similar to: Implementation of a Speex based hardware VOCODER

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:

Thanks for the reply. We are currently using AMBE (4Kbps) for our Traffic/Weather Channels. If you have ever had a chance to hear the service, you will know that AMBE does not do us well. I understand that 2Kbps is low quality, but any poorer than AMBE? If can get a decent quality for other low bandwidth talk channels, such as about 10-16Kbps and have it sound rather clean, then I would be

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

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

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

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

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

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

Hi, > You can try the Ogg DirectShow filter to get Windows support for Speex: > http://www.illiminable.com/ogg/ if you wait a little longer, I am trying to provide speex support for the ffmpeg project (www.ffmpeg.org). It does encode/decode to/from avi and wav, a support for ogg has not been done, and my question is, should it be necessary? I don't really like ogg in first place..

So many courses in CodeSchool will be free in this weekend, enjoy! http://www.codeschool.com/free-weekend -- Atenciosamente, Guilherme Pereira Dutra, Fone: (34) 8407-0109 -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To

G'Day, I want to access in a function called from tapply the current factor. In my example below, all I want to do is to write the current factor on each histogram. Needless to say my example does not work. I would be grateful for pointers in the right direction. Many thanks Bernie McConnell Sea Mammal Reserach Unit cc <- 1:10 ff <- rep(c("a","b"),5) pp<-

-----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 */