similar to: Re: Speex-dev Digest, Vol 35, Issue 24

I have been using the Speex NB mode with an 11025 kHz signal, without adjusting the framesize, and it's been sounding just fine to me. However I was wondering what the best way to make 11025kHz would be. I also have an audio driver that uses 2kB buffers, and the NB framesize of 160 doesn't divide evenly into that so I end up having to shovel tails of buffers around which isn't too

Hello, I'm new on DSP and i want to implant speex echo canceller on the Blackfin DSK version of speex to work on the BF533 EZ-KIT LITE But i can perform the echo canceller to work This is my code, so if somebody can tell me where i'm wrong //----------------------------------------------------------------------------- // CallBackFunction() - Gets called by the audio pseudo-driver

Hello, i want to use speex on a Blackfin DSP to make CELP compression I want to know how many MIPS are necessary to make Speex to work? Thanks Damien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070430/e445920e/attachment.html

Hello, I'm using speex provided with the Blackfin SDK on a BF533 EZ-KIT LITE In this version of speex, the program is used in wide band with a sampling rate of 16KHZ But i want to use it in narrow band mode with a sampling rate of 8KHZ In AudDriverFuncs_AD1836.c file, i see that the supported sample rate are 48000, 24000 and 16000 --> I don't understand because in the readme file it

hello, i try to add a speex compression on the TDM Talkthrough on the BF-533 EZ-KIT LITE but when i compile i obtain the following errors: [Error li1021] The following symbols referenced in processor 'p0' could not be resolved: 'speex_bits_destroy [_speex_bits_destroy]' referenced from 'Debug\Process_data.doj' 'speex_bits_init [_speex_bits_init]' referenced from

Hi Opus list, I'm noticing a discontinuity in the quality between use of 10ms and 20ms framesize for mode=AUDIO at a bitrate of about 28000. Quality drops audibly for voice signals when encoded at 20ms framesize, versus quality at 10ms. This effect is mode=AUDIO only. Using mode==VOIP shows no sig. difference between 10 and 20ms framing at this bitrate. Pesq totally

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

Wine version 20010629 running Lotus Notes 5.08 Linux RedHat 7.1 with Xfree86 Version 4 and FVWM2 window manager When I run this version of Wine with FVWM2 the Wine window always appears on top ( always obscures other windows). I use a click to focus window policy and when I click on another window on the desktop it will not raise it over the Wine window although the window I click on does receive

Hi, when I compile with FIXED_DEBUG enabled, I get an error -- both when make tries to build testenc.o and when I try to link my app against the speex lib: lorenz@panelmaker:~/Blackfin/tests/speex_loopthrough$ make bfin-uclinux-gcc -c -g -I/home/lorenz/include -o main.o main.c bfin-uclinux-gcc -gl -elf2flt -L/home/lorenz/lib -o speex_through main.o -lspeex_debug -lspeexdsp -lm

Hi Jean-Marc, after some problems with getting svn to work here I finally made it. Problem is, you write that I cannot use libspeex and libspeexdsp at the same time now -- because I use a "live" system (mic-in -> speex_enc -> speex_dec -> headphone out) and I can run the AD1836 audio codec on 48 kHz only, I cannot use my program now (because I use speex resampling...) So I

Jean-Marc, Frank, I have stumbled across a similar situation regarding optimization. I seem to have a similar setup as Frank does with a fixed 48khz in and out. The wideband mode and ultra-wideband modes are really what I?m looking for. I have a test application that reads audio, downsample to 16kHz (or 32kHz), speex encode, speex decode, upsample back to 48kHz, and playback. If I remove

Hello all! I'm having trouble with a cross compiled libopus for an embedded application and would like some help. preamble: embedded details: processor: bf537 kernel: Linux release 3.0.8-ADI-2011R1-svn4693, build #2834 Fri Apr 22 12:35:34 ADT 2016 toolchain: bfin-linux-uclibc-gcc release gcc version 4.3.5 (ADI-2011R1-RC4) audio interface: ALSA

Hi Jean-Marc, didn't get a reply to my last post (see below) -- do you have no idea what happens here? After some more tests, I disabled the DIV32_16 Blackfin optimizations and now get good quality on the Blackfin. But when I have overdrive on the input, things become very bad -- I'm not sure if this is really a filter stability issue like I wrote some weeks ago. I use the speex

Hi Jean-Marc, I did some further checks with 1.2 beta 3: The problematic function is DIV32_16 inside fixed_bfin.h. When I comment it out (i.e. replace it by the generic version of the routine), the quality improves significantly. Nevertheless, there is still a horrible distortion when there is some overdrive/saturation in the input signal. For me, it looks like there's an instable filter

Hello, I encode a file with speex so i get a .spx file but i want to decode the same file i get only one second of my file and nothing Some idea why? Thanks Damien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070430/d5a95090/attachment.html

Hi Marc, I encountered the problems even with older versions of speex, including 1.2beta2 and 1.1.12, too. I had no time in the last weeks to investigate this issue further, sorry. Hopefully, I'm able to do some more test the next days, but I have no real idea how to proceed Performance and quality seem to be o.k. for me if I deactivate the Blackfin optimizations inside the file

nautiluz wrote: >I am starting project to develop open implementation of some (naturally open >codec) for simply designed embedded devices which can be used by small to >big VoIP operators or hobbyists which wants to build their own small and >low cost VoIP phone. >Speex seems to be great choice. And greater will be if there should be >some guys who want to help ;-). Sorry

-----Original Message----- From: Robin Getz [mailto:rgetz@blackfin.uclinux.org] Sent: Saturday, June 16, 2007 12:11 AM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain >On Wed 13 Jun 2007 12:37, Michael Shatz pondered: >> >> Hi Jean-Marc >> >> I'm trying to integrate your speex codec on our

Frank Lorenz a ?crit : > And yes, the same "overflow" happens even when I disable Blackfin ASM > optimizations. Indeed, that shouldn't happen. Just to make sure I understand, so far there's two problems: 1) DIV32_16() in Blackfin assembly causes problems 2) The resampler overflows When you fix/workaround those two, is the encoder/decoder working correctly or are there

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