similar to: Vorbis rate and bitrate.

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Hi all, I attached a driver for MiniBox openUPS device ( http://www.mini-box.com/OpenUPS) and a dump of the hid usages. I had the possibility to make a few adjustments to the device firmware for HID usages, and although I haven't managed to produce a good structure many issues from previous firmware were at least fixed and new information added. ATM the driver only shows pertaining

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

Hallo oggs, i googled it but still can't find any answer: how can i set default audio track? I transcode my dvds to ogg, i use gst-launch for it: gst-launch filesrc location=stream.dump ! mpegpsdemux name=demux \ oggmux name=mux ! filesink location=out.ogg \ demux.audio_80 ! a52dec mode=2 ! audioconvert ! audioresample ! audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ taginject

There is a big, big difference between cbr and vbr, btw. The machine in question is a Pentium 3 700 oc'ed to 800, 512MB ram, win nt 5.1 (2k and xp being nt) etc In an earlier version the oddcast dsp ate up so much cpu time when set at say 64kbps (for examples sake) 44.1kHz, stereo that the machine became totally unusable and the winamp task had to be (slowly) ended with task manager. Now,

On Tue, Sep 06, 2005 at 09:56:06PM -0500, Jay Krivanek wrote: > Yea, don't get me wrong, ogg is definitely an awesome format but aacplus v2 > is currently the leader in high fidelity quality at low bitrates. 24kbps > aacplus even kicks ass in my view. Is that 24kbps AACPLUS a full 44.1kHz stereo signal with frequencies preserved up to at least 15kHz? (Just curious, not

Mark Casey wrote: > I think when the real-time cbr encoding issues are sorted > Vorbis (this is my experience with oddcast dsp, winamp and > icecast 2 win32, big cpu hog) will start gaining more ground, I can't use the managed bitrate option in OddCast on my K6II-500 CPU! (Overclocked to 525, WinXP Pro,192MB RAM). I would like to. Normal VBR uses about 90% CPU. This seems too

Hi, I'm trying to generate a skeleton driver with usbhid-ups and gen-usbhid-subdriver but the output seems to be incorrect. Same behavior with stable 2.6.5, so I guess there is something wrong with data coming out of device. I attached the output of usbhid-ups and gen-usbhid-subdriver from the commands: drivers/usbhid-ups -DD -u root -x explore -x vendorid=04d8 -a openups >&

Yes, it works now. Thank you very much. -----Original Message----- From: Chris Wolf [mailto:cwolf@starclass.com] Sent: 05 September 2001 10:12 To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Problems trying to run the examples in windows vorbis sdk On 9/4/2001 at 6:00 PM Pablos-Sanchez, Rolando wrote: >I have just downloaded the vorbis sdk for windows, for researching >purposes. >

As the subject implies, my question is: is it possible to use the -b (or -M) flag at non-44K sample rates? I'm working with an application that is trying to optimize for very small audio filesize. I found that downsampling to 11K and then using q0 gives high compression, but won't seem to drop below 64kbps or so. It seems like the combination of downsampling, then reducing to 30kps

Tried both the binary and compiling the src. Both leaking. It the setting suggested in "INSTALL.lame" didn't work and so I suspected the compiliation settings so I tried various settings. The compiler version is the one coming with RedHat 7.2 gcc3-3.0.1-3. Kristjan <p><p><p><p>-----Original Message----- From: owner-icecast@xiph.org

Hi: I've been experimenting with streaming vorbis across my network from the windows box to an icecast2 server on my linux machine (the server I usually use is having some network issues at present). I've tested both ostream 0.7.1 and oddcast beta26. It seems that neither can encode on the fly on my P2-266 at 44100 fast enough to be able to stream it. Ostream looked the more promising

Hi, Maybe a bit off topic for this list, bt anyway. I have received several feature requests for DarkIce to support downsampling of the audio input before passing it to lame or ogg vorbis. For example the audio read from the soundcard would be 44.1kHz, and lame would get it at 22.05kHz. I figure two ways of doing this: 1. For lame, one can specify the input and the desired mp3 sampling rate,

Hi, A short mail to let you know that the MSVC project files aren't OK compared to the Makefiles. Indeed they include the vorbisenc part (vorbisenc.c file, and vorbis_encode_* function names in the .def file) inside the vorbis library, and they also point to a non-existant file (bitbuffer.h) Here is a patch to fix this, in case you are interested (patch made on the win32 directory using

I have just downloaded the vorbis sdk for windows, for researching purposes. I have compiled the example 'encoder_example.c' just to start, and it compiles, but I got a error in the following line just at the beginning: vorbis_encode_init(&vi,2,44100, -1, 128000, -1); The error is more or less: The instruction at '0xaddresss' referenced memory at '0xaddress'. The

Hello, First of all, I am a complete newblet to anything dealing with audio (a lot of the vocabulary is still rather mysterious to me). I also don't often program in C so please don't throw too many stones if my workflow is less than ideal. :) I would like to port the vorbis encoder to JavaScript. Instead of porting everything by hand from ground up, which would take who-knows-how-long,

Chris, re the DLL additions: In a word, *no*. That much Win32 specific code does not belong in a lib that's pure math and totally platform agnostic. It may be appropriate, assuming it's necessary, to add this in the win32 specific SDKs, but it has no business in the core libvorbis. Please move it out to a more appropriate place as soon as you possibly can. Secondly: I'd like to

You helped me a few weeks ago setup DarkIce. Well I had nothing but problems doing it on RH62 so... here I am again but this time armed with RH71. I followed your instructions for installing lame and ogg vorbis and it's all fine. When I tr to install DarkIce though I get the following error. # make make all-recursive make[1]: Entering directory `/usr/src/download/darkice-0.6' Making

Hi! I'm very impressed with the progress Ogg Vorbis has made, ever since getting that T-shirt at LinuxWorld Expo :-) Using oggenc, I've run into a problem, though: "Mode initialisation failed: invalid parameters for bitrate" This happens when 2 conditions are true: 1) the input file has a sampling rate less than 44100, and 2) the -M parameter is used to specify a maximum