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...ig 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 ________________________________________ From: Jean-Marc Valin [jmvalin at jmvalin.ca] Sent: Monday, April 11, 2011 5:29 PM To: Shridhar, Vasant Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] lpcSize On 11-04-11 11:05 AM, Shridhar, Vasant wrote: > I would like to make lpcSize a mul...

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

...g this? I did try and change the value but I seem to be getting some runtime errors indicating a stack corruption. This seems to indicate that I am not allowed to just change the value in the struct. Is there a way for me to alter the lpcSize without having to rewrite sections of code? Thanks, Vasant Shridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20110411/77fbdb4f/attachment.htm

...rsion is limited, mismatch is still exist after resampling. Someone told me that capture and render codec have different clock generator which shift independently. And LMS algorithm is very sensitive to the difference between sample rates. Sincerely Maoquan At 2011-04-12 21:46:26?"Shridhar, Vasant" <vasant.shridhar at harman.com> wrote: I would imagine that it is handle through basic asynchronous sample rate conversion. There is a lot of literature out there on the different techniques to do this. A common method is sinc interpolation. This is how I have handle these types of...

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....

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...

...Re: Playing captured speex frames (Ashhar Farhan) ---------------------------------------------------------------------- Message: 1 Date: Sat, 9 Jul 2011 07:19:10 +0530 From: Ashhar Farhan <farhan at phonestack.com> Subject: Re: [Speex-dev] Playing captured speex frames To: "Shridhar, Vasant" <vasant.shridhar at harman.com>, Vinod Panicker <vinod.p at gmail.com>, "speex-dev at xiph.org" <speex-dev at xiph.org> Message-ID: <CACUhZhWhg9+37Utmj4ioazHiSckZTdj00ZsQbkFE9kEDggDTXw at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Adding...

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

...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 cards. Vas ________________________________________ From: Li Maoquan [limaoquan2000 at 126.com] Sent: Tuesday, April 12, 2011 2:48 PM To: Shridhar, Vasant Cc: speex-dev Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? 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 th...

I have been trying to understand how to get the VAD algorithm working. I sent an input stream of all zeros into the preprocessor but still got a return value of 1 indicating that speech was detected. Is this feature not available with the latest release? I thought at the very least it would detect this as silence and return 0 but that does not seem to be the case. Does anyone have any

...alk. Have GIPS and Microsoft some secret high efficient method? They AECs converge very quickly, I could hardly hear any echo in the process. How can they do it? Maoquan > > ---------------------------------------------------------------------- > > 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 sol...

On Fri, Jul 8, 2011 at 6:39 AM, Ashhar Farhan <farhan at phonestack.com> wrote: > what you need to do is this: take the wireshark raw dump, read each > udp packet and write it back to another file. While writing back to > the new file skip the sizeof udp header + rtp header. I can't recall > how many bytes you need to skip, however, I suppose it would be in the > range of

...feature outside of speex becauses I want a better control on how it happens and when to do something about it :) Yanick Bourbeau On 11-08-29 04:16 PM, Clifton Craig wrote: > +1 on this question as I wanted to start playing with it as well. > > On Aug 29, 2011, at 9:41 AM, Shridhar, Vasant wrote: > >> I have been trying to understand how to get the VAD algorithm >> working. I sent an input stream of all zeros into the preprocessor >> but still got a return value of 1 indicating that speech was >> detected. Is this feature not available with the latest...

...this kind of feature outside of speex becauses I want a= better control on how it happens and when to do something about it :) Yanick Bourbeau On 11-08-29 04:16 PM, Clifton Craig wrote: +1 on this question as I wanted to start playing with it as well. On Aug 29, 2011, at 9:41 AM, Shridhar, Vasant wrote: I have been trying to understand how to get the VAD algorithm working. I s= ent an input stream of all zeros into the preprocessor but still got a retu= rn value of 1 indicating that speech was detected. Is this feature not ava= ilable with the latest release? I thought at the very least...

...this kind of feature outside of speex becauses I want a= better control on how it happens and when to do something about it :) Yanick Bourbeau On 11-08-29 04:16 PM, Clifton Craig wrote: +1 on this question as I wanted to start playing with it as well. On Aug 29, 2011, at 9:41 AM, Shridhar, Vasant wrote: I have been trying to understand how to get the VAD algorithm working. I s= ent an input stream of all zeros into the preprocessor but still got a retu= rn value of 1 indicating that speech was detected. Is this feature not ava= ilable with the latest release? I thought at the very least...

...roblem I passed the following options to the configure script. --with-ogg-libraries=<libogg_root>/src/.libs --with-ogg-includes=<libogg_root>/include. In this case I built libogg and left all of the files in the default place (so I just did make not make install). Hope this helps. Vasant Shridhar -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of Luca Ottaviano Sent: Wednesday, August 10, 2011 12:18 PM To: Rony Nandy Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] exiting with ogg.h missing On mer, 2011-08-10 at 0...

...better control > on how it happens and when to do something about it :) > > Yanick Bourbeau > > > > > > On 11-08-29 04:16 PM, Clifton Craig wrote: > > +1 on this question as I wanted to start playing with it as well. > > On Aug 29, 2011, at 9:41 AM, Shridhar, Vasant wrote: > > I have been trying to understand how to get the VAD algorithm working. > I sent an input stream of all zeros into the preprocessor but still got a > return value of 1 indicating that speech was detected. Is this feature not > available with the latest release? I though...

Hi All, I am trying to use VAD opption of speex encode on STM32F4 chip. But when I use *speex_preprocess_run *function, It allways return 1. Anyone can show me the problem? And please, give me an example about using VAD. Thanks in advance, Regards, Nhan. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...rs me is that I'm able to send messages to other people using Windows or other Linux distros like Gentoo or Red Hat but neither of the 3 LFS users in my knowledge on my LAN are able to send messages to each other. I mean only LFS machines seem to not recieve the messages. -- With Regards Vasant Tiwari @ QOTD: "Don't let your mind wander -- it's too little to be let out alone."

...Valentina Manea <valentina.manea.m@gmail.com> Valerie Aurora <val@vaaconsulting.com> Valerie Aurora <vaurora@redhat.com> (Original author) Vandana Salve <vsalve@nvidia.com> Varun Sethi <Varun.Sethi@freescale.com> Varun Wadekar <vwadekar@nvidia.com> Vasant Hegde <hegdevasant@linux.vnet.ibm.com> Vasanth Ananthan <vasanth.a@samsung.com> Vasanthakumar Thiagarajan <vthiagar@qca.qualcomm.com> Vasiliy Kulikov <segoon@openwall.com> Vasily Averin <vvs@openvz.org> Vasily Khoruzhick <anarsoul@gmail.com> Vasily Ku...