similar to: Vorbis for non audio stream

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

Hi, But in fact, it really affects the voice quality. One of my tester says, "Is your mouth far way from the mic?" Could you explain why we should cut 200hz below? >The notch filter is specifically designed to cut below 200 Hz when >working in narrowband. In wideband, the cutoff is more around 50 Hz. The >reason is that in narrowband operation (irrespective of the

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 >

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,

Hi, I chose speex initially because i had some work in VQ on speex i.e. modifying split VQ to GMM based parametric VQ and I thought If I train the GMM based VQ codebooks with audio signal and then do audio coding with speex, I probably get a better(smaller) residual signal even with speex. But I couldnt get that. I was trying to get a lossless bitstream by MUXing the speex-bitstream and the

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

On Dec 2, 2010, at 14:53, scott brown wrote: > original 24/48 wav file: 264,904,968 bytes > flac level 8: 105,992,780 bytes > > dithered 16/48 wav file:173,885,996 bytes > flac level 8: 108,700,948 bytes > > truncated 16/48 wav file: 173,885,996 bytes > flac level 8: 105,224,448 bytes > > RMS level of original 24 bit: -15.3dB with peaks at -0.3dB > > if I

Thank you. But it will cost you a long time to get the accurate play and capture frequencies. Does your program test two frequencies of the sound card each time? Because different sound cards have different frequency errors. And the resampling program is also time consuming because the target frequency is so close to the sampling frequency of the input signal, isn't it? I have tested program

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

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 All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the

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 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

On Aug 8, 2009, at 23:11, Didier Dambrin wrote: > Electronic music quite often doesn't leave a computer these days. > And it > mainly consists of drums, synths & vocals/effects. Drums are often > samples > sequenced at sample (not sub-sample) accuracy, thus repeated (of > course if > the song was post-resampled, there will be sub-sample times). Good point. I

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of