search for: scalefactor

...ogies for any breach in posting etiquette. I am new to this language and just learning my way around. I am attempting to run some sample code and and am confused by the error message: Loading required package: rrcov Error in fitNorm2(fdat[, "FSC-H"], fdat[, "SSC-H"], scalefac = ScaleFactor) : Required package rrcov could not be found. In addition: Warning message: there is no package called 'rrcov' in: library(package, lib.loc = lib.loc, character.only = TRUE, logical = TRUE, > that I get when I attempt to run the following sample snippet of code. The error above...

Hi, I think that it would be a good thing to know more about those 2 projects (and also the future patent free format). I think that many people as me know about Lame, but not about Vorbis, and vice-versa. It would be fine that someone (perhaps the maintainer) of every project would introduce to both group of people those projects. 2 things would be interesting (to my mind): - to know about the

Dear list, My query follows on from a question I posted a few days ago. I have the following 2 sets of data: wetMeans[1] 9.904762[2] 6.344828[3] 6.346154[4] 6.855769[5] 9.074324[6] 9.953988[7] 13.482966[8] 14.546053[9] 10.841584[10] 9.752033[11] 6.739336[12] 8.955056burnMeans[1] 0.06214286[2] 0.05396552[3] 0.04096154[4] 0.05302885[5] 0.05831081[6] 0.07392638[7] 0.29969940[8] 0.25596217[9]

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying the ogg123 source...

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying the ogg123 source...

...------------+---------------+ | 1 | 2 | 3 | frame no. +---------------+-+-+-+-+-+-+-+-+---------------+ | L |S|S|S|S|S|S|S|S| L | transform +---------------+-+-+-+-+-+-+-+-+---------------+ | A | B |C| D | E | scalefactor sets +---------------+---------+-+---+---------------+ Vorbis I +---------------+-+-+-+-+-+-+---------------+---- | 1 |2|3|4|5|6|7| 8 | packet no. +---------------+-+-+-+-+-+-+---------------+---- | L |S|S|S|S|S|S| L | transform +------------...

Hello. I create FLAC file decoding, processing and playing program and have the following question : how to convert FLAC 16 bit file data to 32 bit float buffer for CPU processing? I've already inplemented sound playing and tested it with sine wave - it works without problems; I even made writing into decoding buffer values of sine wave, instead of decoded FLAC file data, and it also works

Hello, I'd like to know there is any place (web, document...) where I can see which the vorbis' psychoacoustic model is, at least for the reference encoder (how it works, algorithms, filter banks, scalefactors, etc., or if it doesn't use any of this things at all...), or if the only way of looking at it is within the psy.c. I've been looking for it in the Vorbis' site and in the web but haven't found anything... Thanks -------------- next part -------------- An HTML attachment was scrub...

...cent), MP3(FhG) and MS audio at 128kbs. In typical microsoft fashon, MS audio is the clear looser despite all their hype. There are types of music where MP3 does better than PAC and vice versa, but overall MP3 comes out ahead. In only one sample did MP3 perform poorly. MP3 and PAC are both MDCT/scalefactor type codecs. Anyone know what the technology is in the microsoft codec? Mark --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

...uped" 9 times because this variable consists of 9 levels > ttt <- as.data.frame(ttt) > xbar <- as.matrix(mean(cbind(1,ttt[1:19]))) #1:19 position of variables in dataframe ttt > betaprobit4AKIE <- probit4AKIE$coefficients > zxbar <- t(xbar) %*% betaprobit4AKIE > scalefactor <- dnorm(zxbar) > marginprobit4AKIE <- scalefactor * betaprobit4AKIE[2:20] #2:20 are the positions of variables in the output of the probit model 'probit4AKIE' (variables need to be in the same ordering as in data.frame ttt), the constant in the model occupies the first position...

...d data to 32 bits float CPU buffer, but they give > noise or sound with artefacts. I use Stream Decoder from FLAC C API and my > write_callback is as follows : > I've used code like the following to convert decoded FLAC to normalized floating point audio in the interval [-1,1): float scaleFactor = (1L << ((((frame->header.bits_per_sample + 7) / 8) * 8) - 1)); for(unsigned channel = 0; channel < frame->header.channels; ++channel) { float *floatBuffer = /* target output channel buffer from somewhere */ for(unsigned sample = 0; sample < frame->header.blocksize; ++sample)...

...SSubsets MODISSubsets(LoadDat = period, Products = product, Bands=bands, Size=pixel, SaveDir='.', StartDate=T) ###MODISSummaries MODISSummaries(LoadDat = period, FileSep=',',Product='MOD13Q1', Bands = '250m_16_days_EVI', ValidRange=c(-2000,10000), NoDataFill=-3000, ScaleFactor = 0.0001, StartDate = TRUE, Interpolate = T, QualityScreen = TRUE, QualityThreshold = 0, QualityBand = '250m_16_days_VI_Quality') > On Jun 22, 2017, at 4:50 PM, Bert Gunter <bgunter.4567 at gmail.com> wrote: > > 1. You should always cc the list unless there is a clear reas...

...ets(LoadDat = period, Products = product, Bands=bands, Size=pixel, SaveDir='.', StartDate=T) > > > ###MODISSummaries > MODISSummaries(LoadDat = period, FileSep=',',Product='MOD13Q1', Bands = '250m_16_days_EVI', ValidRange=c(-2000,10000), NoDataFill=-3000, ScaleFactor = 0.0001, StartDate = TRUE, Interpolate = T, QualityScreen = TRUE, QualityThreshold = 0, QualityBand = '250m_16_days_VI_Quality') > > >> On Jun 22, 2017, at 4:50 PM, Bert Gunter <bgunter.4567 at gmail.com> wrote: >> >> 1. You should always cc the list unless th...

I intend to finish my Vorbis document on Monday so I will leave it over the weekend to see if there are any more modifications suggested. Although the document will be internal I will put a copy of the final version at the same address - I know firsthand how annoying it is to find loads of dead links when trying to find background info on Vorbis (I know google has probably already got a link

1. You should always cc the list unless there is a clear reason not to. 2. You still have failed to follow the posting guide: You say you have difficulty troubleshooting your code, but you have shown us no code. You got an error message that seems explicit, but with neither code nor data, I do not know whether anyone can make sense of it. In any case, I certainly cannot. Cheers, Bert Bert

...he code generated "with instruction combining" is: %coupling = getelementptr inbounds i16* %timeData, i32 1060, !dbg !575   %24 = bitcast i16* %coupling to i32*, !dbg !575   store i32 0, i32* %24, align 4, !dbg !575     The FRAME_DATA, is defined as: typedef struct _FRAME_DATA {   Word16 nScaleFactors;                    /*!< total number of scalefactors in frame */   FRAME_INFO frameInfo;                 /*!< time grid for current frame */   Word16 domain_vec[MAX_ENVELOPES];  /*!< Bitfield containing direction of delta-coding for each envelope */   Word16 domain_vec_noise[MAX_NOISE_EN...

Hello all. In the shower the other day (where most of this sort of musing gets done, eh?) I was thinking about graceful degradation of audio signals. Let me apologise in advance if these are elementary concepts or if I demonstrate a complete lack of insight -- I don't rate even a dabbler status in the area of audio codecs. Anyway: If we have a 128kbs signal coming down a *udp* channel with

...ode in instruction combining. Below is table which shows address offset calculation for the struct elements (described earlier).   Type Variable Actual Size(in bytes) pass fail llvm2.9 Address(pass) Size-pass (in bytes) Address (fail) Size -fail (in bytes) with llvm2.9 Size (with llvm2.9) short  nScaleFactors;  2 40ac0198 4 40ac0198 2 40ac0190 2 FRAME_INFO  frameInfo; //struct described below 34 40ac019c 36 40ac019a 36 40ac0192 36 short  domain_vec[MAX_ENVELOPES];  10 40ac01c0 10 40ac01be 10 40ac01b6 10 short  domain_vec_noise[MAX_NOISE_ENVELOPES];  4 40ac01ca 6 40ac01c8 4 40ac01c0 4 INVF_MODE  sbr...

With reference to the previous query, I think, i miscalculated the offset, just recalculating. 1. without instruction combining coupling member variable, is at:   %struct._FRAME_DATA* %2, i32 0, i32 5   where "%2" is defined as:   %arrayidx3 = getelementptr inbounds i16* %Data, i32 1024, !dbg !446   %2 = bitcast i16* %arrayidx3 to %struct._FRAME_DATA*, !dbg !446 i.e. at 5 offset in

Apologies for resurrecting an old thread and likely poor formatting - I wasn't subscribed to the list at the time and have copied it out of the archives (http://lists.alioth.debian.org/pipermail/nut-upsuser/2010-July/006104.html). I've been unable to locate any further progress on this UPS (Liebert 10af:0001) - a thread from 2009