similar to: Hz vs bitrate?

When I encode a monophonic wav file, I would expect the resulting ogg file to be at about half the bit rate specified on the command line, as stated in the "oggenc -h" help text: "The 6 modes are approximately 112, 128, 160, 192, 256, and 350 kbps (for stereo 44.1kHz input. Halve these numbers for mono input).". This doesn't seem to be happening, though. I took a 16-bit,

Hi, <p>apparently, I didn't delete my test samples ... I found them. :) Here's a small table of what nominal bitrate fits to which quality at what sample rate. Quality levels inbetween -1 and 0 are possible, but I didn't test them. Nominal bitrates of those quality settings will be somewhere between those of -1 and 0 and therefore can be guessed. q/channels || -1/mono |

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

I must be living under a rock, im on the icecast mailing list, but yet do not quite understand exactly what Ogg/Vorbis is.. What information I think I know is this.. People behind proxy's like it, and its not exactly Mp3... hehe, anyone care to help clarify? Thanks Scott W. --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To

Hi, I was checking for bit rates support above 500kbps in the libvorbis code, so I tried to encode ogg files with bitrate 576kbps but the encoder OggdropXpd gives maximum bit rate of only 500kbps. Does the Vorbis audio support bit rate 576 kbps? Thanks, Emima -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I am a new user of icecast, and I was wondering, wich source software, gives the option of compressing to low bitrates an ogg stream (lower than 48 kbps). Thanks: CC

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

Hi, I found this list of Speex bitrates in the mail archive. http://www.xiph.org/archives/speex-dev/200306/0004.html Can somebody confirm that this list is correct? I am wondering about the following: - On the Speex website it says: "Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." while the bitrates listed here are e.g. 84400 for

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

I am totally new to icecast, and would appreciate a pointer in the right direction. My application is to encode audio (at the linux server) at a low bit rate, say ~16 kbps, for transmission to a decoder at another location. The audio source will be bandlimited (basically voice grade circuit such as you'd find in a telephone application.) The decoder end (connected via a TCP/IP session)

On Wed, 6 Feb 2002, Brendan Cully wrote: > Is the sample rate changing? The decoder is currently very fussy about > having the audio format stay consistent. Yeah, that's what I thought it might be... most of the MP3s I'm playing are 160, but some are 128 and some are 192... I should convert them all to one bitrate, but I'm just too lazy. And I'm only doing this for fun,

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

Happy New Year from the Xiphophorus Team. It took longer than we thought, but I think everyone will agree this is our best release yet. With drastic quality improvements and new bitrate management features, this release brings us one step closer to 1.0. Along with all the lovely VBR modes you are used to, you now have millions more. oggenc's quality settings are 0-10 in increments of

Hello, I've been postponing some of my encoding for when joint stereo gets implemented. The reason I've been doing this, is that I am under the impression this is the largest step in the quality/bitrate ratio that's left. Now I'm wondering if I am correct in thinking this. lame's documentation seems to imply it doesn't make much of a difference at higher than 128kbps (I

Hi, We would like to use the OPUS codec @ 16 kHz sampling rate and max 32 kbps. What about implementing an OPUS coder and decoder in an FPGA? Has this been done? Would either coder or decoder more suitable for FPGA implementation? Best regards Fredrik Bonde -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello all! After having read http://www.xiph.org/ogg/vorbis/listen.html and listened to the examples on that very educational page, I decided to augment the info. This example is more simple, it involves the comparision between two streams of the same radio station, FranceInter (a station of Radio France in Paris). The ogg stream is running at around 30 kbps/11 kHz in stereo. The WMA stream

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

On Wed, 6 Feb 2002, Brendan Cully wrote: > Strange. The different bitrates shouldn't be a problem, as long as the > music beneath is 44.1kHz stereo (which I would expect at 128+ kbps). I > reencode from 128 to 192 regularly, and ices never hiccups at all. What > version of LAME did you link against? 3.89 beta 1. And, yes, everything's at 44100. Like I said, it's no big