search for: tgrand

...e. But, it's not quite clear as to how this terminator code is used... especially since it refers to figure 4 which doesn't seem to exist. (doc bug?) Might I humbly recommend a section of documentation elaborating on the use of the speex_bits*() calls ... Tom <p>Tom Grandgent (tgrand@canvaslink.com) wrote: > > Hi, > > I have a question about section 3.4 of the RTP payload draft: > > > Speex codecs [11] are able to detect the the bitrate from the > > payload and are responsible for detecting the 20 msec boundaries > > between each frame. >...

..."George Ou" <george_ou@lanarchitect.net> wrote: > > Only thing that bugs me about Speex for podcasting is that it's geared for > 8, 16, or 32 KHz. Most podcasters are set up for 44.1 KHz recording. > > -----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 > Subject: Re: [Speex-dev] How to get podcasters to adopt Speex? > > Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase > in a...

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

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

...hy it's good to be able to set the max AGC gain. One of my users reminded me of it tonight: he wanted to be able to limit the max AGC gain so that when he's not speaking, the AGC doesn't pick up on other sounds from the room. Tom ---- Original Message ---- From: stevek@stevek.com To: tgrand@canvaslink.com Subject: Re: [Speex-dev] Proposed AGC additions Date: Fri Jul 2 20:11:34 2004 >Tom, > > Are you also using the VAD from the preprocessor? Does the VAD >detect this noise as speech? > >I've got this same thing out through a lot of testing (via >iaxclient...

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

...ially since it refers > > to figure 4 which doesn't seem to exist. (doc bug?) > > > > Might I humbly recommend a section of documentation elaborating on > > the use of the speex_bits*() calls ... > > > > Tom > > > > > > Tom Grandgent (tgrand@canvaslink.com) wrote: > > > > > > Hi, > > > > > > I have a question about section 3.4 of the RTP payload draft: > > > > > > > Speex codecs [11] are able to detect the the bitrate from the > > > > payload and are responsible fo...

Yes, you can use it independently by using speex_preprocess(). This function can do noise removal, AGC, and VAD. I've been using it to do all three and it usually works very well. There can be a train going by outside, producing lots of sound that makes it through the noise filter, and yet the VAD knows it's not speech. However, sometimes the VAD seems to get into a bad state and

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 Subject: Re: [Speex-dev] How to get podcasters to adopt Speex? Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is sti...

...s force the customers run some kind of wizard, before the running stage of our software, while (secretly) building up the profile, hoping they do not adjust their gain level etc later. Any comments (experience) on this, from anyone ? <p>-----Original Message----- From: Tom Grandgent [mailto:tgrand@canvaslink.com] Sent: Thursday, September 18, 2003 11:25 AM To: speex-dev@xiph.org Subject: Re: [speex-dev] speex_denoise on non-microphone noise (static ?) Hi, I'm not sure how speex_denoise() works, but my solution to this problem is fairly straightforward and works well in practice (in a...

I've been using Speex's AGC in my VoIP program with mixed results. One of the problems I had was: At the beginning of the call, if the user hadn't spoken yet and there weren't any background sounds for the AGC to lock on to, most of the time the AGC would almost immediately jump way up to 200x gain (the default max). This is undesirable and usually results in nasty feedback. If

...ole 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 Grandgent" <tgrand@canvaslink.com> Cc: <speex-dev@xiph.org> Sent: Friday, March 03, 2006 6:55 AM Subject: RE: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [...

...ome > kind of wizard, before the running stage of our software, while > (secretly) building up the profile, hoping they do not adjust their gain > level etc later. Any comments (experience) on this, from anyone ? > > > -----Original Message----- > From: Tom Grandgent [mailto:tgrand@canvaslink.com] > Sent: Thursday, September 18, 2003 11:25 AM > To: speex-dev@xiph.org > Subject: Re: [speex-dev] speex_denoise on non-microphone noise (static > ?) > > Hi, > > I'm not sure how speex_denoise() works, but my solution to this problem > is fairly str...

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Jean-Marc, I think I can speak for most everyone here when I say we appreciate *very much* your code beyond the codec that helps people who use Speex in real applications. How far you want to take that code is up to you, of course. (And contributors, to a degree.) I don't think any reasonable person is expecting your code to solve everything. But when people are running up against

[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