happening in the course of normal music making. There is nothing magic about dither noise—it’s just mimicking the sort of noise that your electronics generates thermally. And when mastering engineers say they can hear truncation distortion at 24-bit, they don’t say “on this particular brief moment, this particular recording”—they seems to say it in general. It’s extremely unlikely that non-randomized truncation distortion even exists for most material at 24-bit. My point is simply that I’m not going to accept that mastering engineers can hear the 24th bit truncation just because they say they can. On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote: The following published double blind test contradicts the results of the old Moran/Meyer publication in showing (a) that the differences between CD and higher resolution sources is audible and (b) that failure to dither at the 16th bit is also audible. http://www.aes.org/e-lib/browse.cfm?elib=17497 The Moran/Meyer tests had numerous technical problems that have long been discussed, some are enumerated in the above. As far as dithering at the 24th bit, I can't disagree more with a conclusion that says it's unnecessary in data handling. Mastering engineers can hear truncation error at the 24th bit but say it is subtle and may require experience or training to pick up. What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. The problem with failing to dither at 24b is that many such truncation steps would be done
Re: [music-dsp] Dither video and articles
Thanks for the reference Vicki What they are hearing is not noise or peaks sitting at the 24th bit but rather the distortion that goes with truncation at 24b, and it is said to have a characteristic coloration effect on sound. I'm aware of an effort to show this with AB/X tests, hopefully it will be published. I'm skeptical, but definitely hope that such a test gets undertaken and published. Would be interesting to have some real data either way. The problem with failing to dither at 24b is that many such truncation steps would be done routinely in mastering, and thus the truncation distortion products continue to build up. Hopefully everyone agrees that the questions of what is appropriate for intermediate processing and what is appropriate for final distribution are quite different, and that substantially higher resolutions (and probably including dither) are indicated for intermediate processing. As Michael Goggins says: In my own work, I have verified with a double-blind ABX comparator at a high degree of statistical significance that I can hear the differences in certain selected portions of the same Csound piece rendered with 32 bit floating point samples versus 64 bit floating point samples. These are sample words used in internal calculations, not for output soundfiles. What I heard was differences in the sound of the same filter algorithm. These differences were not at all hard to hear, but they occurred in only one or two places in the piece. Indeed, it is not particularly difficult to cook up filter designs/algorithms that will break any given finite internal resolution. At some point those filter designs become pathological, but there are plenty of reasonable cases where 32 bit float internal precision is insufficient. Note that a 32-bit float only has 24 bits of mantissa, which is 8 bits less than is typically used in embedded fixed-point implementations (for sensitive components like filter guts, I mean). So even very standard stuff that has been around for decades in the fixed-point world will break if implemented naively in 32 bit float. E -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
