similar to: What Happened to www.speex.org?

Fabio, You can't mix encoded audio directly. You would have to decode, mix, and reencode. There will likely be some loss of quality. Tom Fabio <fabpet@gmail.com> wrote: > > Hi to all, > > I developed a VoIP application using Speex. Now i want to record the > conversation to a file. > As i have two indipendent thread transmitting e receiving voice i > whish to

Ok, I figured it out. :) This seems to work: 1) Call speex_bits_read_from() once, specifying the location in memory of the compressed data, and the total length of that data. 2) Keep calling speex_decode() until speex_bits_remaining() returns 0. Then you don't have to keep track of the # of frames per packet, or the size of each compressed frame. It's done magically by the codec.

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

It's been down the last 5 hours at least. Anyone know what the problem is, or when it will be back up?

Not sure if this is specific to me or what... but I cannot access the site. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20060414/62b41250/attachment-0001.html>

Ohh... Nice! This is new in 1.0.1, isn't it? It doesn't seem to be included in the reference manual yet, though. Thanks! Tom <p>Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > OK, this is how it works: > > The encoder calls speex_encode any number of times and then calls > speex_bits_insert_terminator before sending the bits. > > The

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

Ok here is something I couldn't see on the description page. We need to convert to 8 bit audio right? I'm assuming narrow band is 8 KHz 8 bit. Wide band is 16 KHz 8 bit. UW band is 32 KHz 8 bit audio? George -----Original Message----- From: Tom Grandgent [mailto:tgrand@canvaslink.com] Sent: Tuesday, October 03, 2006 11:42 AM To: Bertie Coopersmith; George Ou Cc: speex-dev@xiph.org

I am glad to announce the BETA release of the Speex ACM codec. It is based on the Speex 1.0.1 codec release. An ACM codec enables Windows to create and play back WAVE files containing Speex content. It might also be useful for use with Netmeeting (after some registry hacking. Not tested yet.). I also provide a portable playback tool (speexdec_wav) that handles both the Ogg and WAVE containers.

It would be very nice to know about some good success without too much cpu issues on Nokia devices, i tried pjsip.org project on S60 FP1 E65 +200mhz ARM cpu with speex but don't had much luck in using it without having 100% cpu usage :( Please let us to know about your testing E65 CPU: http://www.nokia-tuning.net/index.php?s=processor Fabio Pietrosanti Jordan Dimov wrote: > Thank

hi Tom, i downloaded your speex wrappers and they're really what i'm looking for! i really cannot worry about container support at the moment because i'm working with real-time audio, and hopefully toward VoIP when everything else is in order. im using waveform to capture data from the microphone. the raw PCM sample is stored in the WAVEHDR structure's member lpData, and this

Quoting "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>: > Speex it's too cpu expensive for general S60 usage, it would require a > lot of ASM optimization. Did a quick search and saw ARM CPUs with speeds above 100 MHz. That should actually be enough for Speex, at least for narrowband. > If you are using CSD, like for a secure telephony solution >

Hey guys and gals, Anyone know of a half decent tool like DNSstuff.com only free? I need to run some diag on a few domains but it is basically a 1 shot deal and hard to justify buying. thanks, -Alan -- ?Don't eat anything you've ever seen advertised on TV? ? ? ? ?? - Michael Pollan, author of "In Defense of Food"

Sorry, the 3rd party and (AFAIK) windows builds are still 1.1.6. I have no idea what compiler was used. Of course ARM4_ASM was not used, since the option didn't exist in 1.1.6. Jean-Marc Le samedi 05 mars 2005 ? 13:04 +0100, Fabio a ?crit : > Hi all, > Are the 3rd party builds in page > http://www.speex.org/download.html > based on Speex 1.1.7 ? > > Thanks > Fabio

you can use the marker bit on the RTP. - farhan On Tue, Aug 19, 2008 at 12:04 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote: > Hi all, > > supposing to have speex working at 3.95, when VAD/CNG is employed, which > is the size of those "silence" samples respect to voice samples? > > I am wondering if it could be possible to lower even the

Actually, it looks like this is already done in Speex as of revision 7150 in SVN (7/15/2004). As long as HAVE_CONFIG_H is defined in the project settings (which it is in all of the project files in SVN), win32\config.h should be included, and this has #define inline __inline in it. Oleg - you should make sure HAVE_CONFIG_H is defined in your project settings. And, you didn't say which

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

Hi all, supposing to have speex working at 3.95, when VAD/CNG is employed, which is the size of those "silence" samples respect to voice samples? I am wondering if it could be possible to lower even the transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

Hello, I am having an issue with email on all IP-based site/accounts that I set up on CENTOS 6.3 x86_64 standard ? WHM 11.34.1 (build 5). On the server's main IP address domain account/site, both sending and receiving email works just fine, however, on all IP-based accounts/sites I create the email users can send email but not receive it, and dnsstuff.com shows: All connections to