search for: waveforms

Hi! I'm totally new to audio programming. I managed to create a VoIP with waveform (PCM). The problem was the really high traffic, so I desided to use speex for compressing the data. I'm recording with mono, 16bit/sample, 8000 samples per second. I tried to add the speex compression, but I always get a crash when I try to decode the data. Here is my encode and decode function: int

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,

Hello, I have an p2p voice chat application using WAVE PCM (winmm). Now i am trying to add opus encoding to it to send it over the TCP/IP and then decoding it back to play - but without success (without opus it works ok) Here is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my

...all on neat 1/2 sec or sec boundaries for example. y is a sampled continuous waveform. It is highly noisy, and it is actually well represented by samples 1/2 sec apart or so. z is a very short pulse -- perhaps 5 ms long -- occurring at irregular times. I have problems with how to represent these waveforms at the input stage (during data collection) and at the analysis stage. If I want to represent x and z with the sort of precision I'd like, I'd have to sample every ms or so. But: - that is massive overkill for y, because it is noisy. I will have maybe 500 times as many pts as I require! Tha...

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

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]

Hi, Although this is a repost from Asterisk biz, we would like to ask if somebody may help us to develop a native Asterisk module using AQuA technology for voice quality monitoring using the same web service AQuA Meter is using. Thanks, Sevana Finland/Estonia ---------- Forwarded message ---------- From: Sevana Oy <sales at sevana.fi> Date: Mon, Jun 17, 2013 at 7:30 PM Subject: AQuA Meter

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 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

I have 400 cycles of a noisy waveform. I would like to find the average (over the 400 cycles) of one cycle. Please tell me to do it in R. Any suggestions welcome! Thanks very much for any help. Bill

Im am curerntly trying to use R software to extract uncompressed waveform values with I have acquired from IDLreadGLAS tool. I am very lost with this & know I have to assess the GLA01 product data, however, any advice on the matter atall would be brilliant, particularly on define thresholds and connecting GLA01 and GLA14 data (or any data for that matter) -- View this message in context:

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

...n if I play it trough the function "play" (I'm using an R-window version) I won't hear anything. Then I play: play(10000*s) and I can hear something. What is happening to the audio sample? I saved as wave files both s and 10000*s and then with my musician friend we looked at the waveforms we got. They are totally different from each other and we couldn't fidure out which transformation of s generated the 10000*s waveform. Can anyone explain what is going on? Thanks in advance Giovanna Jona Lasinio

...'C:/Users/Whiz/My_Documents/Year3/GY3072/ICESat_GLAS/IDLreadGLASWF/out_wf_GLA01_09030918_r5947_031_L2E.dat'\par \par plotdir <- 'C:/Users/Whiz/My_Documents/Year3/GY3072/ICESat_GLAS/PLOT'\par \par \par \par ID_min <- 976834501 290018099.3145 1000 #231741042 # specify range of waveforms to be plotted, -1 means all\par ID_max <- 976834506 290018101.2895 1000 ############@@@@@@@@@@@note only one ID used @@@@@@@@@@@@@@@\par \par \par \par noise <- 0.05 # values below NOISE will be disregarded in the waveform fitting\par sdthres <- 0.02 # threshold for a standard deviation o...

What do you mean by "implement"? You're just using the Opus built-in PLC (passing NULL), right? The PLC generally attempts to find periodicity and replicate it. I guess if your signal isn't periodic it can lead to a repetition that isn't great. It's something that could probably be improved in the PLC. Cheers, Jean-Marc On 08/25/2015 01:21 PM, Scott Boekweg wrote:

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 notic...

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

A friend of mine recently had a problem with a "broken" .WAV file (as you call them) because oggenc first printed out a warning and then didn't accept the file because of a "unexpected EOF error". Because I was interested in the issue, I decided to take a look at the oggenc source, and in fact, it is your .WAV reader that's wrong. More precisely: there are two version