similar to: computing speex packets duration

Hi, We'd like to announce the first draft for the Speex RTP profile. It was written essentially by Greg Herlein, with some help from Simon Morlat and I. We'd like to get some feedback on it before it is sent to the IETF. Basically this will allow all SIP based VoIP applications using Speex to inter-operate. For those interested, there's already Simon's LinPhone (www.linphone.org)

Ronald, Although you can give Speex data at any sampling rate with any mode, it's a good idea to use the right mode for whatever rate you have. Speex is built for speech (not just any audio in general) and is sensitive to what kind of stuff is going on in certain frequency ranges. If you tell it your data is at one sampling rate and give it data with another sampling rate, its idea of

Hi all, I have a strange problem and rigth now I can't figure out a solution. Trying to calculate an ANOVA with one between subject factor (group) and one within (hemisphere). My dependent variable is source localization (data). My N = 25. My data.frame looks like this: > ML.dist.stack subj group hemisphere data 1 1 tin left 0.7460840 2 2 tin left

> 1) What should I use for a filter length for speex_echo_state_init()? Well, it depends on your problem. The filter length represents the maximum delay of the echo. For acoustic echo, that value is generally higher than for line echo. You'll need to experiment, but a value around 500 (for 8 kHz) might be a good start. > 2) I'm assuming I want to call speex_echo_cancel() on each

Hello ppl! I asked some time ago how about adding the option to control ogg123 interactively from command line, similar to mpg123. So I got fed up waiting and added it myself today. The code initially comes from mpg123, I already adapted it to work with aviplay and vlc so it's proven and shouldn't introduce many problems. The only problems I see now: - in order to compile ogg123 now

Whoops, left this message in my outbox. I managed to fix the problem. Apparently I was only copying 160 bytes (Frame Size) back into the audio stream when I should have been copying 320 (chars <-> shorts confused me there). Hence why I could hear myself yet it was distorted. Half the wav was missing =) To answer some of the other questions here, for any insight into what I'm doing:

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

Hello, I don't really understand how channel Local works. I need that asterisk initiate a call and get some data (DTMF). So to do that I've created this dialplan : ; extensions.conf - the Asterisk dial plan ; [general] static=yes writeprotect=no clearglobalvars=no [dtmf] exten => 1,1,Verbose(Get User ID) exten => 1,n,Dial(dahdi/1/99999999,120,G(read^1^1)) exten

Hello, I would like to run a script over a speex file which determines, how long the speech contained in the speex file is. So I run through the ogg file, skip the first two ogg _pages_ (speex header and comment header, both contain IMHO only 1 ogg _packet_), and then count in any ogg page the number of the ogg _frames_ (which, unless I used -nframes while encoding, which I did not), and multiply

Hello everybody, I'm new to this mailing list so I'm sorry if it's the wrong place to post this. I'm developing a Speex player and I need to seek with a precision of milliseconds. I used liboggz that supposedly does just that, but it never seeks exactly where it should. For example if I use oggz_seek_units(oggz, 18450, SEEK_SET) result it's 16386 and there is a delay between

I'm trying to install my first channel of FAX. When I run the benchfax utility, I get to various stages and then the program simply hangs. There is no excessive CPU utilization and the benchfax program readily responds to ^C. Sometimes I get fairly far into the test, other times it hangs almost immediately. The version of benchfax version I'm using is 1.0.7. I heard tell of a newer

On Tue, May 16, 2006 at 03:29:47PM +0200, Bj?rn Thalheim wrote: > > I would like to run a script over a speex file which determines, how > long the speech contained in the speex file is. > So I run through the ogg file, skip the first two ogg _pages_ (speex > header and comment header, both contain IMHO only 1 ogg _packet_), and > then count in any ogg page the number of the ogg

2009/5/14 Arnau Alemany <arnau.alemany at gmail.com>: > Hello everybody, > I'm new to this mailing list so I'm sorry if it's the wrong place to post > this. > > I'm developing a Speex player and I need to seek with a precision of > milliseconds. I used liboggz that supposedly does just that, but it never > seeks exactly where it should. For example if I

Sounds great!! I will definitely give these new features a try and let you know how it goes. Every one of them (except for the narrowband mode) will be really useful to me. I have a few questions after looking through the source code: Echo cancellation: 1) What should I use for a filter length for speex_echo_state_init()? 2) I'm assuming I want to call speex_echo_cancel() on each

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

libswfdec/swfdec_as_date.c | 16 ++-- libswfdec/swfdec_as_interpret.c | 146 ++++++++++++++++++++++++++++++++++++++-- libswfdec/swfdec_movie.c | 43 +++++++---- libswfdec/swfdec_movie.h | 2 libswfdec/swfdec_sprite_movie.c | 37 ---------- 5 files changed, 179 insertions(+), 65 deletions(-) New commits: diff-tree 89d295a9455fead858e48ffcae0cc5808d72ee22 (from

I am having some trouble calculating the duration of my speex files. I am writing a speex player in Java and I am using the JSpeex port of the codec which works but doesn't give me the audio duration which I need to show progress or implement a scrubber. JSpeex appears to be all but dead but I don't see any other option for Java playback. I have tried the method posted here:

--- filters/delay/nbdkit-delay-filter.pod | 8 ++++++++ filters/delay/delay.c | 26 ++++++++++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/filters/delay/nbdkit-delay-filter.pod b/filters/delay/nbdkit-delay-filter.pod index c2eb172..2e2ac74 100644 --- a/filters/delay/nbdkit-delay-filter.pod +++ b/filters/delay/nbdkit-delay-filter.pod @@ -11,6 +11,7 @@

> Otherwise Greg, can you send the latest version of the RTP draft so I > can put it on the site (the current one is getting old)? Attached for all to see. Greg -------------- next part -------------- Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin draft-herlein-speex-rtp-profile-06