similar to: A-law and mu-law

Hi Martijn, thank you for your answer. So the problem would be suboptimal compression due to suboptimal assumption about the input signal, right? What I do not understand is how the format of a FLAC format would be affected by supporting A-law and mu-law files as input (and thus output). Despite of suboptimal performance, is it possible to treat 8bit *-law samples as 8bit linear PCM files and

Thank you all for your answers. They were all useful. Il 21/11/2011 07:37, Erik de Castro Lopo ha scritto: > Giulio Paci wrote: > >> thank you for your answer. So the problem would be suboptimal >> compression due to suboptimal assumption about the input signal, right? >> What I do not understand is how the format of a FLAC format would be >> affected by supporting

Because u-law and a-law are non-linear algorithms and the mathematics of FLAC (and AFAIK all lossless encoders) are build for linear PCM. Adding these formats would change the FLAC format altogether, decoders are not made to work with it (they can only output plain PCM) Best option would be converting these files to 16-bit plain WAV and compressing, but I guess files won't be much smaller

On Nov 19, 2011, at 16:42, Giulio Paci wrote: > So the problem would be suboptimal compression due to suboptimal > assumption about the input signal, right? The problem is more that FLAC should not be a collection of code to read every possible file format in existence. That would be a duplication of at least two other audio file format conversion utilities, and quite bug ridden,

Hi, sndfile-convert already converts from all these formats to FLAC, but the flac tool itself has more flac-specific options. Is it possible to use sndfile-convert to provide the input data? In any case Erik is maintaining both libsndfile and libflac, and it's unlikely he'd want to duplicate the code. If anything it'd make more sense to remove code for reading other formats from the

I thought that an uncompressed FLAC file provides better sound quality than a compressed FLAC file. To Erik: How to grab the revision before this one https://git.xiph.org/?p=flac.git;a=commit;h=fc360735ce4d1aa88a94bfccdd3bea5bdd19a8d6? Regards, Maciej ----- Original Message ----- From: "Erik de Castro Lopo" <mle+la at mega-nerd.com> To: <flac-dev at xiph.org> Sent:

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

File size is not important for me. I need audio data from wav file and metadata in one file. How to encode a wav file to an uncompressed flac file? Regards, Maciej ----- Original Message ----- From: "Erik de Castro Lopo" <mle+la at mega-nerd.com> To: <flac-dev at xiph.org> Sent: Monday, December 02, 2013 8:28 AM Subject: [SPAM] Re: [flac-dev] Encode wav to uncompressed

Hmm, what if the FLAC options produce a larger file on output than input? Would -f (force) cause the whole process to fail as soon as the output exceeded the input? Brian On Apr 9, 2007, at 17:01, Josh Coalson wrote: --- Harry Sack <tranzedude@gmail.com> wrote: > is it possible to re-encode an existing FLAC file by using the FLAC > file itself as input to the encoder like

Thanks. sox was exactly what I was looking for. -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu] Sent: Thursday, February 08, 2001 2:06 PM To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format) On Thu, 8 Feb 2001, Giovanni Sanfelici wrote: > I am working on a project

I mean that the first statement [Subset streams must use one of 192/576/1152/2304/4608/256/512/1024/2048/4096 (and 8192/16384 if the sample rate is >48kHz).] published on https://www.xiph.org/flac/documentation_tools_flac.html#flac_options_blocksize page IS NOT EQUAL to second statement [The blocksize bits in the frame header must be 0001-1110. The blocksize must be <=16384; if the sample

> You don't. Audio CDs don't support ReplayGain tags. The only thing > you may do on Audio CDs is to use Volume Normalize techniques > available on certain programs. Normalization of sound is not a good, > clean thing, though. Surely it would be possible to have something apply the ReplayGain to the WAV after decoding and prior to writing them to CD... but I don't know

Linda Walsh <flac at tlinx.org> wrote: ... > That brings up another question then... What is 'hi' (v. low?)... Zero, or close to zero, is low. 12 is high. ... > What's a subset stream? This is explained at: http://flac.sourceforge.net/format.html in the bulleted point beginning, "FLAC specifies a subset of itself as the Subset format." ... > Another

I have just installed CentOS 5.1 on my home server and I am trying to set a mail server. I have diligently followed the instructions on the Wiki - How To on the CentOS website (http://wiki.centos.org/HowTos/postfix). However I cannot send internal email (I haven't yet tried externally). Do you have any suggestions on what to check? To test it I used two normail user: giulio and federica.

"Linda A. Walsh" <flac at tlinx.org> wrote: > what does the exhaustive model search do? I am not an expert on the guts of FLAC, but there is some information about this at: http://flac.sourceforge.net/documentation_format_overview.html and scroll down to "MODELING". > Does it try all of the functions listed under "-A" to find the 'best', >

Hi, I meant that it seems problematic to write an updated file to itself, because the operation could erase the data before it is read. Unless you read the entire file first, or write to a temporary file which is then renamed. I deal with FLAC files that are over a gigabyte, meaning that this could easily exceed normal buffers. Just curious. I guess I could go test it myself... or

Hello everybody. I know, I know, another post about Apache 2, but I have looked a lot on the internet and I haven''t found the answer to my problem. I have a Linux Fedora Core 5 box, with Apache 2.2.2 (compiled and installed), Ruby 1.8.4 (compiled and installed), RubyGems 0.8.11 (compiled and installed) and Rails... I don''t know what version as this one I haven installed

Hello everybody, I was able to set up Ruby 1.8.4 on Rails, using Apache 2.2.2 and MySql 5.0.22 and if you go to http://spagnolo.troccoli.it you have the nice "Welcome abroad" page. I have created a table (modes) with mysql CREATE TABLE `modes` ( `id` INTEGER UNSIGNED NOT NULL AUTO_INCREMENT, `description` VARCHAR(45) NOT NULL DEFAULT '''', PRIMARY KEY(`id`) )

Dear all, I cannot figure out how to solve a small problem (well, not for me), surely somebody can help me in few seconds. I have a series of strings in a vector X of the type "xxx", "yyy", "zzz", "IgA", "IgG", "kkk", "IgM", "aaa". I want to substitute every ENTIRE string beginning with "Ig" with

hi, is it possible to re-encode an existing FLAC file by using the FLAC file itself as input to the encoder like this: flac -V --best inputfile.flac OR do you have decode it to WAV first? thx