similar to: Low sample rates / bit rates

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 everyone. I did a rough recording of an instrumental (electric piano sound) & e-mailed it to a friend in Vorbis 11025 Hz / mono. I was seeking a bitrate in the range 8-16k/sec. The song is 2:55 in length. q=-1 happily achieved a 12.6k/sec bitrate. All file sizes I mention in this are for files without informational tags. And I hope this isn't interpreted as trying to plug my music or

Hai! I'm from Hungary, and my English is not so good. Excuse me! I'm using Linux Debian woody, with kernel 2.4.18 Now I try to encode my music files, that are in .wav format to .ogg format. I used to use the lame program to encode my music files to .mp3 format. I try it now on a Windows NT 4, and this 2 version: Lame version 3.92 MMX OggEnc v1.0 (libvorbis 1.0) I have a question, why is

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

I mentioned many months back that I intend to switch to Ogg/Vorbis from MP3/LAME once the final version is out but I need to have an adjustable lowpass filter option similar to LAME's. If this isn't done yet, could you please add it to your list for OggEnc. Specifically I need to cut all frequencies above 17khz. Thanks, Ross. --- >8 ---- List archives: http://www.xiph.org/archives/

Hello everyone, I'm a student at the Universtiy of Delaware, and will be soon starting some research on the effects of lossy audio compression on speech sounds. I will be preforming test with both mp3 and vorbis. First of all, if I use the '--ogg' switch to lame, does lame use GPSYCHO to encode the wave, or some other psychoacoustic model (perhaps one designed for

Hi, with the Lame MP3 encoder I reduce the amplitude of the input signal prior to encoding with the --scale option to prevent clipping that can be introduced due to resampling errors. Is Vorbis prone to this sort of clipping? <p>Headless --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a

Check this clip (it's small, 373kb) http://www.geocities.com/jdxss/udialwav.zip It left oggenc, lame and MP+ encoders choking in dust. -- Vorbis Xtreme | http://solair.eunet.yu/~aldov/ Ogg Vorbis is the free, open source alternative to MP3 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a

Don't know if anybody is still missing the lame oggenc features for resampling, lowpass/highpass filters etc, but I wrote a little script that uses sox to do all the stuff I need to real-time encode oggenc from the radio, or any input device. #!/bin/bash DATE=`date '+%m-%d-%Y-(%H.%M)'` DESTIN=/video/music/perftoday export DATE=$DATE'-PerformanceToday.ogg' sox -V -r 44100 -c

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

Hello, just one question: is it possible to implement a "quality switch" to Ogg Vorbis like the -V option of the lame encoder? It's the only thing I'm missing. Everything else is nearly perfect! Sebastian --- >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

What I want to do is put up an icecast 2 server streaming vorbis, but the original live stream is mp3. It seems that I want to do something like this (with all the options left out for clarity): mpg123 | lame | something where the "something" is some command that sends the bits to icecast like ices. Does such a thing exist for ogg/vorbis? Is there a better way to do this?

I'm trying to send out multiple streams. My encoder is a PII 233, so I have to conserve CPU cycles. I encoded my mp3s as 96kb/s with a 44.1 sampling rate. I send my 96k stream without reencoding and it sounds great. I also sned a 24k stream that is being reencoded from the 96k mp3s. It works pretty good, the load on my linux box is about 0.5 on average. However, the sampling rate is 8k.

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

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,

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 downloaded all the Beta 4 libraries/tools yesterday and compiled them - everything went fine. ogg123 works fine - it plays files off the website. So does the XMMS plugin (I just built XMMS and had it build the vorbis plugin that it came with) oggenc seems to work fine, in that all of its output looks normal: [ttyp3@dalek:tear-require-0.2.0/CDDB_get-1.4 @00:39] oggenc track03.cdda.wav

Hello List, archives don't look very busy ... I'm interested in putting art into Ogg Vorbis files. This is for a project I'm working on with small local bands. I'm attempting to help get them more exposure by (with permission) distributing their music, especially recorded live from shows they do where they are allowed to record it. We are going with Ogg Vorbis frankly because

Did something change with exception handling recently? A bunch of lit bots are showing slower compile times for many tests. Ciao, Duncan. On 20/06/12 07:53, llvm-testresults at cs.uiuc.edu wrote: > > lab-mini-03__O0-g__clang_DEV__x86_64 test results > <http://llvm.org/perf/db_default/v4/nts/1283?compare_to=1278&baseline=999> > > Run Order Start Time Duration >

Hi Brendan: The diff patched, configured and compiled just greate. A bit of a false start because I chose a bitrate lame didn't like. Once I figured that out however things went fine. You might want to log the point lame dies on something which would help others know where to look for lame errors. On another note, I see ices still only accepts /tmp for the ices:Basedirectory tag. I hope