similar to: Playback 5% Too Fast?

Just checked my figures, and I mean 0.5%-0.7%. Anyway, it is the resulting clicks that are the problem. Any help still appreciated. David -----Original Message----- From: David Brazier Sent: 13 March 2007 00:33 To: asterisk-users@lists.digium.com Subject: Playback 5% Too Fast? Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI

I'm collaborating with and electronic musician to experiment on the production of music from number sequences. As I'm an R user I started playing around with the sound library and I found it very useful. However there are several things I do not understand (I'm not an expert in acustic nor audio signal treatment). The first thing I'd like to understand is: let s be a normalized

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single

I am specifically using Celt Wideband (48kHz) over WiFi multicast that naturally leads to lost packets and am trying to minimize the impact to the audio. I implemented PLC but the audio it produces is robotic. Have I implemented PLC correctly? Checking the waveform it is using the previous received waveform to fill in a missing packet but not the full waveform so it has to repeat. Basically,

I have three time series, x, y, and z, and I want to analyse the relations between them. However, they have vastly different resolutions. I am writing to ask for advice on how to handle this situation in R. x is a stimulus, and y and z are responses. x is a rectangular pulse 4 sec long. Its onset and offset are known with sub-millisecond precision. The onset varies irregularly -- it doesn't

Hello, I am trying to install R packages under linux, some of the packages can not be installed and I got the following error, could anybody give me suggestion on where the problem is and how to fix it? Thanks-e > .libPaths() [1] "/usr/lib64/R/library" [2] "/usr/share/R/library" [3]

I am puzzled by a filtering problem using fft(). I don't blame R. I have a waveform y consisting of the sum of 2 sinewaves having freqs f1 and f2. I do s = fft() of y. Remove s's spike at freq=f2 Do inverse fft on s. The resulting waveform still has a lot of f2 in it! But the filtering should have removed it all. What is going on, and how to fix?? Thanks very much for any help. Bill

Hi all! I would like to use Ogg-Vorbis to encode a non audio waveform. My waveform is in .wav format, on 16 bit mono, with frequency range from 100Hz to 100MHz. It's about 100MB lenght. I need to compact it with lossy for net transfer. Is there something like this, already done, that can help me ?? How can I measure the distortion that Vorbis introduce? I'm sorry for my bad english.

I am forwarding your request to the FLAC development mailing list. ----- Forwarded message from Jamie Heilman <jamie@audible.transient.net> ----- Date: Sun, 8 Sep 2002 16:13:32 -0700 From: Jamie Heilman <jamie@audible.transient.net> Resent-From: Jamie Heilman <jamie@audible.transient.net> To: submit@bugs.debian.org Subject: Bug#160155: gapless playback Package: xmms-flac

Hi, I have been using speex version 1.0.5 on a text-to-speech program. Recently I upgraded to version 1.2beta1 and noticed that the waveform the I got after encoding and decoding on the new versions (beta1,beta2) is much more different than the original than on version 1.0.5. I also ran a PESQ comparison test on 700 voice samples and got better results in the older version (I used quality 9, and

This is rw2010 from CRAN. When running Rcmd check on a package I get: Warning in utils::data(list = al, envir = data_env) : data set 'vowel.test' not found Warning in utils::data(list = al, envir = data_env) : data set 'vowel.train' not found Warning in utils::data(list = al, envir = data_env) : data set 'waveform.test' not found Warning in utils::data(list

This is rw2010 from CRAN. When running Rcmd check on a package I get: Warning in utils::data(list = al, envir = data_env) : data set 'vowel.test' not found Warning in utils::data(list = al, envir = data_env) : data set 'vowel.train' not found Warning in utils::data(list = al, envir = data_env) : data set 'waveform.test' not found Warning in utils::data(list

The following is my R script which I am struggling with to assess ICESat data..perhaps it is the ID_min or ID_max that is wrong? I don't know, any help would be greatly appreciated :( # OPTIONS - CHANGE THESE VARIABLES IF NEEDED\par ######################################################################\par \par icesatfile <-

I'm not an engineer so I hope I'm using the correct terminology here. I have a recorded waveform that I want to apply low and high pass filters too, are tehre already R functions existing to do this or am I going to have to program my own? thanks for any pointers tom

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

Hi, I have one doubt again, that is Vorbis use DCT/MDCT based algorithm and also use psychoacoustic model so this is lossy codec. And I dont think it ca regenerate a better matching waveform than speex. Then there comes FLAC which is the perfect answer to my question, I suppose. But my concern is this that FLAC use simple prediction algorithm and doesnt use any CELP based algo which could have

On Aug 7, 2009, at 21:48, Didier Dambrin wrote: > FLAC doesn't preserve every chunk? I thought it did. I only gave a > quick try > but it seemed to have preserved even the most obscure chunks. > Let me check: it even seems to preserve "MIDI note associated to > marker", > which is a very unknown metadata used by SoundForge (& even defined > in a >

Hello, I'm a student now using libvorbis in my project. I aim to record voice through APIs in winmm.lib in wav buffer, and than encode the raw data in wav buffer and save as ogg file. The wav buffer is offered in a pipeline scheme, and in order to save memory, when a wav buffer is full, I start a new thread to process encoding through libvorbis. Problem is, when wav buffer is full and

Hi, I need to plot spectrogram of a waveform.What package offers this? -- Rajesh.J [[alternative HTML version deleted]]

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are