search for: resampled

Hi, I just built a sample application with speex resampler in linux and I tried to resample 8K sine wave tone mono to 48k using speex_resample_process_int. I am using a tool called EAQUAL for audio quality. I find the quality of Speex resampler to be decreasing when I increase the quality q of the resampler init function. Can some one give me pointers regarding this?? As per the API, if the

Hi, I am new to R (I have most of my experience in SAS and SPSS). I was wondering if anyone has used both Resampling Stats and R, and could comment on strengths/relationships. Also, I have no clue on how to do the various examples from the book "Resampling: The New Statistics" in R. Can anyone give me some possible starting points? Or websites/books? Thanks, Brandon

Hi, I've noticed that the opus-tools is using a really old version of Speex's resampler code - a version that I've seen fail in the wild first-hand under low resource circumstances. I've actually submitted patches for some issues in the Speex resampler a while ago (and IIRC they were accepted): http://lists.xiph.org/pipermail/speex-dev/2009-November/007541.html ,

Hi List, Now I will send to you more specific what I am trying to do. I have one Asterisk Channel where receives Midia Frames in the codecs format: Speex UltraWideBand and Speex NarrowBand. When I use Speex NarrowBand the Asterisk is able to convert this frame to G711. But when I use Speex UltraWideBand the Asterisk don't convert it. Then I need in my Asterisk Channel Source include the Speex

From: Jyri Sarha <jsarha at ti.com> This bug happens quite often when resampling from a low to a high sample-rate with big enough factor. Also the resampling call has to be limited by the output buffer size and some unused samples needs be left in the input buffer. Sometimes when up-sampling with a big factor the resampling function wants to peek one more sample from the input buffer to

Alexander Chemeris wrote: > Hi, > > Here are some questions from user point of view. :) > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, >> and there aren't any obvious artifacts. I also did a 16=>16 test, and the >> results are delayed by 10ms

Attached is a snapshot of work-in-progress of a FFT based resampler. At the moment it works in floating point only, and only basic quality inspection has been done. Some benchmarks comparing the filter-based resampler at Q3 with the FFT resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3 -ffast-math, FFTW3, gcc 4.3.0 on x86_64) 16=>48: 59us vs 19us 16=>44.1: 204us vs

Hi I'm having a problem compiling opus-tools-0.1.7. Version opus-tools-0.1.6 seems to compile OK. I've tried with opus-1.0.3 and opus-1.1-beta. The errors are like this:- "undefined reference to `sqrtf'" etc. This OS is Peppermint Three, similar to Ubuntu 12.04. It uses:- gcc --version gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3 Google says it's maybe something to do

I was attempting to use the residual resampling approach to generate 999 bootstrap samples of alpha and beta and find their confidence intervals. However, I keep getting the error message:Error in nls(resample.mp ~ cases/(alpha + (beta * cases)), start = init.values, : singular gradientafter R has only produced a few bootstraps.Could anyone suggest where I am going wrong? Would greatly

Hi, *"That's in general not very reliable. You can get PEAQ to say all sorts of silly things." Can you provide me links for any more effective tools other than PEAQ? Which is more reliable for Speex resampler? * *"strongly suspect that it's just not compensating for the delay introduced by the resampler. Because higher quality means higher delay, you'd find that PEAQ

Hi, Seems like it's not linking with libm. I suspect it has to do with linking statically with libopus.a (is that intended?). Maybe opus-tools relies on the fact that libopus is linked with libm and doesn't explicitly add it? Greg? Jean-Marc On 09/24/2013 06:09 AM, bat guano wrote: > Hi > I'm having a problem compiling opus-tools-0.1.7. > Version opus-tools-0.1.6 seems to

Hi, I understand bootstrap can be used to estimate 95% confidence interval for some statistics, e.g. variance, median, etc. I have someone suggesting that by resampling certain proportion of the total samples (e.g. 80%) without replacement, we can also get the estimate of confidence intervals. Here we have an example of 1000 obsevations, we would like to estimate 95% confidence intervals for odds

Hello, some misc. cleanup patches for speexdsp, nothing big I'm not sure about how to submit patches, so this is a test balloon :) ultimately, I'd like to fix the FIXED_POINT issue, see http://lists.xiph.org/pipermail/speex-dev/2013-December/008465.html currently, I think the only way to find out how speexdsp has been compiled is to resample some bytes and observe the output; which is

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > > > I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, > > > and there aren't any obvious artifacts. I also did a 16=>16 test, and the > > > results are delayed

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

If you can produce a simple minimal repro that exhibits this issue on some procedural signal (sine wave, etc) I'd love to look into it some more - I'm maintaining a hardened version of the Speex Resampler and would be very concerned if this bug made it into production. - Sherief -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block

...ing. We'll call the FFT and iFFT of this our reference. Then, for each resampler below, I've reported the maximum numerical difference in the time domain(comparing ref[i] with sig[i]) as well as SNR. Since my knowledge of SNR for this is a bit sketchy, it's computed as follows: Pad resampled signal with as many zeroes as there are samples. Convert to long double. Use one long double FFT for the entire thing. Then, for both reference and resampled, let power[i] = sqrt(real[i]^2 + imag[i]^2). We only care about the lower half of this power (remember we padded with zeroes). Then, let S...

Hi, I'm trying to write a code (see below) to randomly resample measurements of one variable (say here the variable "counts" in the data frame "dat") with different resampled subsample sizes. The code works fine for a single resampled subsample size (in the code below = 10). I then tried to generalize this by writing a function with a loop, where in each loop the function should do the calculations with increasing subsample size (say for j in 1:100). Problem is, I can...

Hi, I just started to examine the DIV32_16 function (Blackfin ASM version), and wondered why the return value of the function inside 'fixed_bfin.h' is of type 'spx_word16_t', but the local variable 'res' which is returned by this function is of type 'spx_word32_t'. Is this a trick of optimization or a bug? (Same question for PDIV32_16 and MAX16, too!) best