Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
:: :: Sorry, I don't get your point here, what is FS? :: FS = Full Scale = maximum SPL possible for a sine wave. Average Level is usually 10...14 dB below FS for modern music. For any ATH calculation you must assign FS a SPL (Sound Pressure Level). The ATH is different for different assignments, so the bit assignment depends on this assignment. -- Mit freundlichen Grüßen Frank Klemm eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED] phone | +49 (3641) 64-2721home: +49 (3641) 390545 sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
Hi Frank! Frank Klemm schrieb am Sam, 05 Aug 2000: :: :: Sorry, I don't get your point here, what is FS? :: FS = Full Scale = maximum SPL possible for a sine wave. Average Level is usually 10...14 dB below FS for modern music. OK For any ATH calculation you must assign FS a SPL (Sound Pressure Level). our adjustment for the ATH is 114 dB. (see ATH formula in util.c, adjustment in quantize-pvt.c) The ATH is different for different assignments, so the bit assignment depends on this assignment. I think here is a point of misunderstanding. In Layer3 we don't have a bit assignment loop, but a noise allocation loop. For CBR encoding we assign a fixed base amount of bits depending on the desired bitrate and some additional bits from the bitreservoir depending on the perceptual entropy, some output from our psymodel. The next thing to do is allocate noise from scalefactor bands which can carry more noise so that the distortion in other bands gets lowered (scalefactor colouring) and to control the injected noise (quantizer stepsize). Hopefully, in the end all bands have a distortion less our thresholds, but, in case of too few bits, distorted bands will remain. So, if you vary the ATH adjustment, you vary the amount of noise which can be allocated, but only if the ATH is over our masking thresholds from our psychoacoustic model. -- Mit freundlichen Grüßen Frank Klemm eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED] phone | +49 (3641) 64-2721home: +49 (3641) 390545 sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany Ciao Robert -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
What about an option "adjust-level-for-psycho-model", which increases the level for the threshold computation, so low level music is coded with more bits. To my mind low level pieces of music with a turned up volume control are coded with too less bits. lame is coding for a full scale SPL of about 90 dB and that's too less if you are listening at 100 dB full scale SPL. Options like '-b128' don't solve this problem. For medium level music there are still too less bits. Only the low level parts get enough bits, may be also too less. This seems to be an ATH problem. But isn't the ATH adjusted according to the VBR scale? It's right that a switch to manually adjust the ATH would be good, as I also think that some low level pieces of classic music are encoded with too few bits. A rough solution is to completely disable use of the ATH with --noath Regards, -- Gabriel Bouvigne - France [EMAIL PROTECTED] icq: 12138873 MP3' Tech: www.mp3-tech.org -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
:: What about an option "adjust-level-for-psycho-model", which increases the :: level for the threshold computation, so low level music is coded with more :: bits. To my mind low level pieces of music with a turned up volume control :: are coded with too less bits. lame is coding for a full scale SPL of about :: 90 dB and that's too less if you are listening at 100 dB full scale SPL. :: :: Options like '-b128' don't solve this problem. For medium level music :: there :: are still too less bits. Only the low level parts get enough bits, may be :: also too less. ^ oops, too much is right :: :: This seems to be an ATH problem. But isn't the ATH adjusted according to the :: VBR scale? :: That is VBR scale? :: It's right that a switch to manually adjust the ATH would be good, as I also :: think that some low level pieces of classic music are encoded with too few :: bits. :: A rough solution is to completely disable use of the ATH with --noath :: Average Bitrate (fictitious example) Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0180 185 180 175 165 150 -10175 180 185 180 175 165 -20165 175 180 185 180 175 -30150 165 175 180 185 180 -40130 150 165 175 180 185 -50105 130 150 165 175 180 -60 80 105 130 150 165 175 -70 50 80 105 130 150 165 -80 20 50 80 105 130 150 -90 0 20 50 80 105 130 So if you calculate the bitrate for SPL/FS = 90 dB, you have the following deficiencies: Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0- 5 - - - - -10- - - - - - -20- - -5 - - -30- - -5 10 5 -40- - - 10 15 20 -50- - - 15 25 30 -60- - - 20 35 45 -70- - - 25 45 60 -80- - - 25 50 70 -90- - - 30 55 80 So if you calculate the bitrate for SPL/FS = 90 dB and set B_min=100, you have the following deficiencies: Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0- 5 - - - - -10- - - - - - -20- - -5 - - -30- - -5 10 5 -40- - - 10 15 20 -50- - - 15 25 30 -60- - - 20 35 45 -70- - - 25 45 60 -80- - -5 30 50 -90- - - -5 30 Using a bitrate of max (SPL/FS80, SPL/FS90, SPL/FS100, SPL/FS110) you can hear at any volume without audible noise. -- Mit freundlichen Grüßen Frank Klemm eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED] phone | +49 (3641) 64-2721home: +49 (3641) 390545 sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
Frank Klemm schrieb am Sam, 05 Aug 2000: snip :: This seems to be an ATH problem. But isn't the ATH adjusted according to :: the VBR scale? :: That is VBR scale? look out in LAME's sources for a line like: gfc-ATH_lower = (4-gfp-VBR_q)*4.0; here you can see that the ATH is lowered by 16 dB with -V0, -V4 is the default and there you will get the usual ATH as in constant bitrate coding. :: It's right that a switch to manually adjust the ATH would be good, as I also :: think that some low level pieces of classic music are encoded with too few :: bits. :: A rough solution is to completely disable use of the ATH with --noath :: Average Bitrate (fictitious example) Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0 180 185 180 175 165 150 -10 175 180 185 180 175 165 -20 165 175 180 185 180 175 -30 150 165 175 180 185 180 -40 130 150 165 175 180 185 -50 105 130 150 165 175 180 -60 80 105 130 150 165 175 -70 50 80 105 130 150 165 -80 20 50 80 105 130 150 -900 20 50 80 105 130 So if you calculate the bitrate for SPL/FS = 90 dB, you have the following deficiencies: Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0 - 5 - - - - -10 - - - - - - -20 - - -5 - - -30 - - -5 10 5 -40 - - - 10 15 20 -50 - - - 15 25 30 -60 - - - 20 35 45 -70 - - - 25 45 60 -80 - - - 25 50 70 -90 - - - 30 55 80 So if you calculate the bitrate for SPL/FS = 90 dB and set B_min=100, you have the following deficiencies: Level SPL for FS below FS 70 dB 80 dB 90 dB 100 dB 110 dB 120 dB 0 - 5 - - - - -10 - - - - - - -20 - - -5 - - -30 - - -5 10 5 -40 - - - 10 15 20 -50 - - - 15 25 30 -60 - - - 20 35 45 -70 - - - 25 45 60 -80 - - -5 30 50 -90 - - - -5 30 Using a bitrate of max (SPL/FS80, SPL/FS90, SPL/FS100, SPL/FS110) you can hear at any volume without audible noise. Sorry, I don't get your point here, what is FS? -- Mit freundlichen Grüßen Frank Klemm eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED] phone | +49 (3641) 64-2721home: +49 (3641) 390545 sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany Frank, what LAME version are you using? And have you tried to disable that scalefac_scale feature? (If I remember right you are using -q1, leave it out and listen again) Ciao Robert -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
:: -32000Hz files, filled with noise and :: "Lame -q1 -d -m j -V 2 -B 192 --lowpass 12.0 Hobbit.wav :: Hobbit-q1.mp3" :: and :: :: "Lame -h -d -v -q1 jo3.wav jo3q1.wav" :: :: |56 - 3 - 1,8% :: |||64 - 11 - 6,7% :: 80 - 52 - 31,7% :: ||96 - 59 - 36,0% :: |112 - 28 - 17,1% :: ||128 - 5 - 3,0% :: |160 - 4 - 2,4% :: :: is not really what I would use to archive material. could you try :: adding "-b128" and listening how it sounds then? sounds ok :: to me now. :: :: It's just: -q1 is a bit more agressive, but with -b128 (-V1), I never :: have encountered a file that sounds poor. :: :: I like -q1 alot, but as MT said once @ lower bitrate there is a higher :: risk of problems, I just take the -b128 and never any problems. :: What about an option "adjust-level-for-psycho-model", which increases the level for the threshold computation, so low level music is coded with more bits. To my mind low level pieces of music with a turned up volume control are coded with too less bits. lame is coding for a full scale SPL of about 90 dB and that's too less if you are listening at 100 dB full scale SPL. Options like '-b128' don't solve this problem. For medium level music there are still too less bits. Only the low level parts get enough bits, may be also too less. -- Mit freundlichen Grüßen Frank Klemm eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED] phone | +49 (3641) 64-2721home: +49 (3641) 390545 sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re[2]: [MP3 ENCODER] 3.86a, bug with -h ? [re-send]
Hello Stephan, SE Yes! This is also my experience. Especially when the Wav-File contains a SE lot of noise (such as copys from cassetes) SE No! I realy mean 3.85 -q1. And it´s very audible. SE I will give some samples, if anyone tells me where I can upload them. (never SE done before) for everyone: Stephan uploaded the files to: http://r3mix.50g.com I listened, I also think to hear something (slightly, but not 100% sure) weird about the voices sometimes, but -32000Hz files, filled with noise and "Lame -q1 -d -m j -V 2 -B 192 --lowpass 12.0 Hobbit.wav Hobbit-q1.mp3" and "Lame -h -d -v -q1 jo3.wav jo3q1.wav" |56 - 3 - 1,8% |||64 - 11 - 6,7% 80 - 52 - 31,7% ||96 - 59 - 36,0% |112 - 28 - 17,1% ||128 - 5 - 3,0% |160 - 4 - 2,4% is not really what I would use to archive material. could you try adding "-b128" and listening how it sounds then? sounds ok to me now. It's just: -q1 is a bit more agressive, but with -b128 (-V1), I never have encountered a file that sounds poor. I like -q1 alot, but as MT said once @ lower bitrate there is a higher risk of problems, I just take the -b128 and never any problems. -- Best regards, Roelmailto:[EMAIL PROTECTED] -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
