similar to: Re:Speex-dev Digest, Vol 38, Issue 21

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

> 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

Tabitha Flash a ?crit : > Hi, I am looking for AEC software which can be run on the server > side. This means there will be a fairly constant 600ms or so gap > between sending out an audio frame and getting it back with echo. > Could Speex AEC be configured to handle these conditions? Just use a ring buffer to put the delay back to normal. > If so, how > good can I expect it

How much memory reduction are you after? There's lots of stuff you can delete, starting with defining DISABLE_VBR, DISABLE_WIDEBAND and DISABLE_FLOAT_API. From there, you can delete all the *table.c files except for exc_10_32_table.c, gain_table_lbr.c and lsp_tables_nb.c. In libspeexdsp, you can pretty much delete everything (preprocessor, jitter buffer, resampler, ...) but mdf.c Jean-Marc

Message: 2 Date: 29 Dec 2004 15:30:25 +0000 From: Karl Heyes Subject: Re: [Icecast] icecast2.2 and aac? To: qiang Bao Cc: icecast Message-ID: <1104334224.30220.23.camel@bogus.hackers.club> Content-Type: text/plain On Wed, 2004-12-29 at 08:21, qiang Bao wrote: > it seems that icecast 2.2 can only stream aac at 128kbps! > > mp3 and ogg are ok at all bitrates. > > is it a

I am currently trying to transpose some large panel data set ie transposing multiple rows in into a single column. instead the transpose functionality transposes all rows into columns. my sample data set looks like below: ACCT_NUM ACCOUNT_NAME TRAN_AMT DATE EMPLOYER 101913 GK 7489 30-Apr-10 PENSION 101913 GK 7489 30-May-10 PENSION 101913

Hi All, Got today the Orban Opticodec-PC encoder LE and it sounds great at 32Kbps and 44.1Khz and even lower it sounds great!! Now even people with dial-up can listen to FM quality streams! Got a test stream up, if you're intrested, with some rights free (RIAA free) music (dutch language made by a friend of mine). http://dir.xiph.org/index.php?sgenre=&stype=&search=doosfm there

Hi Jean-Marc, I have the following problem when cross-building speex 1.1.7 with MaverickCrunch support. To add this support I had to define CPPFLAGS as follows: export CPPFLAGS='-mcpu=ep9312 -mfix-crunch-d1' but it seems libtool does not like it, while the object files are compiled fine: ... arm-linux-gcc -DHAVE_CONFIG_H -I. -I. -I.. -I../include

This patch updates the symbian build files to latest svn trunk, and also adds a new makefile for speexdsp. Signed-off-by: Alfred E. Heggestad <aeh@db.org> --- Index: symbian/speexdsp.mmp =================================================================== --- symbian/speexdsp.mmp (revision 0) +++ symbian/speexdsp.mmp (revision 0) @@ -0,0 +1,42 @@ +/* + Copyright (C) 2003 Commonwealth

Trying to get the LE version for months now, still treamguys can?t deliver.... Regards, Dennis Heerema -----Original Message----- From: "Greg J. Ogonowski" <greg@orban.com> To: qiang Bao <jakobao@yahoo.com>, icecast@xiph.org Date: Wed, 29 Dec 2004 11:08:04 -0800 Subject: Re: [Icecast] icecast2.2 and aac? Icecast 2.2 works fine with AAC/aacPlus at any bitrate. 32kbps

Dennis- Orban Opticodec-PC LE is available now, as of last Tuesday, the release date of Icecast2 v2.2. StreamGuys is filling all orders now. Thanks. -greg. At 12:53 2004-12-29, Dennis Heerema wrote: >Trying to get the LE version for months now, still treamguys can?t deliver.... > >Regards, > >Dennis Heerema > >-----Original Message----- >From: "Greg J.

ICecast i capable of sorts of streams, that is not your problem. Your stream source client (DSP) is the part who must be capable of streaming the format you want. For AAC use oddcast DSP www.oddsock.org capabale of AAC, LAME Mp3, and OGG (Free to use) very good at OGG specially with the vorbis 1.1 aoTuVb3 DLLs For AAC Plus (HE_AAC) use Orban opticodec for PC (witch is capable of every bitrate

Hi All, I am trying to generate a function from a linear model. I think there should be build-in function that perform this action but I've had no luck finding it. For example, I have a model created using lm(). model <- lm(sat.d~1+sat.n+I(sat.n^2)) What I would like to have is a function (similar to the one generated by splinefun()) so that I can use it on different data-sets.

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's

I want to know speex compression rate. On the speex website,it says" Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." Is the original voice at bitrates raning from 2 to 44 kbps,or the compressed voice at bits ranging from 2 to 44 kbps? what is the maxmium compression rate? Thanks a lot! Weiqin Bao -------------- next part -------------- An

hi Jean-Marc, I just updated from svn and it looks like there are some errors in the speex.mmp file. the filters.c is duplicated, and the fft/kiss.c files are still missing.. /alfred Jean-Marc Valin wrote: > Applied in svn. Thanks. > > Jean-Marc > > Alfred E. Heggestad a ?crit : >> hi, >> >> this patch should fix the symbian build. >> >> >>

Dear sir, I've a problem that the .wav file couldn't play after calling "sampleenc male.wav|sampledec male_speex_15.wav".I found that the new file male_speex_15.wav is smaller than the original file in size.I implemented the test on Linux system.The original file male.wav is 96044 bytes,while the new file male_speex_15.wav is 96000 bytes.I'm eager to know the reason.Thankyou!

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Could anyone tell me how libspeex_armce.lib Speex 1.1.6 was compiled? (http://www.speex.org/download/libspeex_armce.lib ) Does it exist some modified fixed_arm5e.h for Microsoft EVC arm-asm compiler? or the inline assemlby was compiled with GCC and linked through EVC LINK.exe? if the latter was done could I know how to link with EVC ? Objects

Applied in svn. Thanks. Jean-Marc Alfred E. Heggestad a ?crit : > hi, > > this patch should fix the symbian build. > > > /alfred > > > ------------------------------------------------------------------------ > > Index: symbian/bld.inf > =================================================================== > --- symbian/bld.inf (revision 12114) > +++