similar to: Vocoder for SA-1110 proccessor

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

Can PDAs be used as softphones/clients on asterisk? what i wanted to do is to set up 2 PDAs as softphone(client) which allows them to communicate each other through asterisk server(desktop) devices i have: pda compaq model 3680 pda sharp sl5500 access point desktop(asterisk) can i apply my idea on the asterisk? any guides? thanks in advance :) --------------------------------- Do

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

Compared to todays PDAs the GameBoy Advance (GBA) is low spec. it has a 16MHz ARM7tdmi RISC processor. 32-bit but it can handle 16-bit instructions in a mode that resembles "bytecode" (a 32-bit instruction is executed using a 16-bit instruction). I got a feeling that the GBA would need a FPU. For more information on the GBA visit this info page:

Also, you would be sattisfied with the narrowband decoding because the GBA sound quality isn't very good, however, that depends greatly on what you're going to do with speex on a GBA.... (an RPG with audio dialogs? ;-)) also, using a DSP in a GBA Cartridge sounds like overkill to me, however, that depends greatly on what are you developing and what's your target audience....

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

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

-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

I've been working for the past few months with open-source solutions for syncing Palm PDAs (Coldsync, mostly). I've been becoming increasingly annoyed with Palm's Desktop Link Protocol and the sync mechanism, and am considering replacing it with a third-party application for the Palm. Since I've had good luck with rsync in the past, its seemed an obvious choice. I've

Has anybody used anything other than a PC as a source for icecast2? Are PDAs powerfull enough? I know vorbis encoding takes a signifigant amount of CPU power, but I've been able to do it on 4 year old PCs. Does a 206MHz Intel StrongARM 32-bit RISC processor have as much power as a 200 Mhz Pentium? I'm thinking Compaq iPaq, running Familiar Linux and IceS, would make a nice portable

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 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

-----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

Hi. I''m just starting out with Rails. I''m thinking of writing a web based PIM, partly as a learning experience, partly because I need one. I''ll be using iCalendar and vCard files for for storage because it will let me sync with PDAs, and use other already available PIMs. What would be the best practice for this? a) At the beginning of a session pull all my vevents

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

Hi i am Armin Bauer, one of the developers of Multisync, an application to sync various pdas, cellulars and other programs (see www.multisync.org for further information) We recently ran into design limitations with our current version, which was where we decided to develop a new major version of Multisync. I started working on the new design but quickly noticed to there are a lot of other

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've noticed that the nb decoder uses floating point in several places even when using FIXED_POINT. I don't have to deal with lost packets, so I'm mainly interested in innov_gain and pgain in no transmission mode (nb_celp.c around line 1272) and g and exci in vocoder mode (around line 1557). Is there a reason that these places must use floating point, or would it be possible to

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(). >