similar to: Fw: Mono vs Stereo

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,

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

-----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

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, I would like to create an output file with mono audio. How to do it ? If use a video with mono as input, the result will be a video with stereo. Is there a way to use the --downmix from oggenc ? Regards -- ZikZak

I am considering using OGG for sending mono files across the net. OGG sounds great and I like the philosophy. But when I compress a mono file using OGGDROP or OGGENC Beta4, I can only get compression ratios of about 3:1. If I take the original mono file and convert it to stereo, then encode it, I can get a compression ratio of about 10:1. So the final encoded file for stereo is actually smaller

hello how can I set oggenc to encode cds used with abcde to encode ripped tracks in mono? thanks hank <p>--- >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 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe

Links for all the 8 audio files given in the latest post in the Audiobook-Share section. -- With warm regards P K Kothari http://audiobook-share.blogspot.com http://career-ebook.blogspot.com http://collection-share.blogspot.com http://ebook-share.blogspot.com http://management-article.blogspot.com http://finance-personal.blogspot.com http://powerpoint-presentation.blogspot.com

Hi there ! I checked Vorbis' performace for mono files at approx. 64-80 kbps (it does a good job) and I'm wondering why the current OggEnc still uses Point-Stereo (>10kHz) for -q4 and -q5 I know, we're usually unable to percieve those phase correlations above 10 kHz, But a Dolby Prologic Decoder isn't. I think a future version of OggEnc which is able to use a user-selectable

Looking at the Opus comparison page[1], I can't figure out whether the Opus/AAC/Vorbis/MP3 lines are meant to imply a mono or stereo encoding. Could someone please update the caption to clarify this? The single dot for G.711 is clearly mono, but for stereo music, are the codecs at the top meant to converge near 128 kbps, or 256 kbps? [1] http://opus-codec.org/comparison/

Crew, I've run into something a little odd with the vorbis encoder and would like some input. I use SpacialAudio's SAM3 Broadcaster and am having problems with the ogg encoder when I create a mono output stream. It seems as if the signal level of the left and right sides are being combined before the encoding process, leaving the sound muddy and overdriven. As such, if I reduce

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

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

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary to

Hello, Do you know of any Windows software that is able to record in mono from the sound card's line input and at the same time convert to *.ogg? Ultimately, one without ogg conversion would be OK too provided it can record mono wave files. Important: the software should be able to record WITHOUT first creating an empty wave file with a set length (Adaptec's Spin Doctor versus Goldwave).

Hi, Does Celt (in particular version 0.7.1) exploits correlation between the two channels in stereo mode? In practice, is it possible to use the two channels as two mono signals without affecting quality? Thanks Best Regards Riccardo Riccardo Micci Senior DSP Engineer, Wireless Group Cambridge Consultants Science Park, Milton Road Cambridge, CB4 0DW, England Switchboard: +44 (0)1223

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Hi, If I want to encode two audio channels that are not related, audio from two different sources not stereo, is it best to create two instances of a single channel encoder or use one encoder with two channels? thank you, Paul.

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