similar to: [Flac-users] Should I use 11 Khz or 22 Khz

Hi. I am a student with limited funds and storage space on my linux box. So I am trying to decide what audio compression scheme to use for my collection of audio tapes (recordings of one speaker, preaching in an church auditorium, recorded with low-grade mics and tape decks) converted to digital audio (16bit, stereo, WAV, 44100Hz). Is FLAC, LPAC, or a standard data compression format better for

Hi Peter, Have you considered resampling the raw 44.1kHz stereo source files using a program such as http://audacity.sourceforge.net/, say to 16kHz mono or 32kHz mono, and then using the wideband or ultrawideband speex modes to encode the result? Alternatively, if you want to programmatically do the resampling yourself, you could try the new resampling API in the svn head of speex, or the GPL

Hello, Is there anyway to remove the bootup kernel text with pxelinux? (blank screen or logo image till X is loaded) I tried using LPP (linux progress patch), But it only seems to work with lilo only. Im trying to creat a kiosk type inviorment I dont want users to see the kernel load.. Any help will be greatly appreciated, thank you.. Danny

I am struggling with the simplest thing, I think! I am starting with a fresh config file from the 20020605 wine distro and trying to get the ARS Remedy User win32 based software running. Part way through the install it tell me I need to install "Winsock 2". By this I am assuming it is looking for the 32-bit winsock Dll's, wsock32.dll * ws2_32.dll. I have tried many variations of

I have RedHat 7.2 Workstation that is using Samba to attach to a RedHat 6.0 Server share. I have inserted the following entry in /etc/rc.d/rc.local on the workstation: mount -t smbfs //office/Shared /mnt/shared -o username=xxxx,password=xxxx Access is fine, but I am unable to write to the share. I have Chmod-ed, Chown-ed and Chgrp-ed the share so that it is completely available with no

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hello, I'm new to KDE 3 and Wine. Now, what I like is that now wine creates desktop icons on demand. The only problem i got is that i want to run native dll with the application, because otherwise the application won't work.I can do it using the bash, but i woud like to be a bit more lazy :) Now, how can I do that? I have tried simply adding the --managed -dll command in the execute

Hi, I am quite new using Red Hat, I would like to know how to uncompress the tar.gz file to be able to install it...and after uncompressed how to proceed to install and configured...I'd appreciate your help...thanks.... -- Your favorite stores, helpful shopping tools and great gift ideas. Experience the convenience of buying online with Shop@Netscape! http://shopnow.netscape.com/

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

Audacity results: It actually did save the file correctly, and the extracted WAV itself is 5.14 GBs! Problem is, Windows Media Player says the track is only 1 hour and 47 minutes long, which is incorrect (as I said before, it's an 8-hour track, meant for 8-hours of sleep)... So I played it in Quicktime, iTunes, and Nero Player just to be sure. Both QuickTime and iTunes only played 1 hour

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

Ok, so here's the deal... I've been having a lot of trouble sleeping, so I've done some research and discovered this program called "Hemi-Sync". I decided to torrent it from the Pirate Bay to try it out before spending my money. What I torrented worked incredibly, so bought it, then torrented another Hemi-Sync program: Lucid Dreaming. It's a DVD-Audio program. What I

Le ven 10/01/2003 à 14:39, John Hayes a écrit : > I've been playing with speex for use in a VoIP application between PC's. One > thing I've found (correlating to the documentation) it that speex runs much > faster and produced much better output when it's fed a 32kHz signal instead > of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate > and feed

--- boink <boink@tetter.xs4all.nl> wrote: > The ogg stream is running at around 30 kbps/11 kHz in stereo. The > WMA > stream (provided by yacast.fr) is running at 16 kbps/22 kHz in > mono. Is stereo vs. mono, and 22khz vs. 11khz even a fair comparison? 22khz vs. 11khz: 22khz, by definition, gives the 22khz stream much more frequency range and *opportunity* to sound clearer.

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html

At present my stream is at 11.025 kHz and I want it to be at 44.1 kHz. Input is coming from line-in on my sound blaster card under linux (RH 9) using the sb driver. I presume that it is icecast that sets the sample rate on the dsp in the card, though when I change the settings in icecast.xml and darkice.cfg as show below the stream becomes choppy; or rather the sampling doesn't seem to

Hi, I'd like to know the reasons why CELT supports only signals with sampling frequency in the range of 32-96 kHz. In effect, it can clearly outperform speex at high bitrates, and has potential to be used in high quality voice communications even for 11, 16 and 22 kHz speech signals. It could also compete with SILK codec (to be soon released by Skype). See this page for more specifications

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Moritz Grimm (gtgbr@gmx.net) wrote : > Michael Smith wrote: > > >For the sake of convenience, it's nice that Vorbis' VBR supports 48kHz > > >at decent quality already, but to get back to the start of the > > >discussion, not all people have the equipment to play it back. To bring > > >highest possible satisfaction to the audience, 44.1kHz still is