similar to: Just to dispel any hopes -- RC3 really low bitrate

Since 1.0rc2 was released i have found that ogg@64k sounds almost like mp3@128k (in 95% of the 3000 files i encoded...) I tested 1.0rc3 and 64k bitrate (quality=0.0) gives less good quality than before. And bad : it takes longer to encode ! (1.7x vs. 2.0x on a P2/500MHz) Effectively 64k ogg files encoded with rc3 are smaller than files encoded with rc2. I hear people debatting on how satisfied

hi, i´m looking for an implementation of really low bitrates (16k per channel or less) with ogg vorbis. ome backround informations: i work on a portable device for streaming over the air with hard storage limits, so i like to reduce the filesize without loosing all quality. any ideas or suggestions? thanks oliver --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

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, 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.

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

Hello developers, I have a question about supported bitrates for SPEEX. Regardless of whether I specify a 10kbps or a 8kbps bitrate, the encoded .spx file is the same size and have the same PESQ value. This makes me wonder if SPEEX supports arbitrary bitrates, or only a certain set. A few more details about my configuration: I am using SPEEX encoder version 1.1.12 I am encoding a narrowband

Hello, We're streaming radio programs at both 128kbsp and 32kbps, but only archiving the 128kbps stream to save storage space. I'd like to give users a similar choice of bitrates when they request an archived stream (served through icecast's /file/ functionality). Is there a way to change the bitrate on the fly, or do I really need to save archive both bitrates? Thanks for the

dear vorbis developers, it would be very useful, if oggenc would support constant low-bandwith bitrates for testing. something from 20kbps to 44kbps, so anyone could test streaming over modem and isdn. audio quality doesn't matter. mörk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

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

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

It is commonly said here that if I want to make AM radio-quality stuff at very low bitrates, a good way is to downsample. I downsampled a song to 11025Hz mono and encoded with -q 0, the result is about 18kbps and is at least radio quality. The downsampler I used is from Edinburgh speech tools, named ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis/attachments/20041014/25f999a2/attachment.htm -------------- next part -------------- ? dear All, Iam working on OGG VORBIS TREMOR code to port and optimize on ARM processor. Can any one make me clear regarding the relation ship between this 1. What is the relation ship between Quality factor and Bitrates?

Hi, I'm using Theora for live video chat. My problem is, at higher bitrates the bandwidth peaks caused by key frames are so high that most key frames get dropped in UDP traffic. The peaks are so strong that I still get many packet loss artifacts when sending with 1Mbit/s bitrate on a 2Mbit/s connection. I measured delta frames to be around 12kbyte and key frames up to 100kbyte in

I have got ices running with quality -1 (and on my roommates crappy computer speakers it sounds damn near the real thing - better than FM radio anyway)... I'm just wondering if it's posible to configure it with a maximum bitrate, or a nominal bitrate. i.e. to use managed bitrates instead of quality (I know it's a no no, but this is streaming!). Mark <p>--- >8 ---- List

Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... Cheers, Simone.

Hey, Maybe you guys remember us, I'm from the Neuros forums. I've got to say, having a player that supports Ogg Vorbis has really turned things around for everybody. We're still struggling with higher bitrate Vorbis streams, but that is only because of our terribly underpowered DSP. Anyway, to the point of my email. Recently the discussion of bitrate peeling has been floating

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

Hello- I recently downloaded Verbs and started playing with the encoder. The quality was great when I encoded a few samples at medium and high bitrates, but when trying to encode the same clips at 16 kbps, artifacts are audible during playback- short bursts of screeching noise, not just distortion. I'm using oggenc 1.0 beta 4 to encode, sonique to play back on Windows NT. I've also

If I understand correctly, Ogg Vorbis is a lossless format. But then why does it support bitrates? Ogg Vorbis does not delete any information. When I encode a 128 kbit MP3 file to 32 kbit OggVorbis, the resulting file has a bitrate of about 110 kbit. And when I encode it to 320 kbit, the file will be bigger, but the quality won't improve because of the quality of the original. So what purpose

El mié, 23-08-2017 a las 14:42 +0200, Marvin Scholz escribió: > > On 23 Aug 2017, at 14:40, José Luis Artuch wrote: > > > El mié, 23-08-2017 a las 14:13 +0200, Marvin Scholz escribió: > > > > > > On 23 Aug 2017, at 14:08, José Luis Artuch wrote: > > > > > > > Hello, good morning ! > > > > Running Icecast2 and Ices2 with live