similar to: constant low-bitrate for streaming test

Hey everyone, Here's one way to get .ogg files down to about ~20kbps, and it's not too hard: 1. Use sox to resample the files down to 11025khz mono. 2. Feed these to OGGLAME (Lame 3.84 Alpha 1 is what I had laying around on my hard drive). OGGENC demands that the files be 44.1khz, but this version of ogglame didn't seem to care what the sample rate was. I'm sure the quality

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,

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

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

I'm sorry if this question has been asked before; I've looked through the archives and haven't seen anything. The problem I'm seeing is that oggenc's VBR encoding doesn't seem to pay attention to any sort of bitrate limitation, either the -m or bitrate_hard_min settings. It isn't that it temporarily dips below the minimum; the average for the whole (in this case,

hello, could someone test the my icecast 2 server? there should be a stream at http://real.radiostudio.org:8000/vorbis, but I'm not able to listen to it. I don't know if the problem in winamp or in icecast or in my streaming application. You could also send a stream to it, it's the default icecast password. I would be very happy if someone could test it :)))))) mörk --- >8 ----

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

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

I am using OggEnc v0.8 (libvorbis rc2) (win32 cli) I have access to two machines Pentium 3 450mhz, 192mb, Windows 2000 (SP2) Athlon 1400mhz, 256mb, Windows 2000 (SP2) When I encode the same wave file on each machine at the same nominal bitrate (96kbps), I get output files with different average bitrates. 107kbps on the Athlon, as opposed to 84kbps on the Pentium 3. Anyone

I am using OggEnc v0.8 (libvorbis rc2) (win32 cli) I have access to two machines Pentium 3 450mhz, 192mb, Windows 2000 (SP2) Athlon 1400mhz, 256mb, Windows 2000 (SP2) When I encode the same wave file on each machine at the same nominal bitrate (96kbps), I get output files with different average bitrates. 107kbps on the Athlon, as opposed to 84kbps on the Pentium 3. Anyone

hi, what makes the stream at http://ogg.radiostudio.org:8007/ostream.ogg sound so strange. I don't complain, I just want to know the reason for this (is this pre-echo?). mörk p.s.: the stream will be online for the next couple of days. --- >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

Hey guys. I think Vorbis is pretty cool, but since the current OggEnc only offers 44.1kHz, it limits what I wanted to use it for. So I've been using Lame to get 16kHz mono Vorbis files. I'm curious about whether Lame does Vorbis encoding the "right" way for non-44.1k stuff, or whether it just encodes as it would for 44.1k & changes the sample rate on the output, but I'm

Hi, Was wondering if any gurus could tell me if it is possible to stream Ogg on very low bandwidth links (for example satellite phone) where the expected bandwidth would almost certainly be less than 19Kbps, and probably more like 10Kbps. I am not after great audio at this bitrate, mono with an audio bandwidth less than 5KHz. I have done some testing (not streaming though) listening to 8Kbps MP3

Hi. I'd like to ask a little about where Ogg stands today when it comes to streaming audio. The VBR has been the main argument in the incompatibility for streaming but has this changed somehow or will it? Is there something else besides quick and large scaled bandwith usage variations and the difficulty of buffering, that could cause problems? CBR could be used but some people say that it

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Important (imho)! I found this site in the internet: http://www.plusv.org/ PlusV is an audio enhancement algorithm similar to SRB of Mp3Pro, but 1) better than SBR 2) Fully open-source It seems to me wise to include PlusV into new versions of Ogg Vorbis. P.S. what about some long-awaited features in Ogg Vorbis? 1) ability to turn off the frequency filter in the encoder (especially for high

I'm a bit confused about VBR with Opus. I see that the default is VBR, but I don't see any way to specify a quality setting. I can set a target bitrate, but that's definitely not what I want; I want a constant quality level, like "-q" in oggenc, and for the encoder to select bitrates based on the desired quality. For example, for playback through earbuds or laptop

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

I am writing a long-run sound recording application using OggVorbis. The user can adjust sound quality parameter to make balance between storage space and sound quality. The input sound is 22050Hz, mono. To maximize the recording capability on a given storage volume, we want the result bitrate as low as possible. I use -q0 parameter and it produces a 32.0kbps stream. It is too high (our