Re: Re[2]: [MP3 ENCODER] Voice encoding questions
Frank Klemm schrieb am Sam, 05 Aug 2000: We should support an option (-ma for Mode Auto) which switches between -a -mm for highly correlated channels (r 0.98 = mono), -mj for a normal correlated signals (r = -1.00...-0.20, 0.20...0.91 = stereo) and -ms for nearly not correlated signals (dual channel audio with independent audio, i.e. movies with english/german audio track , r=-0.20...+0.20). The joint stereo coding (-mj) in LAME switches automatically between Stereo and Mid-Side Stereo. Uncorrelated signals will be LR Stereo coded and correlated parts of your waves in MS stereo. Given L=left channel and R=right channel: M = (L+R)/SQRT2 S = (L-R)/SQRT2 note: to get your left right channels back: L = (M+S)/SQRT2 R = (M-S)/SQRT2 As you can see, if your input signal is mono (L=R), only the mid channel carries information, the side channel is empty. The difference to a true mono coding in this situation is, that we now need some bits for our empty side channel which we could use in mono mode for the mid channel too. My observations on old mono like sounds are, that it is a bad idea to let LAME make the side channel really empty. If this happens, it is likely to get an audible glitch. There are a lot of MP3s out there with mono recordings coded with -mj and also -ms. -- Mit freundlichen Grüßen Frank Klemm PS: What's the difference between '-mm' and '-mm -a' ? 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: [MP3 ENCODER] Voice encoding questions
F We should support an option (-ma for Mode Auto) which switches F between -a -mm for highly correlated channels (r 0.98 = F mono), -mj for a normal correlated signals (r = -1.00...-0.20, F 0.20...0.91 = stereo) and -ms for nearly not I am afraid most of decoders can't treat an mp3 file correctly whose mode(stereo - mono) is changing during one file. Switching between any stereo modes (stereo, m/s, is, ms and is) is allowed, but switching between stereo, mono and dual is forbidden by the standard. 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. 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/ )
[MP3 ENCODER] VBR modes question
I've got a question about VBR (the full scale one) Will the mt code completely replace the rh code (in several releases)? Or do you think they will continue to coexist? Do you think that they sould be both documented in the doc? Mark, Robert, anyone else, any opinion about this? Regards, -- Gabriel Bouvigne - France [EMAIL PROTECTED] icq: 12138873 MP3' Tech: www.mp3-tech.org -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] Decoding with lame?
Hi Harald Thanks for your comments. At the moment I am not quite shure what the best implementation would be. My application presently takes the line input data from the MM API and I can save it to a temp wav file (44KHz). I want to be able to play it back at any time if so desired. When I want to save (compress or encode) the temp file, I can now use the ACM or the LAME dll, works fine. My intentions are: 1) instead of using a temp wave file directly encode the line input to an MP3 file 2) to be able to playback (decode) the previously encoded MP3 file from inside the application 3) to be able to stream to a socket in the future I am not sure how this is best implemented in the lame dll, but it would be very useful. Albert Faber has implemented the record/playback option in his CDex program. Not shure how he did it. Thanks for your offer anyway Helmut - Original Message - From: Harald Niesche [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, 3 August 2000 13:28 Subject: Re: [MP3 ENCODER] Decoding with lame? What kind of API do you have in mind? There is some code in Lame that can read from several kinds of input files and delivers a buffer full of samples (actually, a left buffer and a right buffer). Would you like that, some kind of pull interface? You open the stream and then pull frames out of it as long as there are more - that would mean I just have to write a small wrapper around the original APIs and export it from the DLL. I am thinking of lameOpen, lameRead and lameClose with some options or even some intelligence that deduces the type of file automagically. More involved would be the ability to have a push interface, where you deliver some part of the stream (MP3, raw, .wav, whatever) when you feel like it, the encoder engine encodes as much data as possible and returns to you, waiting for you to deliver more data. Harald - Original Message - From: "Helmut Strickner" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 03, 2000 3:33 AM Subject: Re: [MP3 ENCODER] Decoding with lame? The question was regarding the decoding with the dll verion of lame. As Harald Niesche has observed there is no code for decoding in the dll version. It would really be very useful to implement the decode option into the dll. Anyone willing to have a go? Helmut :: I have read some postings on this list about the possibility to use lame for :: decoding. Is it possible, and if true does that apply to the exe version :: only?. I have not found any clue about setting the lame dll for 'decoding of :: an mp3 file' option. I would like to encode and play back without resorting :: to winamp. :: lame --decode Source.mp3 Destination.wav lame --decode Source.mp3 - | Program_to_read_wavs_from_stdin ... -- 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/ ) -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ ) -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ ) -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
[MP3 ENCODER] DC cancelation, Live MP3s without gaps
1st problem ~~~ I have a set of WAV files without silence gaps. After converting to MP3 and reconverting to WAV there are gaps (20...40 ms) between the files. How can I prevent this? 2nd problem ~~~ Some of my CDs have a lot of DC offset. Other are well DC free. Examples: x[AC] y[AC] r typex[DC] y[DC] File 26.110% 30.231% 37.382% MS-Stereo -8.418% -6.779% Giora Feidman/track01.cdda.wav 24.437% 26.815% 62.902% MS-Stereo -7.890% -6.384% Giora Feidman/track02.cdda.wav 28.233% 17.670% 88.636% MS-Stereo -7.832% -5.380% Giora Feidman/track03.cdda.wav 44.400% 42.320% 77.321% MS-Stereo -10.005% -7.835% Giora Feidman/track04.cdda.wav 27.438% 29.094% 74.308% MS-Stereo -7.997% -6.360% Giora Feidman/track05.cdda.wav 38.190% 38.762% 73.798% MS-Stereo -8.416% -6.575% Giora Feidman/track06.cdda.wav 33.937% 35.373% 52.249% MS-Stereo -9.008% -7.106% Giora Feidman/track07.cdda.wav 25.220% 24.489% 83.892% MS-Stereo -7.981% -6.185% Giora Feidman/track08.cdda.wav 23.899% 26.594% 76.254% MS-Stereo -8.135% -6.559% Giora Feidman/track09.cdda.wav 27.257% 29.149% 73.527% MS-Stereo -8.739% -7.057% Giora Feidman/track10.cdda.wav 44.419% 36.258% 79.641% MS-Stereo -9.816% -7.550% Giora Feidman/track11.cdda.wav 22.294% 31.883% 89.198% MS-Stereo -8.411% -7.360% Giora Feidman/track12.cdda.wav 37.214% 35.780% 81.549% MS-Stereo -9.638% -7.372% Giora Feidman/track13.cdda.wav 33.170% 41.641% 87.566% MS-Stereo -9.481% -7.850% Giora Feidman/track14.cdda.wav 33.711% 42.198% 88.124% MS-Stereo -9.345% -7.669% Giora Feidman/track15.cdda.wav 43.713% 45.161% 73.291% MS-Stereo -9.884% -7.836% Giora Feidman/track16.cdda.wav 36.152% 34.970% 73.292% MS-Stereo -9.071% -7.062% Giora Feidman/track17.cdda.wav 37.865% 34.517% 71.336% MS-Stereo -9.279% -7.062% Giora Feidman/track18.cdda.wav 34.980% 39.203% 49.169% MS-Stereo -8.949% -7.188% Giora Feidman/track19.cdda.wav 29.621% 32.464% 81.196% MS-Stereo -8.881% -7.007% Giora Feidman/track20.cdda.wav 70.782% 68.893% 53.126% MS-Stereo -0.006% -0.008% Jazz Lyrik Prosa/track01.cdda.wav 39.614% 45.291% 81.413% MS-Stereo -0.006% -0.007% Jazz Lyrik Prosa/track02.cdda.wav 71.708% 73.715% 99.879% Mono-0.010% -0.010% Jazz Lyrik Prosa/track03.cdda.wav 31.473% 32.412% 99.844% Mono-0.007% -0.007% Jazz Lyrik Prosa/track04.cdda.wav 71.731% 73.783% 99.925% Mono-0.005% -0.005% Jazz Lyrik Prosa/track05.cdda.wav 36.925% 37.863% 99.791% Mono-0.007% -0.007% Jazz Lyrik Prosa/track06.cdda.wav 28.091% 28.811% 99.835% Mono-0.006% -0.006% Jazz Lyrik Prosa/track07.cdda.wav 51.676% 52.912% 99.507% Mono-0.008% -0.008% Jazz Lyrik Prosa/track08.cdda.wav 48.875% 49.597% 81.123% MS-Stereo -0.007% -0.007% Jazz Lyrik Prosa/track09.cdda.wav 55.150% 56.083% 99.873% Mono-0.001% -0.001% Jazz Lyrik Prosa/track10.cdda.wav 47.875% 49.324% 99.749% Mono-0.007% -0.007% Jazz Lyrik Prosa/track11.cdda.wav 85.609% 91.884% 60.015% MS-Stereo -0.005% -0.004% Jazz Lyrik Prosa/track12.cdda.wav 31.389% 38.746% 86.307% MS-Stereo -0.007% -0.007% Jazz Lyrik Prosa/track13.cdda.wav 76.018% 74.017% 80.323% MS-Stereo -0.008% -0.008% Jazz Lyrik Prosa/track14.cdda.wav 41.697% 52.852% 75.873% MS-Stereo -0.006% -0.006% Jazz Lyrik Prosa/track15.cdda.wav 68.815% 71.073% 99.863% Mono-0.009% -0.009% Jazz Lyrik Prosa/track16.cdda.wav 42.912% 43.491% 98.428% Mono-0.006% -0.007% Jazz Lyrik Prosa/track17.cdda.wav 101.106% 99.551% 63.043% MS-Stereo -0.014% -0.020% Jazz Lyrik Prosa/track18.cdda.wav 57.308% 60.230% 90.311% MS-Stereo -0.007% -0.007% Jazz Lyrik Prosa/track19.cdda.wav 87.427% 82.443% 40.229% MS-Stereo -0.007% -0.009% Jazz Lyrik Prosa/track20.cdda.wav 46.851% 47.622% 87.358% MS-Stereo -0.007% -0.007% Jazz Lyrik Prosa/track21.cdda.wav 69.743% 77.524% 71.732% MS-Stereo -0.007% -0.007% Jazz Lyrik Prosa/track22.cdda.wav Problem No.1 is that it creaks Kr#[\/§$%!°^z! terrible on the start and the end of such a file. Problem No.2 is that at the same bitrate DC decreases sound quality (is this a feature of MP3 or a bug in Lame?) I don't like high pass filtering in the frequency domain (don't reduces DC at the beginning of the file, may be removes wanted frequencies), I often do high pass filtering in the polynom domain (Legendre). 0nd order polynom filtering is DC canceling, 1st order is an additional trend elimination, ... A huge disadvantage is that you must read the whole file before
Re: [MP3 ENCODER] DC cancelation
2nd problem ~~~ Some of my CDs have a lot of DC offset. Other are well DC free. Examples: What is DC? -- Gabriel Bouvigne - France [EMAIL PROTECTED] icq: 12138873 MP3' Tech: www.mp3-tech.org -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] DC cancelation, Live MP3s without gaps
Hello Frank, Saturday, August 05, 2000, 1:11:36 PM, you wrote: FK I have a set of WAV files without silence gaps. After converting to MP3 and reconverting to FK WAV there are gaps (20...40 ms) between the files. How can I prevent this? http://albumid.cjb.net has exactly that in mind. -- Best regards, Roelmailto:[EMAIL PROTECTED] -- 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] Voice encoding questions
:: Frank Klemm schrieb am Sam, 05 Aug 2000: :: We should support an option (-ma for Mode Auto) which switches between -a -mm :: for highly correlated channels (r 0.98 = mono), -mj for a normal :: correlated signals (r = -1.00...-0.20, 0.20...0.91 = stereo) and -ms for nearly :not :: correlated signals (dual channel audio with independent audio, i.e. movies :: with english/german audio track , r=-0.20...+0.20). :: :: The joint stereo coding (-mj) in LAME switches automatically between Stereo and :: Mid-Side Stereo. Uncorrelated signals will be LR Stereo coded and correlated :: parts of your waves in MS stereo. Given L=left channel and R=right channel: :: M = (L+R)/SQRT2 :: S = (L-R)/SQRT2 :: :: note: to get your left right channels back: :: L = (M+S)/SQRT2 :: R = (M-S)/SQRT2 :: That's clear. :: As you can see, if your input signal is mono (L=R), only the mid channel :: carries information, the side channel is empty. :: Take some mono recordings and prove this. It's very seldom that L=R. See also alt.binaries.sounds.mp3.* . FM radio mono recordings (tuner set to mono): * differences between the channels from the MPX decoder - AD converter FM radio mono recordings (tuner set to stereo, p.e. News): * differences between the channels from the MPX decoder - AD converter * additional noise, distortion, whistle, ... in the X signal Mono CDs: * Both channels are converted by different AD converters with different parameters (offset, amplification). Records: * a lot of noise and rumble :: The difference to a true mono coding in this situation is, that we now :: need some bits for our empty side channel which we could use in mono :: mode for the mid channel too. :: I've never seen a True Mono Coding. It is mono (historic reasons), it sounds like mono, statistic says it's mono, but L != R. Example: CD: Jazz - Lyrik - Prosa No: (Amiga 74321326192) Title:My Bonnie is over the Ocean Interpreter: Jazz-Optimisten Berlin Length: 11239032 samples [19114 CD frames, 4:14.64] Correlation: r = 0,99879 Coder: Lame: 3.86 alpha Options: -V0 -d -q1 --cwlimit 11.5 -X6 Results: -mm 3642584 bytes 114.3 kbps -mm -a 3642584 bytes 114.3 kbps (bitwise identically with -mm) -mj 5637588 bytes 177.0 kbps (+55%) -ms 7223049 bytes 226.7 kbps (+98%) :: :: My observations on old mono like sounds are, that it is a bad idea to :: let LAME make the side channel really empty. If this happens, it is likely :: to get an audible glitch. :: -mm Use Mono -mi Use Intensity Stereo, MS-Stereo and LR-Stereo -mj Use MS-Stereo and LR-Stereo -ms Use LR-Stereo -ma Analyze FIle before any converting, select -mm, -mj or -ms Another question: Is there any tool to analyze the number of SI, MS and LR frames in a MP3? -- 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: [MP3 ENCODER] VBR modes question
Hi Gaby! I've got a question about VBR (the full scale one) Will the mt code completely replace the rh code (in several releases)? Or do you think they will continue to coexist? I think they will coexist for a while. Do you think that they sould be both documented in the doc? A note about them will be useful. Mark, Robert, anyone else, any opinion about this? Regards, -- Gabriel Bouvigne - France [EMAIL PROTECTED] icq: 12138873 MP3' Tech: www.mp3-tech.org Ciao Robert -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[2]: [MP3 ENCODER] Voice encoding questions
Frank Klemm schrieb am Sam, 05 Aug 2000: snip Another question: Is there any tool to analyze the number of SI, MS and LR frames in a MP3? I don't know if there is any, but you can write a simple tool scanning all mp3 headers and counting the type of each frame. Or you can look out for a tool called mp3check and let it make a dump, all you would have to do is count the different stereo mode extensions. -- 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: [MP3 ENCODER] DC cancelation
:: :: 2nd problem :: ~~~ :: Some of my CDs have a lot of DC offset. Other are well DC free. Examples: :: :: :: What is DC? :: Direct Current. Non wriggling part of the audio signal ;-) n-1 DC := 1/n Sum x_i i=0 -- 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: [MP3 ENCODER] corrupted bitsream
Takehiro Tominaga schrieb am Don, 27 Jul 2000: "R" == Robert Hegemann [EMAIL PROTECTED] writes: Hi, all I fixed this bug and committed to the CVS. pls check out. R Now I get assertion failures in bitstream.c uum the problem is not easy as I thought I restored old takehiro.c and now it works, but the problem is still there. I will try again in the weekend. --- Takehiro TOMINAGA // may the source be with you! Hi Takehiro, any new ideas how to fix the problem? I noticed that all corrupted frames have something in common, the big values are below 4 and that does not look right to me. Ciao Robert -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] --voice Modus
Frank Klemm schrieb am Die, 01 Aug 2000: For = 56 kbit/s Voice-Mode of Lame always sounds better than the normal mode. I've tested several kind of music and also spoken words. So there are 2 questions: * Does voice mode become standard for low bit rates? * What makes voice mode (it's not IS, it also works with mono)? * is IS for lame planned? -- Frank Klemm LAME's voice modes turn off short blocks and apply a lowpass filter around 12 kHz. In its first draft some lower MDCT coefficients were dropped (something like a highpass filtering, we had no filter code at that time), but not nowadays. I can't tell you if someone is working on an intensity stereo mode for LAME, but I don't think so. Maybe for lower constant bitrate modes we have to rework the bit allocation scheme. Ciao Robert -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] Voice encoding questions
Another question: Is there any tool to analyze the number of SI, MS and LR frames in a MP3? Frank, you just need a GTK enabled version of lame :-) run lame -g on the mp3 file, scroll to the end, and then click 'show' under the 'stats' pull down menu. It shows the info you want, and any additional statistics would be easy to add. You can also use to to examine the mid/side bit allocation frame by frame. You could test your ideas about near mono files via the following: Modify reduce_side() function in quantize-pvt.c to be more aggressive. Right now it allocates at most a 33/66 split between side channel and mid channel, based on the side_channel_energy/total_energy ratio. As Robert mentioned, a more aggressive split can create artifacts. I think the problem is that allocating just a few bits to the side channel can produce audible glitches which will sound worse than if 0 bits were used. But no one has done a detailed study of this. -mm Use Mono -mi Use Intensity Stereo, MS-Stereo and LR-Stereo -mj Use MS-Stereo and LR-Stereo -ms Use LR-Stereo -ma Analyze FIle before any converting, select -mm, -mj or -ms I think -ma would be beyond the scope of LAME. A seperate analysis program should be written, and then a GUI front end should run the analysis and make the selection. This is similar to automatic level adjustment. A couple people have expressed interest in adding a volume adjustment to LAME, which is a fine, but the additional step of runing some analysis on the file to determine the adjustment should be left to a seperate program. Mark -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] corrupt mp3 finder
Hi Francois, FdT a program to check the integrity of mp3's. Check out this piece of code: http://www.geocities.com/mp3utility/ Greetings, -E. [EMAIL PROTECTED] -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
