search for: scalefactors

This is my first post requesting help to this mailing list. I am new to R. My apologies 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) :

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

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

...-------------+---- | L |S|S|S|S|S|S| L | transform +---------------+-+-+-+-+-+-+---------------+---- | N | O |P| Q | floor curves +---------------+---------+-+---------------+---- L = long transform S = short transform A-E = sets of scalefactors (AAC) F-N = floor curves (Vorbis I) M-Q = floor curves (Vorbis II) Obviously Vorbis I is wasting space in this example by coding 5 floor curves (G-K) that are very similar. AAC *shares* the scalefactor set B with these 5 windows thus saving space. Vorbis II could allow the storage of multiple ...

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 scrubb...

I just found this link on slashdot: http://www.soundandvisionmag.com/SoundAndVision/FrameSet/0,1670,_sl_SoundAndVision_sl_Article_sl_0_cm_1653_cm_130_2020_1_cm_00,00.html It is a Sept 1999 article describing some very detailed, double blind listening tests comparing PAC(Lucent), MP3(FhG) and MS audio at 128kbs. In typical microsoft fashon, MS audio is the clear looser despite all their hype.

Dear R-community, I am currently replicating a study and obtain mostly the same results as the author. At one point, however, I calculate marginal effects that seem to be unrealistically small. I would greatly appreciate if you could have a look at my reasoning and the code below and see if I am mistaken at one point or another. My sample contains 24535 observations, the dependent variable

On Sat, Mar 8, 2014 at 7:49 AM, ??????? ?????? <rodionkarimov at yandex.ru>wrote: > 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

##MODISTools example library(MODISTools) library(lubridate) setwd('~/Documents/Modis data') #####MODISTools with buffalo data ###Read in data rename for easier coding tbdata <- read.csv('~/Desktop/All TB data for EVI, NDVI.csv') firstobs <- subset(tbdata, capture.ID == 'B1-1108') firstobs <- firstobs[,c(1,2,2,3,4)] colnames(firstobs) <- c('id',

On 6/22/2017 7:05 PM, Caroline wrote: > ##MODISTools example > library(MODISTools) > library(lubridate) > setwd('~/Documents/Modis data') > > #####MODISTools with buffalo data > > ###Read in data rename for easier coding > tbdata <- read.csv('~/Desktop/All TB data for EVI, NDVI.csv') Since this dataset is only on your desktop it cannot help us

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_ENV...

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