similar to: In which release did FLAC support 192kHz sample rate?

Hi Josh, On Sat, 2004-10-09 at 22:41, Josh Coalson wrote: > --- Marek Peteraj <marpet@naex.sk> wrote: > > Hi all, > > > > i'd like to ask what the best options are for safe 24bit 96kHz > > archiving. Currently i'm only using -8 but there are also some other > > options like block size etc. Can anyone suggest? > > I haven't heard of

Hi all, i'd like to ask what the best options are for safe 24bit 96kHz archiving. Currently i'm only using -8 but there are also some other options like block size etc. Can anyone suggest? Also i'd like to ask whether a no-lax 192kHz mode is planned in flac, seems like 192kHz isn't directly supported although flac can compress such rate in lax mode. My last question - is there

--- Marek Peteraj <marpet@naex.sk> wrote: > Hi all, > > i'd like to ask what the best options are for safe 24bit 96kHz > archiving. Currently i'm only using -8 but there are also some other > options like block size etc. Can anyone suggest? I haven't heard of anyone using FLAC for 192khz, so you'll probably just have to experiment. when you have some

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

Hi. I've been browsing the archive on this topic and only found a few notes, all dating from a year ago (almost too precisely :) ) -- hope I haven't skipped the mails on that matter, sorry if I did. <p>Basically, I will get in the next few months a MOTU 896, that have 8 i/os in 24/96. I do pro sound recording, so it's more or less my business to have such a piece. Of

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for

Dale, Sorry for taking so long to answer, I've been traveling. Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dnoll at wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk

On Aug 26, 2013, at 2:39 PM, Stephen Crane <sjcrane at uci.edu> wrote: > We have been working on adding randomness into code generation > to create a diverse population of binaries. This diversity prevents > code-reuse attacks such as return-oriented-programming (ROP) by > denying the attacker information about the exact code layout. Putting on my security hat (as opposed to

Hi people, looking around for a new audiocard, my eye fell on the M-audio audiophile 2496. It has 4 digital in/out and is 24bit, 96kHz. The sound quality is very good, if I can believe the reviews. <p>My question is: can vorbis do 24bit, 96kHz ? -- --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,

Hi! On z/OS, there is the need to convert strings from EBCDIC to UTF-8 and vice versa. Using the POSIX iconv functions has some challenges, so I created a small wrapper around this functionality to get the same result on all platforms. This functionality is required for reading and writing GOFF object files and can also be used in the frontend. I put up the code on Phabricator

Please don't top-post. On Fri, 26 Jun 2015, Dale Noll wrote: > I turned on the messages that he had in the file again, all the logs were in /var/log/asterisk and it does not show anything for syslog. > asterisk -rx 'logger show channels' > Channel ? ? ? ? ? ? ? ? ? ? ? ? ? ? Type ? ? Status ? ?Configuration > ------- ? ? ? ? ? ? ? ? ? ? ? ? ? ? ---- ? ? ------ ?

>>> Jan Stary <hans at stare.cz> schrieb am 05.11.2018 um 11:05 in Nachricht <20181105100534.GB44329 at www.stare.cz>: > (Are we off‑list now by intention?) No, just fooled by the list defaults (some need just reply, others need reply to all) > >> Did you also try to listen at the beginning, shortly before the real tone > appears in the audible spectrum?

Hi! Implementing the Memory::allocateMappedMemory() function on z/OS, I see a failure in the AArch64 COFF test case. The test case has 3 sections. For each section, Memory::allocateMappedMemory() is called to reserve memory. If the distance between the pointers gets too large, then the test case fails. It can be reliable produced with a distance of 1MB between the pointers. An easy way to

I came across an interesting test sample the other day while trying to compress some of my music and trying out different encoders. You can grab the sample in question from here: http://www.animus-facticius.org drone_clip.zip is a 9 second clip of the track where the artifact is most apparent, drone_short.pac is a 1 minute clip of track. The sample is from the song "Drone" from

On Sun, 23 May 2004, Conrad Parker wrote: > On Sun, May 23, 2004 at 07:04:28AM +1000, Kenji Chan wrote: > > Is that ok for vorbis to encode 24bits/96KHz audio? > yes, oggenc should be able to handle it. Not only does it handle it fine: but most players will downsample the 96k audio for playback on non 96k able hardware. (24bittage isn't an issue as Vorbis is 24bit internally,

Pam_SMB allows Linux clients to validate their passwords against an NT PDC, so the only thing you have to do is set up the accounts on the Linux side with an '*' in the /etc/passwd entry. This can be done using a list of users: #!/bin/bash for i in `cat myuserlist`; do /usr/sbin/adduser -p '*' $i with various other command line options, such as "-s

I've been reading through a few forums about ripping DVD audio / Sony SACD's (I'm not sure if these are the same or if they are different formats). Anyway, being that I haven't found a way to rip this stuff yet as there may not be any consumer product out yet that enables the feat, I have been curous about FLAC's ability to encode in a 6 channel format at the high bitrate

I recently bought a DVD Audio disc and I searched the web and wasn't really able to find much about DVD Audio and Linux. I have a few questions that I was hoping some people on this list might be able to answer. How can I extract the uncompressed DVD Audio (not the DTS Audio portion of the disc, but the 96khz/24bit/5.1 pcm part)?

Jaren Stangret wrote: > Interestingly, I noticed compression level '0' is omitted, so I've added it > in. Thanks. > Also, I tested this on a 24bit/96kHz FLAC file and the test failed between > compression level 0 and compression level 1 -- the file size was greater > with a compression level of 1, rather than 0. The test works on > 'noisy-sine.wav'. It