similar to: Ogg Voxpop

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

Yes, the arch is just as you said, something like AMD GPU, but Intel GPU don't have separate register file for 'scalar/vector'. In fact my idea of defining the register tuples was borrowed from SIRegisterInfo.td in AMD GPU. But seems that AMD GPU mainly support i32/i64 register type, while Intel GPU also support byte/short register type. So I have to start defining the registers from

Hi. I have search for the answer to this question, to no avail. Does anybody know the authoritative pronounciation of samba. Maybe there isn't an authoritative pronounciation. If anybody can help me I would greatly appreciate it. Thanks, -Mark The information contained in this e-mail message is privileged and/or confidential and is intended only for the use of the individual or entity

Ok, this is slightly offtopic, but relates to the quality of input for speex :) I'm working on echo cancellation by means of sampling the wave mix of the sound card as well as the microphone. I originally had two sound cards, which had some synchronization problems (now solved, more or less), but I have also discovered a much better solution using ASIO 2.0, which enables me to sample

Hi Escha, Great to have your comment! Do you have any specific reason for not doing like this? I am not sure whether I understand your point correctly. For "just model one thread", do you mean "only considering ONE of the 8/16 working lanes that running in lock-step way"?? For my case, may be something like I only need to define r0~r127 as register for i32 register (each r#

On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote: > AM radio is lower quality (mono) but I don't know > what the digital equivalent would be. Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad)

Hello List, I am trying to do the following: 1. Use a BandPassFilter h[n] on a series x[n] 2. Then Decimate the series by a factor D such that y[k]=x[k*D] The decimation factor is considerable (10,000) so that filtering before the decimation, using convolve(), seems foolish. Can you think of an R implementation for a polyphase filter, that uses the fact that most samples will be thrown away...?

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

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:

Hi students (non-students may skip this)! As some of you may be aware, the deadline for the Google Summer of Code (http://code.google.com/soc/) program is approaching quickly (March 24th). For those interested, there are two Speex-related projects (below) and lots more in Xiph.Org in general (see http://wiki.xiph.org/index.php/Summer_of_Code). You can apply at

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. I was wondering if there was any interest in extending the flac container format to support sample rates above the 655k current limit. Please note this is not for using ridiculously high sample rate sourced audio for some imaginary audible benefit. I've been involved in some experimentation with offline upsampling in software prior to delivery to an external DAC as a way to bypass the

There is a port for the TI C55X, in the TI directory of the source tree. This builds and runs directly on the TI C55x simulator (Code Composer). I have run this on the Spectrum Digital C5509A EVMPLUS board, by modifying some of the sample audio drivers (I could provide some details if you chose that board). You can certainly get started in the simulator. The Speex port does not include any

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter for audio? thanks, nir -------------- next part

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

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

-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, For transforming stereo to mono, averaging is fine and that's what everybody does. For sampling rate conversion, it's another matter (too long for this email) and you should read a bit about it a perhaps grab a library that does that. As for echo cancellation, it will be less complex (and as good) on a (cleanly) down-sampled signal (and certainly not on stereo). Jean-Marc Le mardi

-----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