similar to: flac conversion

On 5 June 2016 at 21:34, Frank Esposito <fpesposito at gmail.com> wrote: > Hello -- > > I have a flac file encode 92/24bit -- what tools are available to > convert these files to wave 44.1/16 bit so then I can convert to > mp3? Thanks > > -- > Frank Esposito Don't you mean 96/24? Also, which platform (*nix, Mac, Windows)? Off the top of my head: Audacity,

I was getting a glicthy checkerboard mess at the cursor in some programs (konsol+) with # cat /proc/cmdline BOOT_IMAGE=/boot/vmlinuz-4.9.0-2-amd64 root=UUID=4107d1db-62a3-427c-8515-c954a103ce25 ro text quiet changed the command line in a blind attempt to: $ cat /proc/cmdline BOOT_IMAGE=/boot/vmlinuz-4.9.0-2-amd64 root=UUID=4107d1db-62a3-427c-8515-c954a103ce25 ro nouveau.config=NvForcePost=1

> Regarding dithering, I am not aware of many programs that do that without > very specific user selection. Any user savvy enough to turn on dithering > would hopefully be paying attention well enough to avoid promoting 16-bit to > 24-bit without noticing their mistake. I suspect that this is sometimes done dishonestly in order to sell hi-fi enthusiasts DVD-A's made from

Thanks for the replies! My first thought was that the file had low levels (before he sent me the file), but that's definitely not the case with this file. There are many peaks that reach 0dBFS. He sent me the original wav this morning and I loaded it into Wave Editor on OS X. I dithered to 16 bit using MBIT+ (high/ultra setting) and saved the 16 bit file. I did nothing else (no

Audacity results: It actually did save the file correctly, and the extracted WAV itself is 5.14 GBs! Problem is, Windows Media Player says the track is only 1 hour and 47 minutes long, which is incorrect (as I said before, it's an 8-hour track, meant for 8-hours of sleep)... So I played it in Quicktime, iTunes, and Nero Player just to be sure. Both QuickTime and iTunes only played 1 hour

Nicholas is probably right about noise. Another factor would simply be the amplitude of the resulting file. A 275 MB 24-bit file which compresses to 110 MB is probably not very loud. I assume that the average level is somewhat low, with few if any peaks that reach 0 dBFS. FLAC is very good at compressing audio that is not loud. In fact, the quieter the recording, the smaller the

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

In this topic on Hydrogen Audio(http://www.hydrogenaud.io/forums/index.php?showtopic=105586) someone asked a question about the sample rates that FLAC supports for ReplayGain. The outcome was that the current documentation of MetaFLAC is outdated since Commit http://git.xiph.org/?p=flac.git;a=commit;h=0554a4aee6966bc5b251364753ef85de72dfab19 because as of 1.3.0 FLAC supports Replaygain with many

The problem is really as I wrote: 1. Metaflac is no option for me, I use libFLAC.dll 2. There is no way (at least how I read the code) to avoid saving comment with libFLAC; I would appreciate an extra option to avoid it, which can default to old behavior if compatibility is important. 3. I have a high speed application, where re-initializing an encoder is really significant. On corner cases it

Oops, I was a bit sloppy in my organization, and also sloppy on a few facts. I did not intend to imply that A-52 was lossless - I was merely using it as an example of an open specification format that cannot be implemented and sold without being licensed. I'm glad that we are finally getting lossless formats accepted in large public standards. The liba52 library may not be legal. I

Hi Brian. I also agree with you on these points you mention. If you guys are familiar on how the piracy groups work on the internet, you are aware that they have "releases" with their names on it. In the piracy "scene", some groups are competing on getting the first release out, and could only be beaten by another group releasing another higher quality release. Some groups (or

Hello Tony, Samba isn't the 'culprit', just the media used. Samba does NOT insert anything into the files that you copy over or have an application create on a samba share. The issue is that you are using a PC program to check out/modify the files, and pc os'es (Win/dos, i mean) use crlf as a ascii line terminator. Unix uses a lf only. So when you view the files with a pc

I wasn’t suggesting that you run metaflac, but that you examine its source to see how it creates new FLAC files without the Vorbis comment. As far as I know, metaflac uses the standard libFLAC and creates files without the Vorbis overhead. My quick review of the source seemed to indicate that calling FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT) will create the comment, but I

First off, I'm not an expert in this field, but I thought I could probably answer a few of your questions. If anyone else wants to give a more complete explanation, please do. Brian Willoughby wrote: > Hi folks, > > I'm trying to get a handle on the many lossless audio formats from > the perspective of a software developer. I love the FLAC format > because it is open

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

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

On Jul 18, 2015, at 10:46 AM, Martijn van Beurden <mvanb1 at gmail.com> wrote: > Op 16-07-15 om 07:50 schreef Brian Willoughby: >> On Jul 14, 2015, at 8:18 AM, Declan Kelly <flac-dev at groov.ie> wrote: >>> Can anyone on the list (possibly someone who works for MSFT) get this >>> fixed before Win10 is released? >> What size differences are we talking

On Nov 16, 2011, at 14:11, Declan Kelly wrote: > On Wed, Nov 16, 2011 at 05:41:21AM -0800, avuton at gmail.com wrote: >> Hate to be Capt. obvious here, but there's a lot of development going >> on here that should be encouraged. If the FLAC project isn't going to >> open up, it would make a lot of sense for someone to take over >> maintenance on a github account

I for one am not worried about getting mp3 encoded stuff in my flac files, but I want to respond about "legitimate" OggFLAC. OggFLAC as a format for files, I agree, used by no one. However, I don't know of any other open source way to stream lossless audio. Maybe I did not look hard enough. Certainly nothing I can think of that would hope to be compatible with standard media player

Hello, I would like to sample a population but the result needs to be reproducible, using 'runif' or 'sample' is the good way to do it but I can't manage to make the results reproducible even with the 'set.seed' function. My aim is that th call to 'sample(1:100,10)' gives always the same result, how can I do that? Thanks! Eric Esposito