similar to: More than 8 channels possible?

On Jan 17, 2013, at 21:41, Ralph Giles wrote: > On 13-01-17 7:26 PM, Brian Willoughby wrote: >> I vote for documenting the --channel-map option in the --help > > Do you ever use --channel-map yourself, or recommend it to clients? Professional surround mastering is delivered on very specific media, and FLAC is not an option (to my knowledge). I use FLAC for archival of original

I vote for documenting the --channel-map option in the --help I don't like the idea of rejecting a multichannel file merely for mapping, so there should be a documented option plus an error message pointing to the option. This should compare to the WAVE and AIFF errors where the utility suggests to the user how to get the file converted safely. Brian Willoughby Sound Consulting On

Are there any recording applications which create files with more than 8 channels? For that matter, are there any recording applications which create files with more than 2 channels? I do a great deal of live multi-track recording. Every tool I use creates mono recordings. At most, I combine microphone pairs or stereo electronics channels into stereo files after recording. Then, when

Per https://xiph.org/flac/faq.html#general__channels currently FLAC supports 1 through 8 channels. This is fine for 5.1 surround sound (6 channels). But some of the newer surround sound systems can handle many more channels. Is it still convenient to have all of the audio channels in one file for, say, a 16 channel audio track? I don't know much about the history of FLAC. How did this

Cool, thanks for all the great comments. I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

Hi, I have posted this three times to the flac-dev, vorbis-dev, and ogg-dev mailing lists. I wanted to see what code there was currently to support Ambisonics. So I downloaded the code from the xiph download page for libogg-1.1.3, libvorbis-1.1.2, vorbis-tools-1.1.1 and flac-1.1.4, but wasn't able to find anything. If it exists then I missed it, so could somebody please point me to it.

Hi. It has been some time since I posted to the list, and may be some time in the future until I have an oppurtunity to revisit FLAC. So, as I have been quite satisfied yet always mindful with FLAC, I simply wanted to encourage any potential cumulative bugfix release... and or repository tag/update as such... should it be warranted. It would not go unappreciated by the base users, myself included.

I'm new to the mailing list but am interested in picking up a thread from earlier in the month but which I thought had become confusing so I am starting again. I should admit from the beginning that I am a colleague of Alex Brims who started the original thread. The thread in question related to a wav file with an extra two bytes at the end causing a partial sample error in the reference

Hi, are there any plans for FLAC to implement spatial audio? Or maybe Atmos on FLAC? renatas -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/flac/attachments/20170810/57a79181/attachment.html>

Hi all, In commit 3eb4094b859 I think I have fixed all the cast-align warnings. I have tested this in amd64/Linux (little endian) and powerpc64/Linux (big endian) and it passed all tests (including the new MD5 tests). I also did a little performance testing on amd64/Linux with a one hour long stereo WAV file and could not find any mesasurable difference between the old and the new code. I

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

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)

dd if=$file ibs=1 count=$(($(stat --printf='%s' $file)-2)) of=$file.new of course if you run this on one of the files that doesn't have the extra 2 bytes you're gonna lose something you didn't want to On 11/1/07, Alex Brims <alex.brims@gmail.com> wrote: > Ok, we actually worked this out - there were 2 extra bytes doing nothing at > the end of the files. Opening

On May 13, 2007, at 05:45, Harry Sack wrote: > If I encode 192 kHz sound @ 24 bit for some days (WAV file) and I > encode it to FLAC, I think you can have a very big file and 1.5 TB > is reached very quickly. > And in the future audio will even get bigger, when used for HD-DVD > en Blu-ray media and 5.1 channels is considered the 'minimum' > setting for surround

Is there a portable flac recorder with line-in? tom_a_sparks Light travels faster then sound, which is why some people appear bright, until you hear them speak

--- Pascal Haakmat <a.haakmat@chello.nl> wrote: > Hello, > > First, thanks a lot for FLAC. > > A few questions. Apologies if this is not the right forum or if these > questions have come up before; the SourceForge archive search didn't > turn up anything. > > 1. Would it be possible for FLAC to support more than 8 channels? > 2. If (1) is not desirable

Hey everyone, I have a few questions about FLAC that aren't answered in the FAQ or the documentation. Maybe someone knows about these details. 1. What exactly is meant by "reference encoder"? 2. Concerning interchannel decorrelation, what is the point of creating separate signals, why not just keep the channels you have? For example, you might create "mid" and

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 *

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

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