search for: waveform

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 functi...

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, within a single lost 20ms packet I hear two replicas of the last 10ms of the previous waveform. Sometimes the repeats are only 4ms long thus repeating 5 times. This m...

...ck 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 headphones. So how can i use Opus in this way? Thanks in advance WAVEFORMATEX waveform; waveform.cbSize = 0; waveform.wFormatTag = WAVE_FORMAT_PCM; waveform.nChannels = 2; waveform.wBitsPerSample = 8; waveform.nSamplesPerSec = 48000; waveform.nBlockAlign = waveform.nChannels * (waveform.wBitsPerSample >> 3); waveform.nAvgBytesPerSec = waveform.nSamplesPerSec * wave...

...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 fall 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...

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

...====================== downloaded 303 Kb * Installing *source* package 'mlbench' ... ** libs gcc -m64 -std=gnu99 -I/usr/include/R -I/usr/local/include -fpic -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -c waveform.c -o waveform.o waveform.c:17:26: error: R_ext/Random.h: No such file or directory waveform.c: In function ?waveform?: waveform.c:35: warning: implicit declaration of function ?GetRNGstate? waveform.c:38: warning: implicit declaration of function ?PutRNGstate? waveform.c: In function ?execute?...

...op 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 ? waveform analysis to get continous MOS scores for your network To: asterisk-biz at lists.digium.com [image: AQuA Meter]<http://blog.sevana.fi/wp-content/uploads/2013/03/screenshot.png> Hi, We would like to offer you to learn about our new application that performs scheduled voice test calls to a p...

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...

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 hel...

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 th...

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 disabled perceptual enhancement). Do you...

...ser 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 sample and suppose it generates a waveform with non audible frequencies. Then 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...

...'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...

...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, within a single lost 20ms packet I hear two replicas of the > last 10ms of the previous waveform. Sometimes the repeats are only 4ms > long thus repeat...

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 noti...

...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 starts, the remote end finishes sooner. I can find points in the remote recording where parts of the waveform have been missed out, leading to jumps in the waveform, which correspond to the audible clicks. These "jumps" seem like dropped packets, and I'm deducing that Asterisk is sending data slightly too fast (i.e. more frequently than 50x160 sample per second) for the remote end, which has...

...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 = al, envir = data_env) : data set 'waveform.train' not found However, I have no problem with this when using the package. This datasets are loaded, multiple datasets at a time, under another name. data(vowel) loads the two first in t...

...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 = al, envir = data_env) : data set 'waveform.train' not found However, I have no problem with this when using the package. This datasets are loaded, multiple datasets at a time, under another name. data(vowel) loads the two first in t...

...use 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 of the format chunk, one that's old and belongs to the original .WAV specification (the corresponding structure is called WAVEFORMAT in the win32 sdk and has 14 bytes, the variant you'll typically see in (PCM) wave files is called PCMWAVEFORMAT and has 16 bytes - the size that you expect the format chunk to be). But there's also a newer one, called WAVEFORMATEX, which was created to suit the requirements of non-PCM cod...