similar to: SPEEX and OPUS questions and minor issues

Hi, On Fri, Jul 29, 2016 at 1:50 PM, dos386 <dos386 at gmail.com> wrote: > I recently stumbled upon <speex/speex_resampler.h> and made a chain of > discoveries: > > - "http://speex.org/downloads" some links are broken ("rarewares" and git) What do you mean by broken? Those links seem to work here. > - there is some (minor) development (whitespace

Thanks ... > Those links seem to work here. 2 links "http://rarewares.hydrogenaudio.org" redirect to "https://hydrogenaud.io" ... no "rarewares" subdomain "http://git.xiph.org/speex.git" maybe better "http://git.xiph.org/?p=speex.git;a=summary" "http://git.xiph.org/speexdsp.git" maybe better

This is on my TODOOn Wed, Aug 3, 2016 at 1:19 PM, dos386 <dos386 at gmail.com> wrote: > Thanks ... > >> Those links seem to work here. > > 2 links "http://rarewares.hydrogenaudio.org" redirect > to "https://hydrogenaud.io" ... no "rarewares" subdomain > > "http://git.xiph.org/speex.git" maybe better >

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

Hi all, I build successfully with speex-1.2rc2. And with speexdsp 1.2rc3, I build with i386, X86_64, armv7 and armv7s all passed. But when I build for ARM64 (for iPhone 6), it failed with: /Applications/Xcode.app/Contents/Developer/usr/bin/make all-recursive Making all in libspeexdsp CC preprocess.lo CC jitter.lo CC mdf.lo CC fftwrap.lo CC

I've filed a bug for aarch64 https://github.com/xiph/speexdsp/issues/7 and provided the port in a fork with a pull request. We need someone to review/merge in the pull request? It provides the source code, but my testing was under Android builds, so there would be some configure changes needed to build it stand alone. On Tue, Apr 19, 2016 at 4:32 PM, Frank Barchard <fbarchard at

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

Dear Nikos, thanks! But you use Opus only for resampling, not for entirely replacing Speex, don't you? Greetings! Hermie Am 07.05.2013 22:53, schrieb Nikos Chantziaras: > The Opus resampler is actually a bugfixed version of the Speex one. Same > interface/API, but with the bugs removed. It's why I recommended it :-) > Otherwise I would have recommended something entirely

Hi, On Sat, Mar 28, 2015 at 2:34 PM, Evan JIANG <firstfan at gmail.com> wrote: > Hi all, > > (Sorry that may be duplicated that I was not a mail-list member before, > so last mail sent failed) > > I build successfully with speex-1.2rc2. And with speexdsp 1.2rc3, I > build with i386, X86_64, armv7 and armv7s all passed. > But when I build for ARM64 (for

Hi, These patches improve the resampler set_rate function. The first 2 patches do some cleanups in the GCD calculation and avoid an overflow when calculating the new phase. This patch could probably be simplified if we allowed 64bits operations in speexdsp. The 3rd patch avoid rounding errors in the phase calculation. The problem is that the new rate is calculated with the reduced rate, which

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter for audio? thanks, nir -------------- next part

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 David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

>> 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

Can anyone recommend a good library for performing audio downsampling? I intend to start playing around with "libresample" (http://ccrma-www.stanford.edu/~jos/resample/README-libresample-0.1.3.txt), as well as taking a look at "Secret Rabbit Code" (http://www.mega-nerd.com/SRC/), but I'd love some opinions before I get too involved with either. Free would be best, but

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

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 everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

Hi I am using speex for cancelling echo.but i am not succeeded. I need echo cancellation in urgent.Please help m. Thanks in advance Devi