similar to: Scale of 2nd y-axis

Hello all, Hoping that there is a fairly simple solution to my query... I'm trying to overlay a line plot of some data onto a barplot of different data. The y-axes are different for each set of data but the x-axes are the same (1:12, corresponding with 12 months of observations). The problem I'm having is that the centre of the bars and the points on line don't match up, which is

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

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

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

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.

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) :

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

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

##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

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

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

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

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

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

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