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#121
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Bob Cain wrote:
Now, that's cool and certainly makes the point. How do you make a GIF animation? Well, y'know, I've already uploaded a screenshot of that maximum peak in my Pink Floyd MFSL CD at (1) it's original level (2) at my +4.5dB personally preferred level and (3) at a rudely pumped up +10dB level where the limiting actually becomes visible. Maybe I should animate those as well. I just assumed that everybody could look up, look down and then look straight ahead and see the differences - 'specially since they're lined up perfectly straight one on top of the other. Doh! Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#122
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Geoff Wood wrote:
Foolish not to normalise maybe, but defintiely not to "normalise" - your version which includes applying the compression that you seem to acknowledge as a Bad Thing while similtaneiously explaining that you prefer it . Duhhh.......................... Doh! Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#123
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Geoff Wood wrote:
"Martin Tillman" wrote in message What? You are a troll. However, if normalizing the WAV causes the amplitude of that frequency to be boosted enough that it then becomes audible, it will be retained. Troll, troll, troll. I've spotted you, hehehehe! The real sad thing is that I don't think he is ..... Truth really hurts, don't it? Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#124
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StArSeEd wrote:
Alchemy Mindworks' GIF Construction Set is probably the most popular oldschool GIF animator.. at least it used to be. Microsoft made one too at some point, though as I recall, it wasn't quite as full-featured as the former. No, no, no... All you need is The Gimp. It's like having Alchemy and PhotoShop in one package - and it's free 'cuz it's open source. And it's available for both Linux and Windows. http://www.gimp.org Get it today and never screw with another stupid commercial graphics app unless your job requires you to have to do so. And you don't even have to pay a dime to prevent those stupid annoying Shareware windows from popping up every time you open it up - 'cuz they're never there to start with... 'cuz it's FREE! Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#125
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Geoff Wood wrote:
"Yosah Akbah Muhammed" wrote in message MP3 to 'archiving', is like chopping 10% off all your $ notes and putting them in a draw, while throwing the rest away and calling it 'saving'. Is that a Geoffact or just an opinion? Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#126
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On Mon, 30 Jun 2003 03:22:54 -0500, Lord Hasenpfeffer
wrote: Pete Carney wrote: The first image shows the original at the location of the max peak in the file. Then the second image is proper normalization in CoolEdit 2000 of the exact same time frame. Then the third is what the command line program "Normalize" does to the file. With what parameters? Normalize doesn't just do one thing and nothing else. Weren't you claiming that this was *exactly* what it did with the MFSL disc? That it just increased *every* sample until the peaks were just below 0dB FS, with no other effect *whatever*? -- Stewart Pinkerton | Music is Art - Audio is Engineering |
#127
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On Mon, 30 Jun 2003 03:22:54 -0500, Lord Hasenpfeffer wrote:
Meanwhile, in my "normalized" WAV of MFSL's "Dark Side Of The Moon", there is no limiting present at all even at the greatest peak. There is only textbook normalization there. Wrong. Explain your levels at 11 min and 24.5 min with respect to the original. And for causing that to happen, I'm a *heathen*. Yep. Even moreso because you claimed adamantly that you were not doing it (distorting the dynamic range). But Capitol's version's OK 'cuz everybody's heard of them. No one has said as far as I recall. |
#128
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On Sun, 29 Jun 2003 18:37:05 -0500, Lord Hasenpfeffer wrote:
This is why I normalize before I encode. If you're just talking about boosting amplitude for subsequent playback of the WAV from CD, fine. There is no advantage to that Hey, you've finally seen the light? It's about ****ing time that you've finally realized that I understand this You certainly demonstrated that you absolutely did not believe this earlier. You claimed your normalised DSOTM was better than the original. Of course, what you did wasn't just normalisation... and that CD-audio quality after normalization has nothing at all to do with my hypothesis. Previously... Message-ID: It amazes me that I can rip a WAV directly from an older commercial CD, "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it an MP3 that sounds dramatically better than its own CD source. Which means, if this is true: If you're just talking about boosting amplitude for subsequent playback of the WAV from CD, fine. There is no advantage to that meaning that this is the equivalent up twiddling the knob, then you're saying, given the exact same source, but at different levels, you can make two dramatically different mp3s!! This is just pure nonsense. I suggest you restate exactly what you think you are doing and why, because your story has changed several times over the last few days. |
#129
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On Sun, 29 Jun 2003 02:02:09 -0500, Lord Hasenpfeffer wrote:
What about the Christopher Cross, "Another Page" CD I mentioned early on in the other thread? That "older, quieter" disc required a good 13.33dBs of pumpitude before I was able to do anything reasonable with it. And here are two MP3 samples in a zip archive: http://www.mykec.com/mykec/audio/All_Right.zip Eat that! Pity we can't see the original, or that you didn't encode to a higher bitrate. However, despite that, there is very little doubt in my mind that your 'after' has limiting applied to it in order to make it sound louder. Yep, the only difference between them is that one is louder. Muppet. All I had to do to make them sound the same was to add 2.5dB to the quieter one (and let it clip in a few places). I could post a compilation of the two and challenge you to spot the joins... |
#130
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"Stewart Pinkerton" wrote in message
Which part of 'no effect on masking' did you fail to understand? The threshold of hearing is dependent on sound levels, not on closeness to 0dB FS on the CD, IOW just crank up the amplifier a little. Point of order... The threshold of hearing as usually given *is* a sound level, and therefore can't possibly be dependent on a sound level. |
#131
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#132
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#133
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Martin Tillman wrote:
Previously... Message-ID: It amazes me that I can rip a WAV directly from an older commercial CD, "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it an MP3 that sounds dramatically better than its own CD source. This is true. It does amaze me. Only now I know more about why this is the case. The original CD source is "unnormalized" in my sense of the term. The MP3 is made from a normalized WAV. In in general, most people (including me) tend to believe that louder is better ... because with loudness comes clarity. This remains... an MP3 made from an older, quieter, unnormalized WAV sounds poor to me compared to an MP3 made from a normalized one at the same level of volume. I frequently listen to my MP3s in random shuffle mode. Without "normalization", "remastered MP3s" sound are louder and clearer sounding that "unremastered MP3s". Therefore, it is useful for me to normalize older WAVs so that all of my MP3s have a nice, even loudness. If I don't the older MP3s sound like crap in comparison the newer "remastered" ones. This has become a very boring conversation. meaning that this is the equivalent up twiddling the knob, then you're saying, given the exact same source, but at different levels, you can make two dramatically different mp3s!! This is just pure nonsense. Do as you like. I suggest you restate exactly what you think you are doing and why, because your story has changed several times over the last few days. My mistake: Providing no well-defined hypothesis in this thread. My correction: Providing a well-defined hypothesis in my other thread. So it's been done. Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#134
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Arny Krueger wrote:
"Stewart Pinkerton" wrote in message Which part of 'no effect on masking' did you fail to understand? The threshold of hearing is dependent on sound levels, not on closeness to 0dB FS on the CD, IOW just crank up the amplifier a little. Point of order... The threshold of hearing as usually given *is* a sound level, and therefore can't possibly be dependent on a sound level. To clarify: An ATH chart shows that frequency Y is "inaudible" below amplitude X. In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y is discarded and does not become a part of the final MP3. However, after "normalizing" the WAV by a factor of +3dB, frequency Y's amplitude now becomes X+1dB, therefore, frequency Y is retained and becomes a part of the final MP3. Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#135
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In article ,
Lord Hasenpfeffer wrote: This is true. It does amaze me. Only now I know more about why this is the case. The original CD source is "unnormalized" in my sense of the term. The MP3 is made from a normalized WAV. In in general, most people (including me) tend to believe that louder is better ... because with loudness comes clarity. This remains... an MP3 made from an older, quieter, unnormalized WAV sounds poor to me compared to an MP3 made from a normalized one at the same level of volume. I frequently listen to my MP3s in random shuffle mode. Without "normalization", "remastered MP3s" sound are louder and clearer sounding that "unremastered MP3s". Therefore, it is useful for me to normalize older WAVs so that all of my MP3s have a nice, even loudness. If I don't the older MP3s sound like crap in comparison the newer "remastered" ones. The real question, then, is "Where does the enhanced sense of clarity come from?" Or, conversely, "Why do the MP3 encodings made from lower-level CDs sound poorer to you, and why?" I can see at least two possible hypotheses: Hypothesis 1: the _relative_ amount of frequency content in the two MP3 versions is identical, to within the inevitable low-level errors implied by any coding system. Or, in other words, the lower-level (duller-sounding-to-you) MP3 would sound identical to the higher-level (clearer-sounding) MP3, if you simply turned up the volume a bit. The differences in how you perceive the two aren't due to the actual content of the MP3, only to the playback amplitude. You might be perceiving the two as substantially different either due to the well-known Fletcher-Munson effect: your ears' relative sensitivities to bass and treble fall off faster than their sensitivity to midrange, as the volume is reduced, and thus quieter signals tend to sound as if they lack both bass and treble. Or, it might be a masking and ambient-noise effect. If you typically listen to MP3s in conditions of high ambient noise (office, car, outdoors, etc.) then more of the lower-level-MP3 playback would be drowned out by ambient noise, making it sound less clear. The cure for this would be simply "Turn up the volume during playback." Hypothesis 2: the MP3s encoded from lower-amplitude (non-normalized) WAV inputs are actually, and significantly different than their higher-amplitude cousins in ways other than just amplitude. Possibly some frequencies are missing (excessive masking, or sounds below threshold), possibly some frequency bands are less accurately encoded and have a higher noise level. This would suggest that the MP3 encoder you are using is less than optimal. Possibly it has a poorly-set threshold detector, which is sensitive only to the absolute signal level and not to the relative levels. Possibly one of its other encoder or bit-allocation algorithms is misbehaving, and is dedicating larger portions of the available bit bandwidth to certain frequencies and is being forced to discard other bands for lack of sufficient bit-reservoir in the encoding. The cure for this would be "Use a better encoder, or better settings for the one that you have." -- Dave Platt AE6EO Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior I do _not_ wish to receive unsolicited commercial email, and I will boycott any company which has the gall to send me such ads! |
#136
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![]() An ATH chart shows that frequency Y is "inaudible" below amplitude X. In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y is discarded and does not become a part of the final MP3. However, after "normalizing" the WAV by a factor of +3dB, frequency Y's amplitude now becomes X+1dB, therefore, frequency Y is retained and becomes a part of the final MP3. AH.... I think I see the problem with your understanding of frequency masking (as it was originally called). The threshold of audibility is not a defacto amplitude. It is the amplitude of a one frequency as compared to another frequency. The codec works based on the concept that one loud tone will mask our perception of another quieter tone which very close in frequency to the louder tone. If you have a tone of kHz at -5dB and another tone of 1.001KHz at -20dB, that second tone may be removed from the signal since it is unlikely that anyone could distinguish the presence of the quieter tone. If the kHz tone goes away and the 1.001KHz tone is still at -20dB, the codec will restore the 1.001KHz tone to the signal. as it will now become audible since it is no longer masked by a louder, nearby tone. If, however the second tone is several octaves away from the first, say 200Hz at -20dB, it will still be audible even though it is much quieter. Since there is not another, louder tone nearby to mask it, will be audible. - FLINT |
#137
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![]() Lord Hasenpfeffer wrote: Arny Krueger wrote: "Stewart Pinkerton" wrote in message Which part of 'no effect on masking' did you fail to understand? The threshold of hearing is dependent on sound levels, not on closeness to 0dB FS on the CD, IOW just crank up the amplifier a little. Point of order... The threshold of hearing as usually given *is* a sound level, and therefore can't possibly be dependent on a sound level. To clarify: An ATH chart shows that frequency Y is "inaudible" below amplitude X. In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y is discarded and does not become a part of the final MP3. However, after "normalizing" the WAV by a factor of +3dB, frequency Y's amplitude now becomes X+1dB, therefore, frequency Y is retained and becomes a part of the final MP3. Have you checked with the Lame developers to see if this is how the ATH is applied or if it is applied after the block is normalized internally? Bob -- "Things should be described as simply as possible, but no simpler." A. Einstein |
#138
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On Tue, 01 Jul 2003 20:48:05 -0500, Lord Hasenpfeffer
wrote: Arny Krueger wrote: "Stewart Pinkerton" wrote in message Which part of 'no effect on masking' did you fail to understand? The threshold of hearing is dependent on sound levels, not on closeness to 0dB FS on the CD, IOW just crank up the amplifier a little. Point of order... The threshold of hearing as usually given *is* a sound level, and therefore can't possibly be dependent on a sound level. To clarify: An ATH chart shows that frequency Y is "inaudible" below amplitude X. In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y is discarded and does not become a part of the final MP3. However, after "normalizing" the WAV by a factor of +3dB, frequency Y's amplitude now becomes X+1dB, therefore, frequency Y is retained and becomes a part of the final MP3. We are well aware of what you're talking about, but there are two problems with this: 1) The reference is to *sound level*, not amplitude on the CD. As previously noted, just turn up the volume. 2) Have you checked that this is how your MP3 encoder actually works, rather than on time and frequency-proximate masking (which is a completely different matter)? -- Stewart Pinkerton | Music is Art - Audio is Engineering |
#139
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On Tue, 01 Jul 2003 20:38:11 -0500, Lord Hasenpfeffer
wrote: Martin Tillman wrote: Previously... Message-ID: It amazes me that I can rip a WAV directly from an older commercial CD, "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it an MP3 that sounds dramatically better than its own CD source. This is true. It does amaze me. Only now I know more about why this is the case. The original CD source is "unnormalized" in my sense of the term. The MP3 is made from a normalized WAV. In in general, most people (including me) tend to believe that louder is better ... because with loudness comes clarity. No, with loudness comes compression, and a *false* impression of clarity given by the restricted dynamics of the reproduction. Radio stations have been doing this for decades. This remains... an MP3 made from an older, quieter, unnormalized WAV sounds poor to me compared to an MP3 made from a normalized one at the same level of volume. I frequently listen to my MP3s in random shuffle mode. Without "normalization", "remastered MP3s" sound are louder and clearer sounding that "unremastered MP3s". Therefore, it is useful for me to normalize older WAVs so that all of my MP3s have a nice, even loudness. If I don't the older MP3s sound like crap in comparison the newer "remastered" ones. This has become a very boring conversation. This is true................ Basically, mix 'em up any way *you* like, but please refrain from coming on here claiming that you've discovered some wonder product, and 'whopped the ass of MFSL', 'cos both statements are *wayyy* wide of the mark. What you're talking about has been known for decades, and is just the sort of barbarity that MFSL deliberately tried to avoid. -- Stewart Pinkerton | Music is Art - Audio is Engineering |
#141
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Dave Platt wrote:
I can see at least two possible hypotheses: Dave, I have drawn a picture. Please click the on the link provided below and tell me which of your two hypotheses are best described by what you see. http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif Myke Hypothesis 1: the _relative_ amount of frequency content in the two MP3 versions is identical, to within the inevitable low-level errors implied by any coding system. Or, in other words, the lower-level (duller-sounding-to-you) MP3 would sound identical to the higher-level (clearer-sounding) MP3, if you simply turned up the volume a bit. The differences in how you perceive the two aren't due to the actual content of the MP3, only to the playback amplitude. You might be perceiving the two as substantially different either due to the well-known Fletcher-Munson effect: your ears' relative sensitivities to bass and treble fall off faster than their sensitivity to midrange, as the volume is reduced, and thus quieter signals tend to sound as if they lack both bass and treble. Or, it might be a masking and ambient-noise effect. If you typically listen to MP3s in conditions of high ambient noise (office, car, outdoors, etc.) then more of the lower-level-MP3 playback would be drowned out by ambient noise, making it sound less clear. The cure for this would be simply "Turn up the volume during playback." Hypothesis 2: the MP3s encoded from lower-amplitude (non-normalized) WAV inputs are actually, and significantly different than their higher-amplitude cousins in ways other than just amplitude. Possibly some frequencies are missing (excessive masking, or sounds below threshold), possibly some frequency bands are less accurately encoded and have a higher noise level. This would suggest that the MP3 encoder you are using is less than optimal. Possibly it has a poorly-set threshold detector, which is sensitive only to the absolute signal level and not to the relative levels. Possibly one of its other encoder or bit-allocation algorithms is misbehaving, and is dedicating larger portions of the available bit bandwidth to certain frequencies and is being forced to discard other bands for lack of sufficient bit-reservoir in the encoding. The cure for this would be "Use a better encoder, or better settings for the one that you have." -- -================================- Windows...It's rebootylicious!!! -================================- |
#142
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Stewart Pinkerton wrote:
To clarify: An ATH chart shows that frequency Y is "inaudible" below amplitude X. In a given WAV, frequency Y's amplitude is X-2dB, therefore, frequency Y is discarded and does not become a part of the final MP3. However, after "normalizing" the WAV by a factor of +3dB, frequency Y's amplitude now becomes X+1dB, therefore, frequency Y is retained and becomes a part of the final MP3. We are well aware of what you're talking about, but there are two problems with this: 1) The reference is to *sound level*, not amplitude on the CD. As previously noted, just turn up the volume. Where is the volume knob on my MP3 encoder? 2) Have you checked that this is how your MP3 encoder actually works, rather than on time and frequency-proximate masking (which is a completely different matter)? I am absolutely certain that MP3 and ATRAC employ (1) masked frequency and (2) Absolute Threshold of Hearing filtration methods. Frequencies which are masked are removed AND frequencies which are too quiet to be heard are removed. My concerns lie NOT with masked frequencies. My purpose in normalizing WAVs prior to encoding them is to assist them in successfully passing through the A.T. of H. filter so that they end up as part of the final MP3 rather than on the cutting room floor. If the A.T. of H. filter prevents the under-amplified frequencies from getting across and into the resultant MP3, you can playback that MP3 all you want at whatever volume you want and those frequencies will still be absent from the file. Here's a pictu http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#143
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Stewart Pinkerton wrote:
No, with loudness comes compression, and a *false* impression of clarity given by the restricted dynamics of the reproduction. Radio stations have been doing this for decades. Radio stations have not been encoding MP3s for decades! Basically, mix 'em up any way *you* like, but please refrain from coming on here claiming that you've discovered some wonder product, and 'whopped the ass of MFSL', 'cos both statements are *wayyy* wide of the mark. What you're talking about has been known for decades, and is just the sort of barbarity that MFSL deliberately tried to avoid. http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#144
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![]() "Lord Hasenpfeffer" wrote in message My concern with creating better MP3s from louder WAVs lies not with frequency masking but with filtration based upon what it known as the Absolute Threshold of Hearing. For the benefit of those not following the r.a.p thread. The (now) famous "Absolute Threshold Of Hearing" relates to signals at extremely low levels that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB and lower. geoff |
#145
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Geoff Wood wrote:
Radio stations have been *restricting dynamics* for decades. http://www.mykec.com/mykec/images/Su...Sunday_012.gif 0) Original WAV 1) "Normalize" (default -12dBFS) 2) "Normalize" (default -10dBFS) Where is the evidence of this gawd-awful *restriction of dynamics* of which you accuse me of wreaking against my WAVs? Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#146
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Geoff Wood wrote:
For the benefit of those not following the r.a.p thread. The (now) famous "Absolute Threshold Of Hearing" relates to signals at extremely low levels that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB and lower. Well, thank God, somebody finally cited some hard numbers as opposed to hard opinions regarding the Absolute Threshold of Hearing! Sheesh!!! What was so hard about that? And what is your source for this information? Or are you just picking them out of the blue? -60dB in relation to what? Full Scale? Of course this value fluctuates higher and lower depending on frequency value. It's definitely *not* a straight line even though I drew it to look like one in my "Good_MP3_Bad_MP3.gif". Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#147
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Geoff Wood wrote:
For the benefit of those not following the r.a.p thread. The (now) famous "Absolute Threshold Of Hearing" relates to signals at extremely low levels that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB and lower. It's nice to see you talking more on the same level with me here. Maybe we're finally beginning to see eye-to-eye just a little bit better. Reading your quote above sure beats having to read about how no frequecies magically or disappear as a result of the "normalization" - which I've never said nor even implied. http://www.mykec.com/mykec/images/Good_MP3_Bad_MP3.gif Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#148
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how no frequecies magically or disappear as a result of the
should have been: "how no frequecies magically appear or disappear as a result of the" Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#149
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"Lord Hasenpfeffer" wrote in message
Geoff Wood wrote: For the benefit of those not following the r.a.p thread. The (now) famous "Absolute Threshold Of Hearing" relates to signals at extremely low levels that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB and lower. Well not really but kinda-sorta. The usually-given threshold of hearing is the "0 dB" line on charts like this one: http://www.webervst.com/fm.htm However, due to masking the just noticeable levels for reliable perception in normal day-to-day circumstances are way far higher. Well, thank God, somebody finally cited some hard numbers as opposed to hard opinions regarding the Absolute Threshold of Hearing! Sheesh!!! What was so hard about that? And what is your source for this information? Or are you just picking them out of the blue? For a real thrill, search google for "Fletcher Munson Curve". -60dB in relation to what? Full Scale? Actually, the threshold of hearing is closer to 0 dB SPL @ 1000 Hz, and 0 dB SPL is defined in terms of physical units relevant to acoustics (pressure over area). Of course this value fluctuates higher and lower depending on frequency value. Well doooh! It's definitely *not* a straight line even though I drew it to look like one in my "Good_MP3_Bad_MP3.gif". I can't believe how much bandwidth and energy has been used up with this tiring discussion when a few minutes of research with google would provide most of the relevant facts. |
#150
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"Lord Hasenpfeffer" wrote in message
KikeG wrote: Agreed that AAC is likely the best of the bunch. True. And ATRAC would be at the bottom. You're implying that ATRAC is worse than MP3? High bitrate MP3 can be really pretty good, expecially compared to low bitrate ATRAC. If you want to listen for yourself, please see: http://www.pcabx.com/product/coder_decoder/index.htm and http://www.pcabx.com/product/mds-jb920/index.htm You might want to start looking at this site starting with its home page at www.pcabx.com . |
#151
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![]() "Lord Hasenpfeffer" wrote in message ... Geoff Wood wrote: For the benefit of those not following the r.a.p thread. The (now) famous "Absolute Threshold Of Hearing" relates to signals at extremely low levels that are disgarded in MP3 and ATRAC coding. Not levels like -1dB, or -10dB, or -20dB, but in the order of -60dB and lower. Well, thank God, somebody finally cited some hard numbers as opposed to hard opinions regarding the Absolute Threshold of Hearing! Sheesh!!! What was so hard about that? And what is your source for this information? Or are you just picking them out of the blue? Out of the blue. -60dB in relation to what? Full Scale? Either FS or the wavfile peak level, which in most circumstance will be not too dissimilar. As long as the peak isn't bizarrely low. geoff |
#152
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Arny Krueger wrote:
You might want to start looking at this site starting with its home page at www.pcabx.com . Thanks Arny. I've bookmarked all three pages. I'll look at 'em closer ASAP. It's beginning to look a lot like Usenet... the one I used to know... ![]() Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#153
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On Wed, 02 Jul 2003 06:15:29 -0500, Lord Hasenpfeffer
wrote: Arny Krueger wrote: The usually-given threshold of hearing is the "0 dB" line on charts like this one: http://www.webervst.com/fm.htm Ba-da-bing!! That's definitely right on track with what I've been trying to find. God heavens! Where did you look? Really? This is so basic for most of us around here, that it is hard to envision somebody not knowing it. You cannot open any book, however basic, on hearing that does not start of with these facts! Do you hate books that much, Myke? If not, your public library could probably supply you with a book that takes you thru this a lot faster than it takes to ask all these questions on Usenet. Per. |
#154
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Lord Hasenpfeffer wrote in message ...
You're implying that ATRAC is worse than MP3? Yes. And not only ATRAC3 long play modes at 132 kbps and below. Best quality, standard 292 kbps ATRAC, is quite inferior to a LAME 3.90.2 200 kbps variable bitrate MP3, according to my blind tests: see first and last post at http://www.hydrogenaudio.org/index.p...bb10a72975fb36 One of the ATRAC compression artifacts, the pre-echo of the castanets sample, is blantantly evident. LAME 3.90.2, 3.90.3 or 3.91 with the --alt-preset standard commandline are far better in this particular sample, being transparent for many untrained people. Artifacts in the other samples are more subtle, but LAME still does better in all of them. I'd dare to say that even a 192 kbps constant bitrate, joint-stereo LAME mp3 is probably better than 292 kbps ATRAC. |
#155
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Per Stromgren wrote:
Do you hate books that much, Myke? I own a huge personal library of programming manuals persuant to my purpose as a web-designer and Linux system/database administrator, etc.. Too many demands on my time for programming related needs throughout the past 4 years have not left time for any other kinds of book-related studies. My needs along these lines are very few and far-between. If not, your public library could probably supply you with a book that takes you thru this a lot faster than it takes to ask all these questions on Usenet. Well, the chart isn't the answer to everything I'm trying to find. It's just a good, solid clue towards the solution - much closer and a *lot* more helpful than all that other sometimes obvious stuff that's been being discussed which while possible true is completely beside the point and tangential at best. Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#156
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On Tue, 01 Jul 2003 20:38:11 -0500, Lord Hasenpfeffer wrote:
Martin Tillman wrote: Previously... Message-ID: It amazes me that I can rip a WAV directly from an older commercial CD, "normalize" it to 2dB beyond zero (i.e. -10dBFS) and then encode from it an MP3 that sounds dramatically better than its own CD source. This is true. It does amaze me. Only now I know more about why this is the case. The original CD source is "unnormalized" in my sense of the term. The MP3 is made from a normalized WAV. Your definition of normalise is flawed, as is your practice of it. In in general, most people (including me) tend to believe that louder is better ... because with loudness comes clarity. Louder is just louder. Depending on the circumstances louder is either better or worse than quieter. Therefore, 'louder is better' is actually totally meaningless. This remains... an MP3 made from an older, quieter, unnormalized WAV sounds poor to me compared to an MP3 made from a normalized one at the same level of volume. I've proved you completely wrong with your Chris Cross example. Care to listen to my proof? I frequently listen to my MP3s in random shuffle mode. Without "normalization", "remastered MP3s" sound are louder and clearer sounding that "unremastered MP3s". Therefore, it is useful for me to normalize older WAVs so that all of my MP3s have a nice, even loudness. If I don't the older MP3s sound like crap in comparison the newer "remastered" ones. Faulty terminology, faulty logic, faulty use of the software. Hint: There is nothing fundamentally wrong with true normalistion, nor the desire to to have certain tracks sound more or less as loud as other certain tracks, under certain circumstances. However, buggering the dynamic range is totally wrong if you care about music, and even more wrong when you don't realise you're doing it. Hint 2: Do a search for 'Replay Gain'. If you can understand Replay Gain you will be a better person. (Shock Horror! Replay Gain will require that you normalise many of your tracks to around 6-10dB below maximum!!!!!) |
#157
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![]() Lord Hasenpfeffer wrote: Actually, flint, what you've just described *is* frequency masking which is only one of 2 different filtration methods employed by lossy compression methods such as MP3, AAC, MiniDisc, etc.. Just a minor terminology quibble, Myke. Filtration is what is done to sewage. I'm an old E.E. with a DSP upgrade and have never heard the word applied to signals. "Filtering" is what you want. Bob -- "Things should be described as simply as possible, but no simpler." A. Einstein |
#158
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On Wed, 02 Jul 2003 04:46:35 -0500, Lord Hasenpfeffer wrote:
Geoff Wood wrote: Radio stations have been *restricting dynamics* for decades. http://www.mykec.com/mykec/images/Su...Sunday_012.gif 0) Original WAV 1) "Normalize" (default -12dBFS) 2) "Normalize" (default -10dBFS) Where is the evidence of this gawd-awful *restriction of dynamics* of which you accuse me of wreaking against my WAVs? Right in front of your eyes! It shows absolutely that you are compressing the original. If you can't see it you really don't know what you are looking for. Clues: Left channel peaks at .55 and .58 mins retain relationship to lower peaks. No compression. Left channel peak just before 2.5. Look at the slightly lower peak just before that one on top of waveform. See how it gets closer to the value of the higher peak on the two higher level waveforms. What do you think we could call this phenomenon? See also grosser example on left channel lower waveform at 4.37 and 4.55. Thanks for providing such a classic example of how wrong you are. While there is no scale, I can guess what it is, and this also shows that the highest peaks in the original are pretty close to full scale, perhaps a couple of dBs down, indicating that there is precious little room for REAL normalisation to maximum level anyway. |
#159
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Bob Cain wrote:
Just a minor terminology quibble, Myke. Filtration is what is done to sewage. I'm an old E.E. with a DSP upgrade and have never heard the word applied to signals. "Filtering" is what you want. Hahahaha!!! As they say, "my bad!" :-D Well here's what my gDict for Linux has to say about this issue: Filtration Fil*tra"tion, n. Cf. F. filtration. The act or process of filtering; the mechanical separation of a liquid from the undissolved particles floating in it. So, I guess we're both right! ![]() The answer is clearly "yes and no"! ;-) Myke -- -================================- Windows...It's rebootylicious!!! -================================- |
#160
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Martin Tillman wrote:
Where is the evidence of this gawd-awful *restriction of dynamics* of which you accuse me of wreaking against my WAVs? Right in front of your eyes! It shows absolutely that you are compressing the original. If you can't see it you really don't know what you are looking for. Yeah, I believe I'm on record as having stated there is some slight *limiting* going on there to avoid clipping. Note however too that this is the loudest song on the CD. All other songs would not suffer from any limiting at all because of this +5dB boost in RMS level. 5 or so peaks being slightly limited in a single WAV does not offset the advantages of boosting the average RMS level of the entire file. It is a compromise I am perfectly willing to accept. Myke -- -================================- Windows...It's rebootylicious!!! -================================- |