similar to: fixed / floating point

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Hi everyone, I?m a graduate student in a Brazilian Intitute of Technology, and I?m doing some academic research regarding secure voice transmission over phone lines. One of our reserach goals is to implement a hardware vocoder, with low bit rates, and a preferably free algorithm, to be used in this secure voice system. Actually, there is a functional system using a proprietary AMBE

Hi, My name is Jean Quirion and I am a DSP engineer. Currently I am working on a project where it is desired to implement a VoIP solution over a GSM GPRS link. I would like to use Speex as the vocoder for this application. This application would require the Speex encoder/decoder and possibly the pre-processor to run on a low power fixed-point DSP such as a TI C55x. Thus, I am interested in

Hello, I'm interested in a low rate vocoder to run on SA-1110 or XScale processor (for PDAs voice application). Is there a suitable speex vocoder for that? Thanks in advance, Eyal. ________________________________________________ Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag --- >8 ---- List archives:

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07

Jim, Thank you very much for your suggestions. I managed to get the C55x code working on the simulator. I would like to port Speex both on a C5502 EVM and a C5509A EVM. As such, if you can provide me with the details of your port on the C5509A, it would be greatly appreciated. Furthermore, I am looking for some technical documentation on the CELP algorithms. I would like to better

I was wondering if the developers were using anything to "objectively" test the quality of the speex vocoder. For instance PSQM or one of the many derivatives. Mean Opinion Scoring seems an expensive route. Is there some open source software to use for this? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

--- libcelt/Makefile.am | 6 +- libcelt/bands.c | 26 +++++++++- libcelt/bands.h | 2 +- libcelt/celt.c | 23 +++----- libcelt/pgain_table.h | 133 ------------------------------------------------- libcelt/quant_pitch.c | 117 ------------------------------------------- libcelt/quant_pitch.h | 44 ---------------- 7 files changed, 37 insertions(+), 314

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

I was also wondering if there is a standard set of input sequences people are using to test Speex. I haven't stumbled upon it/them yet. > -----Original Message----- > From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]On > Behalf Of Jean-Marc Valin > Sent: Tuesday, February 25, 2003 7:24 PM > To: speex > Subject: Re: [speex-dev] Quality > > > > I

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Thursday, June 14, 2007 11:17 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain Michael Shatz a ?crit : >>> Actually, you're the first I know using the VisualDSP++ toolchain >>> :-) >> >> I guess

I just got a grandstream SIP phone Here is my sip.conf for the phone [mlh] type=friend insecure=yes username=mlh secret=mlh host=dynamic canreinvite=no The phone as the default config on it. If I use the phone to call a Zap interface (a tdm card) the voice sounds all choppy. If I use the phone to call a x100p card, it does not dial what I dial (no DTMF) I don't know

I'm trying to get ChansSpy to work. It works, in the pass/fail sense, but it is difficult to understand the various speakers. I can hear users on our end just fine, but the other end sounds like their going through a vocoder, if I can understand them at all. Otherwise it is just garbled. We are using the following setup: all of our phones are SIP phones; for our outgoing calls we make use of a

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Tuesday, June 19, 2007 6:38 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain >> Yes, data footprint in the new version is quite manageable. Still I would >> wish better documentation for speex_alloc_scratch(). >

Has anyone else created Vorbis files with more than 6 channels? I'm a game developer trying to use multichannel Ogg Vorbis files for a PS3 and XBOX 360 title. Using the latest release libs and tools, I've created a Mac application that weaves up to 8 stereo AIFF files into one 16 channel Ogg file. What I'm finding is that if anything more than 3 stereo files (corresponding

Hello, I'm porting CELT 0.7.1 to an embedded platform and unfortunately (at least for me) CHAR is defined as 16 bits. I now got the vocoder compiling but when i compare the encoded output with a windows build, they don't match. Among the other problems i think that the char definition is one of the biggest players. I've seen in arch.h the following definitions: /* 2 on TI C5x DSP */

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

I have been thinking about what is needed to make language teaching/learning tools. (Like talking flash cards.) The main thing needed is a low bit-rate encoding of human voice. At first I thought I could take one of the gevernment standard vocoders and embed it as an Ogg stream. But: (1) there is not a standard vocoder, there is are half a dozen, at least. (2) they are fixed bit rate, we

Is there any place where I can see a summary of what is being done and what is still pending with the fixed point version of the libraries? I have some experience with vocoders and fairly good experience with de-jitters and suchs. I may be able to help. Thanks Baja