similar to: Hi speexdev

Hi Sanjiv, There does indeed appear to be some trouble with the Javasound SPI encoder (the SPI decoder works well). I am working on it, and hope to have a fix soon. In the mean time, you can use the encoder outside of JavaSound, that one works well (see the command line encoder as example code of how to implement it). sincerely Marc Gimpel Head of research Wimba <p>On

? hi all, i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" in my VOIP application. Is it that the frame length to be chosen should always be 20ms or can i have flexibility in chosing the frame lenght? on what parameters does the frame length choise depends? thank you all, Shri. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Jean-Marc Valin <Jean-Marc.Valin <at> USherbrooke.ca> writes: > > Le mercredi 07 juin 2006 ? 10:03 +0000, shridhar desai a ?crit : > > > > hi all, > > i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" > > in my VOIP application. Is it that the frame length to be chosen > > should always be 20ms or can i have

Speexdec plays ogg files. You can simply modify it to play your raw file or even rtp capture. Jozsef -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of speex-dev-request at xiph.org Sent: Saturday, July 09, 2011 12:00 PM To: speex-dev at xiph.org Subject: Speex-dev Digest, Vol 86, Issue 3 Send Speex-dev mailing list submissions to

Okay, Not exactly the answer I was looking for. This sounds like a big change. I don't mind re-writing the LSP quantizer but re-training code books and breaking compatibility is not what I want to do. I am working on an optimization for an ARM cortex-A8. It is desirable to process things in 4 element blocks. Is there a simpler approach you could recommend? Vasant Shridhar

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

On mer, 2011-08-10 at 09:41 -0400, Rony Nandy wrote: > Hi All, > I have downloaded libogg-1.3.0 along with speex.But,during > build speex is exiting with ogg.h missing.Any suggestions will be highly > appreciated. IIRC, speexenc encodes your data into a speex stream which is encapsulated into an OGG container, so you need to libogg to compile it. Though, it has been ages

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Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

All: The latest draft RTP Payload Format for Speex is available via the IETF. See below for details. Greg ---------- Forwarded message ---------- Date: Tue, 09 Mar 2004 15:56:23 -0500 From: Internet-Drafts@ietf.org To: IETF-Announce: ; Subject: I-D ACTION:draft-herlein-avt-rtp-speex-00.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. <p>

Hi, I think the page http://www.speex.org/software/ should include a link to http://nspeex.codeplex.com/ which is an open source port of JSpeex to .NET. I tried to mail this request to webmaster at speex.org but something went wrong Best regards, Jader Dias ---------- Forwarded message ---------- From: Mail Delivery Subsystem <mailer-daemon at googlemail.com> Date: Sun, May 30, 2010

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I would like to make lpcSize a multiple of 4. In the current library for narrow band mode lpcSize is currently set to 10. I would like to increase this 12 for optimization reasons. Is it a simple matter of just changing lpcSize in the structures in mode.c or are there other implications to doing this? I did try and change the value but I seem to be getting some runtime errors indicating a

On 11-04-11 06:48 PM, Shridhar, Vasant wrote: > Okay, > > Not exactly the answer I was looking for. This sounds like a big > change. I don't mind re-writing the LSP quantizer but re-training > code books and breaking compatibility is not what I want to do. I am > working on an optimization for an ARM cortex-A8. It is desirable to > process things in 4 element blocks.

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

Hello, I tried to use speex echo canceller for a live performance setup to cancell feedback. My program buffers input until collecting a speex frame size of samples, then calls speex_echo_cancellation passing this buffer as input signal and previous speex output as echo signal, then immediately plays current speex output. First thing I noticed was huge high pass filtering. Then I found out there

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Hello! I hope this Email is still available. My goal is to cancel the echo caused by the client's mic and speaker through the Internet, which is also the aim of Speex. I have read the manual and the api reference, but still have some problems about the usage. 1. about function speex_echo_cancellation(echo_state, input_frame, echo_frame, output_frame); Parameters (input_frame,