similar to: encoder configuration

Hi, How can I force encoder to create stereo? OPUS_SER_FORCE_CHANNELS only works for mono How can I force encoder to encode at a specific bandwidth? OPUS_SER_BANDWIDTH only works for FB (1105), and the encoder Seems to encode at the input samples rate, no rate conversion is done Thanks, Michal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Echoing Hiroka, who is likely on the right track. Simply telling the encoder 'one channel' and then passing stereo data will not convert the stereo data to mono. The encoder encodes whatever it is given... Monty On Tue, Jan 3, 2017 at 6:51 PM, IHARA Hiroka <ihara_h at live.jp> wrote: > Hello Ross, > > The attachment appears at least to me as a valid monaural Vorbis

Is there a list somewhere of "standard" encoding rates? I know, for example, CDs are encoded at 44100, as is a lot of digital sound, but I've seen programs that specify different levels of quality (like radio, phone, tape, CD) and I'd like to know if there are some encoding rates that are accepted as standardized for recording at different levels of quality. If so, is there

I’m using a Windows development component which uses vorbis.dll, ogg.dll, vorbisenc.dll for encoding an Ogg Vorbis file. It's all working well except for one user occasionally has a 1 hour file appear as 2 hours and it plays at half speed. It is being converted from stereo to mono before feeding the encoder with a channels=1 configuration. Here is an example file which will be available for

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

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

On Sat, Nov 17, 2001 at 06:57:46PM +0100, Maroy Akos wrote: > On Fri, 16 Nov 2001, Samuel Hathaway wrote: > > > I agree! Also, something I've been looking for is a way to pull sound from > > the dsp device at 44kHz and then downsample it to 22kHz for one of my two > > streams. Ideas? > > DarkIce does this already. What it doesn't do, is to have mono and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello For some reasons, I need to encode in mono a stereo wave file, as with the --downmix option from oggenc. I searched in speexenc man page and in the on line documentation, unsuccessfully. Of course I could first use something like sox to down mix my file, but it would be more practical to have an option directly in speexenc. - -- RIQUER

On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote: > AM radio is lower quality (mono) but I don't know > what the digital equivalent would be. Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad)

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

I'm using Opus for a networked ham radio communications receiver. Kudos on the whole project; it sounds excellent even at low bit rates and the API was surprisingly simple and easy to use. And god knows the world needs a really good, general purpose, non-proprietary codec. One question, though. Depending on the mode my receiver can generate mono or stereo. Most operating modes are mono, but a

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

I have some OGG files which would benefit from the following simple corrections: a) removing left channel (which is empty) and changing right channel to the mono channel (mono sound stereo-recorded by mistake) b) volume change Of course I can open the files in audacity and do those corrections but that would require re-encoding. And I am afraid that it will result in additional loss of the

My apologies for the confusion, I think I have the mapping layout correct in this patch. Cheers! On Tue, Apr 25, 2017 at 10:07 AM Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 25/04/17 10:12 AM, Drew Allen wrote: > > We assume that the input file is ordered first by ACN ambisonic channels > > followed by a (possible) stereo track, and we want to swap the order for

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On Thu, 15 Nov 2001, E wrote: > - streaming at multiple bitrates (where the 128 bit stream is playing > the same thing as the 56 bit) DarkIce supports this feature --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word

On Fri, 16 Nov 2001, Jerome Alet wrote: > one thing that would be nice in DarkIce would be to allow the user to pass > specific reencoding options for each server, e.g. DarkIce could acquire > the audio in stereo and send it to a server in mono and in stereo to > another server, which is AFAICT impossible today. I agree! Also, something I've been looking for is a way to pull

Hello all, I created today a fake stereo piece of music (the left and right channels were completely equal), and encoded it with the all the modes the RC2 encoder supports, and found that: * * dual * waste * waste * * mode * joint * stereo * in * in % * * * stereo * /2 * kbits * of JS *

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html