similar to: speex resampler

If more than one speaker receives the *same* signal, it doesn't matter the number of speakers. It only gets tricky when the speakers are playing slightly different signals (e.g. from a stereo song). Jean-Marc Quoting Greger Burman <greger at mobile-robotics.com>: > One thing I can think of is if you are using two or more speakers. If the > speakers are not at the exact same

The reasons I have posted these questions are: 1) To find out if Speex can take advantage of SIMD extensions. 2) To maybe learn from someone with previous experience in optimizing Speex for moderns x86 architectures before I set off trying all kinds of things on my own. See answers inline: 2009/6/15 Tom Grandgent <tom at grandgent.com> > Why haven't you tried using release build

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

I have a question about the overall performance of Speex and what I can do to improve it. I'm running Speex Windows x86, Visual C++ EE compiler. I will say right away that I've only compiled debug so far and used no compiler optimizations at all. I use the uwb-mode, preprocessing, denoising and echo cancellation. I've noticed that speex consumes a lot of cpu resources. When I run this

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

On 2010-03-16 14:22, Josh Gargus wrote: >> If more than one speaker receives the *same* signal, it doesn't >> matter the number of speakers. It only gets tricky when the >> speakers are playing slightly different signals (e.g. from a stereo >> song). > > Does "tricky" mean that the Speex AEC won't handle such situations > well? Or just that you

I believe there is a documentation error regarding speex_preprocess_ctl(preprocess_state, request, ptr); The manual states: SPEEX_PREPROCESS_SET_DENOISE Turns denoising on(1) or off(2) SPEEX_PREPROCESS_SET_AGC Turns automatic gain control (AGC) on(1) or off(2) I noticed that setting a value of 0 turns the setting off. Setting a value of 1 or 2 turns the setting on. While I do think 1 and 0 is more

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: comunip.gif Type: image/gif Size: 1663 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.gif

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

Hello. I have the following situation. AEC is used in network chat software over DirectSound API. Echo and reference signals are almost aligned (delay is no more than 30ms). When echo is emulated in notebook (built-in speakers + mic) everything goes fine and echo is cancelled. But when configuration includes stand-alone speakers and mic no echo is removed. Audio is in 22050 hz at 16 bit

.. And a version without the API changes. -------------- next part -------------- Index: libspeex/resample_sse.h =================================================================== --- libspeex/resample_sse.h (revision 0) +++ libspeex/resample_sse.h (revision 0) @@ -0,0 +1,128 @@ +/* Copyright (C) 2002-2008 Jean-Marc Valin + * Copyright (C) 2008 Thorvald Natvig + */ +/** + @file resample_sse.h +

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 ,

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

>> 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

Hi, Here's the (hopefully) final version of the resampler, now always using st->mem as the buffer area. It only allocates buffers on the stack when it's necesarry to convert the output between int and float. -------------- next part -------------- Index: include/speex/speex_resampler.h =================================================================== ---

Hi Jean-Marc, Thanks for taking a look. I've added an example program in the patches that changes the rate frequently. You can run test-resample2 >test.raw and open in audacity or so to look at the spectrum etc. I've attached a before/after screenshot. In theory, depending on the current phase and the rate changes that are applied, the error can be audible as a pop when changing

Hello? I am a new user of speex.I am currently working on speex frames and I have some questions. I am using narrowband and long tail length, and it works very well with speex test DEMO. But it is very difficult to have speaker input in perfect sync with mic input. Speex does not work at all. Any suggestion? Regards? -------------- next part -------------- An HTML attachment was scrubbed...

Hi, Here's an updated resampler patch against current SVN. It includes SSE and SSE2 optimizations (the latter if included by _USE_SSE2). Best regards, Thorvald -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: speex-resampler-update.diff Url: http://lists.xiph.org/pipermail/speex-dev/attachments/20080426/e055077f/attachment-0001.txt

Quoting Greger Burman <greger at mobile-robotics.com>: > The reasons I have posted these questions are: 1) To find out if Speex can > take advantage of SIMD extensions. If you define _USE_SSE, CELT is already able to use SSE instructions. You must be careful on Windows though because it doesn't support C99 var-arrays and alloca() isn't properly aligned for SSE, so you have to