similar to: want rsync -v to report perm/mode changes

I am forwarding your request to the FLAC development mailing list. ----- Forwarded message from Jamie Heilman <jamie@audible.transient.net> ----- Date: Sun, 8 Sep 2002 16:13:32 -0700 From: Jamie Heilman <jamie@audible.transient.net> Resent-From: Jamie Heilman <jamie@audible.transient.net> To: submit@bugs.debian.org Subject: Bug#160155: gapless playback Package: xmms-flac

Hello all, I saw on the Release notes that 2.6.0 is using protocol version 27 which is different than 2.5.7 and 2.5.6. I did a quick peak through the rest of the release notes and quickly browsed the archives on the list and I didn't see any notes on how well 2.6.0 works with older versions of rsync. At our site, we do have 2.5.7 and 2.5.6 installed. Are there any documented problems when

Hi, I have a couple of samples that produce interesting artifacts when encoded with the CVS snapshot of 2001-02-13. Both are about a meg. ftp://slumber.dhs.org/tmp/4.wav.bz2 When encoded with oggenc -b 128, there is a sort of stereo separation in the drums, while in the original they are 'solidly' positioned in the stereo field. This also occurs with -b 160 and is barely audible with

Tevfik On Tuesday 09 Dec 2003 18:13, Tevfik Karag?lle wrote: > I would also like to inform that I updated my cwRsync package with the > latest rsync binary (2.5.7). http://itefix.no/cwrsync I have tried your rsync 2.5.7 in cygwin under Windows 2000, synchronising to a windows share on an unknown (probably Windows) file server. Sorry to report that it hangs at the end of transfer, as did

For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to

On Nov 05 11:32:49, hans at stare.cz wrote: > On Nov 05 11:05:34, hans at stare.cz wrote: > > > Did you also try to listen at the beginning, shortly before the real tone appears in the audible spectrum? While significantly larger, Opus had produced significant ghost noise (much less than Vorbis did)... I experience the "same" low level noise even in a wav file, even on

So I got a new Pocket PC this weekend (yes, you read that right, sorry, don't like the Palms on offer currently and I can at least install software on it via Windows on VMWare). One of the main things I wanted to do with it is get Audible.com again and listen to All Things Considered from NPR on the way home like I used to. Now I find out they not only don't have an agreement with Audible

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

Sorry for disturbing you, but we have some problems with an installation with multiple (84) T1s from Quest. Now, our Problem is disconnected numbers are reported by sending in- band channel alert message and the B-Channel will have the tri tone and respective message but the line is never "picked up" and stays in ringing when dialling. So disconnected numbers are never detected as

It will not let me authorize my books. I saw a post on SSL issue. But, it did not mention the current wine ver or Audible ver. I am using Wine 1.3.1 Audible 5.5.0.4

Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the

Hi, there. Although I'm not a hacker at all, I noticed an audible noise on Beta 4 encoded files. So, I'd like to report it here. I ripped and encoded 'A Day Without Rain' on the album with the same name By Enya into Vorbis using Oggenc and Oggdrop Beta 4 from the Vorbis site. (You know, she is quite an encoder-killer.) At all the bitrates, files encoded by Oggdrop had audible

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

I'm using server on OpenBSD 3.4 to backuping data from remote server via rsync.On backup server I am install rsync-2.5.6 and on remote machine rsync-2.5.7 (RedHat 8.0) . For backuping data I'm using the following command : rsync -av -e ssh backup@log:/var/backup/ALL-15-Mar.tar /var/backup/3 This command copy file ALL-15-Mar.tar from host log(RedHat box) to local directory on OpenBSD.

Hello, We've integrated the speex echo canceller into PulseAudio, and coupled with the AGC/denoiser, it works reasonably well and has been a breeze to plug in, so thank you! One thing we're seeing, though, is that the canceller seems to take some time (a few seconds) to "learn", so initially the echo is clearly audible and it slowly fades out. This sometimes occurs in the middle

I noticed that clipping occurs in lossy audio compression even if there is no clipping in the original file (though the original file has peaks that are just below the maximum). I know that this happens due to all the filtering involved during compression, but I'm wondering just how audible this clipping is because I don't hear anything wrong. I mean, as things are now, is it safe to say

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft

On 29 April 2015 at 17:22, Timothy B. Terriberry <tterribe at xiph.org> wrote: > > Viswanath Puttagunta wrote: >> >> This patch series is follow up on work I posted on [1]. >> In addition to what was posted on [1], this patch series mainly >> integrates Fixed point FFT implementations in NE10 library into opus. >> You can view my opus wip code at [2]. >

I think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():