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#1
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Audio editing and converting
I thought I'd start a new thread here on this subject, since the YouTube one
had drifted off subject. LostGalliFreyan wrote: Bill, if you want to hear a 32 kbps MP3 test of a clip about 52 seconds long, I posted it in alt.binaries.test.test "For Bill, a small MP3 at 32 kbps test" The MP3 is there, plus the WAV clip too. I don't know how it will compare with WMA, I'm just exploring MP3 to see if it will even go there, and it's not too shabby. I spent a couple of hours trying to get those files LGF, but had only partial success (and NOT with OE). The collection of WAV fragments, when combined and decoded, came out to 29 seconds, instead of the expected 52 seconds (as in the mp3), for some weird reason (and yes, I tried doing the downloading and combining again, but to no avail (using Xananews, which is about as far as I could go). But at least it was listenable, and so I could at least compare the abbreviated WAV file with the mp3 (which did come out in full). So, so much for downloading binaries; it seems to be a bit problematic at times (i.e., in successfully combining and decoding the multipart fragments). However, since it was in mono, I couldn't do a good comparison with WMA, using a joint stereo, low bitrate, A/B comparison (32 kbps 22 KS joint stereo). However, I did notice that the mp3 was a bit duller (lacked the highs) as compared to the source WAV file, unlike what happened when using 32 kbps 22 KS sampling WMA, as one might expect. But I didn't find an option to convert to a 32 kpbs MONO 22KS file format, only stereo, which renders the comparison useless. So unless you can convert some stereo WAV file (with some music) to both MP3 and WMA formats at 32 kbps joint stereo, 22 KS, to really push the bar, we may be stuck here. Again, I needed the 32 kbps AND stereo for the reprocessed Command Performance broadcasts to get some presence in the file (presence meaning like you feel you are right there in the audience listening to the performances, instead of using just mono). And as I pointed out in the other thread, IF you use a good pseudostereo plug-in like PSP, the "presence" effect is quite good (on headphones), AND it's pretty much mono-compatible, to boot. I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). ----- BTW, if anybody else wants to jump into an audio editing and conversion topic, feel free to do so at any time. :-) I'll post what I know. |
#2
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Audio editing and converting
"Bill in Co" wrote in
news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). I can't follow this up, I need to do other things, but I have to say that I mentioned that. The conditions for a test have to be reasonable, I described in extreme and repeated detail that I was showing that LAME does not default to the best settings for low bitrate encoding, and described a change in filtering that would REDUCE the high frequencies a bit more to gain bits for a clean encode free of artifacting, such that a simple tone control can fix the HF enough to restore legibility. So of course the MP3 sounds a bit duller. Not only would I expect you to notice, I pointed it out, so there's no new finding there. As far as WMA goes, I'll NEVER use it, period. I will never waste my time trying to prove someone else's case for a format I have stated good reasons for me to avoid. Others can do what they want. That means I can do what I want too. All I'm concerned with is that people become aware that MP3 can go further then they likely expect. I did say I thought that it might contend with WMA, but it really doesn't matter. WMA = smaller file = bigger playback overheads. Doesn't matter HOW you cut it, this is true, and people pick the compromises that suit them best. The reason that my finding matters is simple: it offers improvements in the SAME context that many people already choose. |
#3
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Audio editing and converting
"Bill in Co" wrote in
news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). Last poing from me, just closing the circle to where I came in on this. An MP3 encoded slightly 'duller', with less HF, is far less objectionable than artifacting in the mioddle of the audio band. It amounts to listening to a loudspeaker off axis, at worst, something most of us do every day. We don't even care, we don't notice until we decide to consciously test to see what the on-axis sound is like. Now compare that MP3 method to the dreck often posted in files TWICE the size. Come to that, compare ANY encoding wioth the original WAV, at 32 kbps you're going to hear a difference. If WMA could REALLY encode a stereo 32 kbps WAV so well that there is no difference, no- one would be using LAME, never mind lossless compression. Obviously it's not that good. So, never mind the WMA thing which I thing is a red herring given that I started out offering a way to make a LAME encode better than the default offered, for anyone who does want to try the methods of reducing HF as a price to pay for lower artifacting at low bitrates, here are two settings to try: LAME --preset 80 --resample 24 --lowpass 11 (My original, for getting FM broadcast quality from a CD source using only 80 kbps for modern high quality drama that includes complex batural soundtrack and high quality music. Compare this with the defaukle LAME --preset 80, to verify the method is viable! Note the default filter they use, and hear the difference, you won't need to ABX test this, it's very evident.). LAME.EXE --preset 32 --resample 22.05 --lowpass 12 --lowpass-width 6 -mm (Recent modification to test at extreme compression, to 32 kbps. It's designed for sources where HF at 9 KHz is below -48 dB, specifically designed for old mono radio broadcasts, and uses a wide shallow filter that begind slightly ABOVE the Nyquist frequency but still manages to avoid aliasing because the source is already 'dull' as all such old recordings are. A lot of people like those old shows, and most Usenet posts for them are either in files so big that it's silly, or small ones that sound desperately poor, often unlistenable. Worse, some people transcode the bad ones into BIG ones, thinking they'll get better! THAT, ultimately, is the reason for my efforts. There IS a better way, even without abandoning MP3. Last point: The more you can do to restore a sound, or change it usefully, AFTER the encode/decode stages, the smaller the file will be. Analogous to Dolby, there is no reason a method like mine could not have been consistently modelled and applied to ALL lossy formats as a way to save bits for storage and transfer. I have taken that notion further with a method based on noise reduction and harmonic restoration, but even in the simpler EQ form, it has not been used as a standard as far as I know. Like noise shaping, such methods seem to have taken a back seat in favour of psychoacoustics, which in my view was the wrong way to go as it depends more on subtleties we can't know, than on those we can, and was therefore dependent on large volume statistical analyses. I chose methods that do not. WHile it won't make TINY files, I strongly recommend WAVpack's lossy format for those who normally prefer LAME at bitrates of 192 or better. It amazed me when I heard how well it works with no psychoacoustics, and no artifacting of the kind we're normally lumbered with when trying to compress audio. I haven't played around with that like I did with MP3, but there may be scope for it. But I haven't got the time. I'm way to good at wasting that already, which is why I'm closing my side of this talk with a final complete statement on my methods and intent. |
#4
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Audio editing and converting
Lostgallifreyan wrote:
"Bill in Co" wrote in news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). I can't follow this up, I need to do other things, but I have to say that I mentioned that. The conditions for a test have to be reasonable, I described in extreme and repeated detail that I was showing that LAME does not default to the best settings for low bitrate encoding, and described a change in filtering that would REDUCE the high frequencies a bit more to gain bits for a clean encode free of artifacting, such that a simple tone control can fix the HF enough to restore legibility. So of course the MP3 sounds a bit duller. Not only would I expect you to notice, I pointed it out, so there's no new finding there. Sorry, I didn't intend to imply that there was. I was just pointing out my observations. I think you have already made your point that with tweaking you can get some improvements over the default mp3 settings, and that was never in dispute. As far as WMA goes, I'll NEVER use it, period. I will never waste my time trying to prove someone else's case for a format I have stated good reasons for me to avoid. Others can do what they want. That means I can do what I want too. All I'm concerned with is that people become aware that MP3 can go further then they likely expect. I did say I thought that it might contend with WMA, but it really doesn't matter. WMA = smaller file = bigger playback overheads. Doesn't matter HOW you cut it, this is true, and people pick the compromises that suit them best. The reason that my finding matters is simple: it offers improvements in the SAME context that many people already choose. OK. I just thought there was a chance you might want to at least try it, and I was clearly wrong. I did indeed try yours out however. What I found weird was that the 15 part WAV file didn't come out fully intact for some reason, although it appeared to have equal length segments before being combined. |
#5
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Audio editing and converting
Lostgallifreyan wrote:
"Bill in Co" wrote in news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). Last poing from me, just closing the circle to where I came in on this. An MP3 encoded slightly 'duller', with less HF, is far less objectionable than artifacting in the mioddle of the audio band. I agree completely. It amounts to listening to a loudspeaker off axis, at worst, something most of us do every day. We don't even care, we don't notice until we decide to consciously test to see what the on-axis sound is like. Now compare that MP3 method to the dreck often posted in files TWICE the size. Come to that, compare ANY encoding wioth the original WAV, at 32 kbps you're going to hear a difference. If WMA could REALLY encode a stereo 32 kbps WAV so well that there is no difference, no- one would be using LAME, never mind lossless compression. Obviously it's not that good. No, nothing is THAT good. I didn't intend to imply it was. So, never mind the WMA thing which I thing is a red herring given that I started out offering a way to make a LAME encode better than the default offered, for anyone who does want to try the methods of reducing HF as a price to pay for lower artifacting at low bitrates, here are two settings to try: LAME --preset 80 --resample 24 --lowpass 11 (My original, for getting FM broadcast quality from a CD source using only 80 kbps for modern high quality drama that includes complex batural soundtrack and high quality music. Compare this with the defaukle LAME --preset 80, to verify the method is viable! Note the default filter they use, and hear the difference, you won't need to ABX test this, it's very evident.). LAME.EXE --preset 32 --resample 22.05 --lowpass 12 --lowpass-width 6 -mm (Recent modification to test at extreme compression, to 32 kbps. It's designed for sources where HF at 9 KHz is below -48 dB, specifically designed for old mono radio broadcasts, and uses a wide shallow filter that begind slightly ABOVE the Nyquist frequency but still manages to avoid aliasing because the source is already 'dull' as all such old recordings are. A lot of people like those old shows, and most Usenet posts for them are either in files so big that it's silly, or small ones that sound desperately poor, often unlistenable. Worse, some people transcode the bad ones into BIG ones, thinking they'll get better! That is pretty pathetic. But I've seen evidence of that, too, in some of the questions I've seen posted (i.e. some people think they can improve the quality by reencoding the original at a higher bitrate - go figure!). THAT, ultimately, is the reason for my efforts. There IS a better way, even without abandoning MP3. And that was never in dispute. :-) Last point: The more you can do to restore a sound, or change it usefully, AFTER the encode/decode stages, the smaller the file will be. Analogous to Dolby, there is no reason a method like mine could not have been consistently modelled and applied to ALL lossy formats as a way to save bits for storage and transfer. I have taken that notion further with a method based on noise reduction and harmonic restoration, but even in the simpler EQ form, it has not been used as a standard as far as I know. Like noise shaping, such methods seem to have taken a back seat in favour of psychoacoustics, which in my view was the wrong way to go as it depends more on subtleties we can't know, than on those we can, and was therefore dependent on large volume statistical analyses. I chose methods that do not. I don't know why either. I am impressed, however, that you have taken the time to go into all of this stuff on your own. At any rate, I'm sorry if I offended you (by trying to compare mp3 with wma in the quest for best compressibility) which wasn't my intent, but we can leave it at that. I just found those tests to be interesting, and thought I'd share my observations. |
#6
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Audio editing and converting
"Bill in Co" wrote in
news OK. I just thought there was a chance you might want to at least try it, and I was clearly wrong. I did indeed try yours out however. What I found weird was that the 15 part WAV file didn't come out fully intact for some reason, although it appeared to have equal length segments before being combined. No WMA for me. Nope. Never. About the WAV bits, I don't know. I sent it via a btinternet server using PowerPost, which reported it sent ok, and I just looked on the test group with Xnews, and it's all there too. I can't check any further, but if you can't see it all it means some propogation failure beyond my control happened. Anyway, you don'ty need all of it, you heard some, which is enough. I only set it to 52 seconds for a neat break in the stuff being said in the show.. (I might have suggested a good binaries grabber but I don't know of one. I wanted one to rival PowerPost, and there is actually a PowerGrab but it's horrible, only the names seem anything like related to each other. I use Xnews for binaries, except when a freind lets me use an Easynews account for big stuff). |
#7
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Audio editing and converting
"Bill in Co" wrote in
: Lostgallifreyan wrote: "Bill in Co" wrote in news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). Last poing from me, just closing the circle to where I came in on this. An MP3 encoded slightly 'duller', with less HF, is far less objectionable than artifacting in the mioddle of the audio band. I agree completely. It amounts to listening to a loudspeaker off axis, at worst, something most of us do every day. We don't even care, we don't notice until we decide to consciously test to see what the on-axis sound is like. Now compare that MP3 method to the dreck often posted in files TWICE the size. Come to that, compare ANY encoding wioth the original WAV, at 32 kbps you're going to hear a difference. If WMA could REALLY encode a stereo 32 kbps WAV so well that there is no difference, no- one would be using LAME, never mind lossless compression. Obviously it's not that good. No, nothing is THAT good. I didn't intend to imply it was. So, never mind the WMA thing which I thing is a red herring given that I started out offering a way to make a LAME encode better than the default offered, for anyone who does want to try the methods of reducing HF as a price to pay for lower artifacting at low bitrates, here are two settings to try: LAME --preset 80 --resample 24 --lowpass 11 (My original, for getting FM broadcast quality from a CD source using only 80 kbps for modern high quality drama that includes complex batural soundtrack and high quality music. Compare this with the defaukle LAME --preset 80, to verify the method is viable! Note the default filter they use, and hear the difference, you won't need to ABX test this, it's very evident.). LAME.EXE --preset 32 --resample 22.05 --lowpass 12 --lowpass-width 6 -mm (Recent modification to test at extreme compression, to 32 kbps. It's designed for sources where HF at 9 KHz is below -48 dB, specifically designed for old mono radio broadcasts, and uses a wide shallow filter that begind slightly ABOVE the Nyquist frequency but still manages to avoid aliasing because the source is already 'dull' as all such old recordings are. A lot of people like those old shows, and most Usenet posts for them are either in files so big that it's silly, or small ones that sound desperately poor, often unlistenable. Worse, some people transcode the bad ones into BIG ones, thinking they'll get better! That is pretty pathetic. But I've seen evidence of that, too, in some of the questions I've seen posted (i.e. some people think they can improve the quality by reencoding the original at a higher bitrate - go figure!). THAT, ultimately, is the reason for my efforts. There IS a better way, even without abandoning MP3. And that was never in dispute. :-) Last point: The more you can do to restore a sound, or change it usefully, AFTER the encode/decode stages, the smaller the file will be. Analogous to Dolby, there is no reason a method like mine could not have been consistently modelled and applied to ALL lossy formats as a way to save bits for storage and transfer. I have taken that notion further with a method based on noise reduction and harmonic restoration, but even in the simpler EQ form, it has not been used as a standard as far as I know. Like noise shaping, such methods seem to have taken a back seat in favour of psychoacoustics, which in my view was the wrong way to go as it depends more on subtleties we can't know, than on those we can, and was therefore dependent on large volume statistical analyses. I chose methods that do not. I don't know why either. I am impressed, however, that you have taken the time to go into all of this stuff on your own. At any rate, I'm sorry if I offended you (by trying to compare mp3 with wma in the quest for best compressibility) which wasn't my intent, but we can leave it at that. I just found those tests to be interesting, and thought I'd share my observations. No worries. I wasn;t lookign either for agreement or competition, is all, just to put that MP3 thing out because I think I likely won't be the only one who gets something out of it. The whole thing about comparisons and testing gets to the point where I learned to stay with one direct assertion and stay there. The whole debate about LAME settings used to get beyond imagining. I learned to stay well out of it. Part of the reason for my posting of LAME settings is to show that despite YEARS of frenetic wrangling between self- professed gurus and audiophiles, there are STILL easy ways for novices to get better results with it. |
#8
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Audio editing and converting
Lostgallifreyan wrote:
"Bill in Co" wrote in news OK. I just thought there was a chance you might want to at least try it, and I was clearly wrong. I did indeed try yours out however. What I found weird was that the 15 part WAV file didn't come out fully intact for some reason, although it appeared to have equal length segments before being combined. No WMA for me. Nope. Never. About the WAV bits, I don't know. I sent it via a btinternet server using PowerPost, which reported it sent ok, and I just looked on the test group with Xnews, and it's all there too. Yup. All 15 parts show up in the listing. I can't check any further, but if you That's ok. can't see it all it means some propogation failure beyond my control happened. Anyway, you don'ty need all of it, you heard some, which is enough. I only set it to 52 seconds for a neat break in the stuff being said in the show.. No, I did see all 15 parts, and they appeared to be ok from the listings (and with equal reported lengths), but when decoded and combined into one wav file, it came out to be only 29 seconds long when played, and it wasn't contiguous. I mean there were jumps in it (or parts skipped in it) when it was played (really weird). But I got the gist of it, and it was a bit humorous, too. (I might have suggested a good binaries grabber but I don't know of one. I wanted one to rival PowerPost, and there is actually a PowerGrab but it's horrible, only the names seem anything like related to each other. I use Xnews for binaries, except when a freind lets me use an Easynews account for big stuff). |
#9
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Audio editing and converting
Lostgallifreyan wrote:
"Bill in Co" wrote in : Lostgallifreyan wrote: "Bill in Co" wrote in news I will say this, however. The loss of highs was noticeable in your mp3. Not too bad, but definitely noticeable (I mean noticeably duller than the source WAV file). Last poing from me, just closing the circle to where I came in on this. An MP3 encoded slightly 'duller', with less HF, is far less objectionable than artifacting in the mioddle of the audio band. I agree completely. It amounts to listening to a loudspeaker off axis, at worst, something most of us do every day. We don't even care, we don't notice until we decide to consciously test to see what the on-axis sound is like. Now compare that MP3 method to the dreck often posted in files TWICE the size. Come to that, compare ANY encoding wioth the original WAV, at 32 kbps you're going to hear a difference. If WMA could REALLY encode a stereo 32 kbps WAV so well that there is no difference, no- one would be using LAME, never mind lossless compression. Obviously it's not that good. No, nothing is THAT good. I didn't intend to imply it was. So, never mind the WMA thing which I thing is a red herring given that I started out offering a way to make a LAME encode better than the default offered, for anyone who does want to try the methods of reducing HF as a price to pay for lower artifacting at low bitrates, here are two settings to try: LAME --preset 80 --resample 24 --lowpass 11 (My original, for getting FM broadcast quality from a CD source using only 80 kbps for modern high quality drama that includes complex batural soundtrack and high quality music. Compare this with the defaukle LAME --preset 80, to verify the method is viable! Note the default filter they use, and hear the difference, you won't need to ABX test this, it's very evident.). LAME.EXE --preset 32 --resample 22.05 --lowpass 12 --lowpass-width 6 -mm (Recent modification to test at extreme compression, to 32 kbps. It's designed for sources where HF at 9 KHz is below -48 dB, specifically designed for old mono radio broadcasts, and uses a wide shallow filter that begind slightly ABOVE the Nyquist frequency but still manages to avoid aliasing because the source is already 'dull' as all such old recordings are. A lot of people like those old shows, and most Usenet posts for them are either in files so big that it's silly, or small ones that sound desperately poor, often unlistenable. Worse, some people transcode the bad ones into BIG ones, thinking they'll get better! That is pretty pathetic. But I've seen evidence of that, too, in some of the questions I've seen posted (i.e. some people think they can improve the quality by reencoding the original at a higher bitrate - go figure!). THAT, ultimately, is the reason for my efforts. There IS a better way, even without abandoning MP3. And that was never in dispute. :-) Last point: The more you can do to restore a sound, or change it usefully, AFTER the encode/decode stages, the smaller the file will be. Analogous to Dolby, there is no reason a method like mine could not have been consistently modelled and applied to ALL lossy formats as a way to save bits for storage and transfer. I have taken that notion further with a method based on noise reduction and harmonic restoration, but even in the simpler EQ form, it has not been used as a standard as far as I know. Like noise shaping, such methods seem to have taken a back seat in favour of psychoacoustics, which in my view was the wrong way to go as it depends more on subtleties we can't know, than on those we can, and was therefore dependent on large volume statistical analyses. I chose methods that do not. I don't know why either. I am impressed, however, that you have taken the time to go into all of this stuff on your own. At any rate, I'm sorry if I offended you (by trying to compare mp3 with wma in the quest for best compressibility) which wasn't my intent, but we can leave it at that. I just found those tests to be interesting, and thought I'd share my observations. No worries. I wasn;t lookign either for agreement or competition, is all, just to put that MP3 thing out because I think I likely won't be the only one who gets something out of it. The whole thing about comparisons and testing gets to the point where I learned to stay with one direct assertion and stay there. The whole debate about LAME settings used to get beyond imagining. I learned to stay well out of it. Part of the reason for my posting of LAME settings is to show that despite YEARS of frenetic wrangling between self- professed gurus and audiophiles, there are STILL easy ways for novices to get better results with it. And they keep coming out with lame revision updates, but I think I'm still using an older version of 3.93, but I can't recall now. Plus it depends on which program I'm using, as some install a DLL into their own program directories, which overrides the one I have in the system folder. Maybe you should submit some of your presets to the Lame group, and see if they might want to add them (like that one for broadcast AM). |
#10
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Audio editing and converting
"Bill in Co" wrote in
: And they keep coming out with lame revision updates, but I think I'm still using an older version of 3.93, but I can't recall now. Plus it depends on which program I'm using, as some install a DLL into their own program directories, which overrides the one I have in the system folder. Maybe you should submit some of your presets to the Lame group, and see if they might want to add them (like that one for broadcast AM). Oh no.. I wouldn't do that. The moment I do that it quickly becomes insane. I mean, there were just two of us, and there was potential for misunderstanding. Now imagine a bunch of people all with cat avatars on Hydrogenaudio all vying for their place in the pyramid. If a person wants to get shot down in flames, by all means go there, otherwise just try things, based on methods we can all use. LAME's development all but got derailed due to the politicking and such at one point, and the core developers had to put distance between themselves and some of those fighting to establish their ideas of the best preset. The fact that I can show a variation worth trying is as much to do with that as anything else, it's clear;ly not all about real science, and being psychoacoustically based it's hard to be such anyway, as statistical bases are more useful than any 'objective'. In short, I have a better chanmce of beign heard by ONE thoughtful individual here, than there. I also posted the method (and some files made by it) in the radio groups I hung out in for a while. There were no complaints, but I doubt I changed their world either. LAME versions changem, so presets also change. I'm using 3.98.2, and I think those presets may have been the same in v3.93 but I don't know. ABR presets got improved, I know that much, but not what releases were involved in that change at its greatest shift. I explored it mainly because it had improved while I wasn't looking, before. About the WAV file, I grabbed it to see if it worked ok. It did. If yours is 9,289,832 bytes, try for an MD5 checksum of C753CA7E193CFF576D3D1616F91F65EF. I don't need to know the outcome, but you might. It's possible that your client is breaking it, but I don't know. Checking for integrity is all I can think of to try. Got to go... but not as long as Captain Oates. This year I have no appetite for cold weather anyway. |
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