similar to: First frame is fading in (?)

I believe if you add the *decoder* lookahead time to the encoder lookahead time you are already referencing, you will get the numbers that you have calculated. --John On 4/11/2012 1:00 PM, speex-dev-request at xiph.org wrote: > Message: 1 > Date: 10 Apr 2012 22:07:20 +0200 > From: Thilo K?hler<koehlerthilo at gmx.de> > Subject: [Speex-dev] Speex Codec Delay Problem > To:

Hello All! SPEEX introduces an additional delay to the audio data, I found out by reverse enginiering (it is NOT the lookahead time): narrow band : delay = 200 - framesize + lookahead = 200 - 160 + 40 = 80 samples wide band : delay = 400 - framesize + lookahead = 400 - 320 + 143 = 223 samples uwide band : delay = 800 - framesize + lookahead = 800 - 640 + 349 = 509 samples To get the

Dear Jean-Marc and Peter: Thank you both very much for your time and advice. I did not realize that Lame MP3 code has a -r option (without reading its code). I have tried the suggested command lines verbatim with the added -r option, along with other combination of option settings. I could not figure out how to eliminate the distortion in the result, like voice turning either high or low

>jesus: > Now I see that it happens when I specify a framesize in > speex_echo_state_init greater than the one used in > speex_preprocess_state_init. > > Next combination works: > codec framesize = 160 > preprocessor framesize = 160 > eco framesize = 160 > > > Next combination DOESNT work (produces heap corruption): > codec framesize = 160 >

Hi, I am using speex 1.2beta2 on a 24bit DSP that has a severe program and data space limitations. I am only interested in the speex decoder for 11kbps narrowband implementation. I am using the following parameters and structures 160, /*frameSize*/ 40, /*subframeSize*/ 10, /*lpcSize*/ 17, /*pitchStart*/ 144, /*pitchEnd*/ /* 11 kbps medium bit-rate

I've had several users today inform me that whilst they were on a call, the volume kept fading in and out to such an extent that they thought the caller had hung up. I would dismiss this if it were a single person mentioning it, but it isn't .. Has anyone else seen anything like this ? We haven't, and we've been running for nearly a year ... CVS head as of 3 days ago, TE410p

Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from people. Soon I figured some network problem, I did a tcpdump and analyzed by wireshark ...the strange thing is this ... all packets that

Hi I am trying to find a possible cause of 'audio fading in and out' which effects about 0.1% to 0.15% of calls placed to a voip provider for termination. (I am using SIP & alaw throughout in this case) I don't believe this to be something that is network / jitter etc related (?) and looking at historical posts this may be a telco issue (voltage?) at the voip provider gateway.

Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. I've got a user who either can't be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me.

Hi: To do fading, you'll need to use a source that will: 1. Decode your songs 2. Do the necessary crossfading 3. Re-encode the stream and send it to icecast. right now, I know of no streamer that will do this. I'm aware of a very comprehensive package under development, but there's no public access to it yet that I know of. Note that there are strong feelings on both sides as

HI all , i'm using icecast1.3.12/IceS0.23. Is it possible to "fade" songs ? i mean , begining the following song a few seconds before the previous ends ? (Tell me if it's necessary to upgrade to icecast2/ices2 ) Thanks in advance .... <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this

> Any idea or suggestion to avoid having two sound cards to do what I need to > do ? > > Or how to fade in / fade out without using the card' mixer, while playing > i.e. using mpg123 ? Depends on how much you want to program. Basically, the easiest would probably be to use OSS software mixing; if you buy the commercial OSS drivers, you can have several programs writing to

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

On Tue, 2003-01-14 at 12:29, gtgbr@gmx.net wrote: > Scott Manley wrote: > [crossfading] > > well ... liveice has done this for the past 5 years.... but it's not an easy > > package to use > > One could add Winamp based streaming to this list - by using a > crossfading plugin and using a feature like Oddcast DSP's "Adv. Rec". thankyou ... but i run only

well ... liveice has done this for the past 5 years.... but it's not an easy package to use ----- Original Message ----- From: "Geoff Shang" <gshang@uq.net.au> To: "icecast mailing list" <icecast@xiph.org> Sent: Monday, January 13, 2003 10:59 PM Subject: Re: [icecast] Fading between songs <p>> Hi: > > To do fading, you'll need to use a source

I found a problem with trying to use the speex plugin for xmms on my PowerBook G3 'Pismo' PPC machine, running kernel 2.4.19. It turns out that the kernel sound driver ignores the endian of the data format flag and instead expects the data to be native endian. This is the same problem that was affecting speexdec. I have now fixed this problem, and another compilation issue that I found. I

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Hi, there is a suitable HowTo on how to create your own CA at: https://checkmk.com/de/blog/how-become-your-own-certificate-authority So long Thom Am 28.05.24 um 09:20 schrieb Matthias K?hne | Ellerhold Aktiengesellschaft via samba: > Hello Thomas, > > we've done the exact same thing: we have a few nextcloud instances bound > to Samba (now 4.20, but 4.19 worked too). >

Hi, I've been trying to encode a live audio input from the microphone on iOS device using opus. Uncompressed audio recording works fine with http://theamazingaudioengine.com/ Then, when I tried to do encoding, I'm stuck at figuring out why the buffer is empty: static int ec_write_byte(ec_enc *_this,unsigned _value){ if(_this->offs+_this->end_offs>=_this->storage)return

Hello R experts, I have two questions related to the nnls library (http://www.inside-r.org/packages/cran/nnls), and more broadly to linear regression with positive coefficients. Sample code is below the Qs. Q1: Regular regression (with lm) gives me the significance of each variable. How do I get variable significance with nnls? If there's no ready function, any easy way to manually derive