similar to: libvorbis 1.2.3 not generating 96kHz ogg file

Hi I have built and installed ffmpeg 0.6, Libvorbis Version 1.2.0 and, VP8 (libvpx-0.9.1). When I runn ffmpeg on the command line everything appears to work fine. I give it an input webm file downloaded from youtube.com and generate a different ouput webm file. However when I try and encode the same webm input file hosted on my local HTTP server using the ffmpeg API over a HTTP stream I get the

I''m trying to install CiscoWorks 2.5 on Solaris 10 update 1, and after the install when I try to start the daemon, it errors: # /opt/CSCOpx/objects/dmgt/dmgtd.sol ERROR: open file dmgtd failedERROR >>>>>>>>>>>>> open msg catalog failed. NLSPATH incorrect or objects/share/nls/C/dmgtd.cat is missing. # echo $NLSPATH

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

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,

Well, you seem to be using FRAME_SIZE but only defining frame_size. Otherwise, the code looks OK, but it's always hard to tell. I suggest you start from speexenc/speexdec or from the example I wrote in the manual at: http://www.speex.org/manual/node12.html Jean-Marc Le ven 19/12/2003 à 05:22, Fabio a écrit : > Hi > i'm developing a sort of VoIP application > for my

On Fri, Jul 20, 2001 at 08:14:55PM -0700, Josh Coalson wrote: > --- Matt Zimmerman <mdz@debian.org> wrote: > > On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > > > > > This version seems to work at least partially on ia64. I am able > > to encode my > > > usual test WAV file now, but I still get a segfault during the > >

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

--- Matt Zimmerman <mdz@debian.org> wrote: > On Fri, Jul 20, 2001 at 08:14:55PM -0700, Josh Coalson wrote: > > > --- Matt Zimmerman <mdz@debian.org> wrote: > > > On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > > > > > > > This version seems to work at least partially on ia64. I am > > > able > > > to

On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > This version seems to work at least partially on ia64. I am able to encode my > usual test WAV file now, but I still get a segfault during the self-tests. Interestingly enough, when I recompiled with --enable-debug to get a stack trace, it worked. Any ideas how to track down the problem? -- - mdz

Here is my situation: I have large Speex encoded stream which I would like to playback on wave out device. I could decode whole stream to pcm and play it back as pcm audio, but this is not what I want. Instead I want to decode minimum decodable bits (bytes?) to pcm and play it ... then decode another minimum data chunk and play it, etc. This way I can achieve closest to real-time playback or

I found a bug with FLAC v0.6 raw encoding. It appears that the file pointer in the source file is not reset after seeking to the end for checking the size. I've attached a patch. I'm excited about FLAC!! I've been looking for a good GPL lossless RAW audio compressor for use with sound fonts. Sound font files contain 16 bit samples that are word aligned, so just treating it as raw

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,

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

>>> 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?

(Please wrap your lines.) On Oct 26 01:38:34, Ulrich.Windl at rz.uni-regensburg.de wrote: > Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. Can you please post the original wav? I am not sure what Audacity means by a logarithmisch sweep. Is that a fixed number of Hertz per

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

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)?

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

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