similar to: identical audio but not identical unrecognized chunks

Hi all, Currently I'm looking for users of the --keep-foreign-metadata feature of FLAC. There has been some improvement of this feature in FLAC 1.4.0. Since 2007 there has been a warning in FLAC that --keep-foreign-metadata is a new feature. I think removal of this warning is long overdue, but there are still some issues surrounding it. So, if there are users of this feature on the mailing

Hello, I'm attempting to convert fairly large WAV files (90 - 800 MB each) using flac but the files do not work after the encoding. (The play fine in wav format) Command I'm using: flac --verify -8 file.wav Attempting to run the file with either flac123 or the default player for Ubuntu (Movie Player?) results in the extremely terse messages: Default Player: "An Error Occurred:

Thank you for the reply! I know that my system can play flac files, I've played others I've managed to convert using both of those programs. I'm only running into difficulty when it comes to these large WAV files. By "Does not work" I mean that they do not play, and instead I receive the errors I mentioned in my original post. I wasn't actually intending to use

I need uncorrupted metadata, as I use data in Base64 encoding in a long text tag for my music synchronized light show. Implemented now in .mp3 ID3 tags, I hope to extend to FLAC, but that only works if metadata is kept exactly in its original content. Scott Burkhart Scott Burkhart Effects, LLC http://www.scotteffx.com/ scott at scotteffx.com https://www.linkedin.com/in/scotteffx 1-925-202-8852

Op do 3 nov. 2022 om 19:39 schreef Federico Miyara <fmiyara at fceia.unr.edu.ar>: > > > Martijn, > > Currently FLAC already stores and restores most kinds of metadata corruption without problems, so in most cases the conversion is already bit-accurate. However, there are some kinds of corruption it cannot handle. These are the kinds of corruption that invalidate your

Hello, I am having a problem using sqldf. I'm trying to choose a subset of observations from a data set based on the date and maximum value of a variable by date. Here is the code I am using: test<-sqldf("select distinct * from bextuse group by sdate having bext=max(bext)",method="raw"); The result I get back is a data frame with 0 rows and 0 columns. I have tried

Dear All, I've sent yesterday this issue, but as it contained two screenshots, I'm not sure if it reached the list. I post it again with some updates. 1) I had encoded a WAV file three years ago. Examining the FLAC file with an HEX/text viewer, i find at the beginning, after some headers, it declares reference libFLAC 1.3.1 20141125. However, I'm pretty sure I didn't use that

Hi Has anybody done THD or THD-N measurements with the CELT Codec (bext would be on various bit rates) If someone could share results for Mono at 64kBit and Stereo at 128 and 196kBit it would be great. thank you very much Jochen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20100817/b1da17cc/attachment-0002.htm

Martijn, > Currently FLAC already stores and restores most kinds of metadata > corruption without problems, so in most cases the conversion is > already bit-accurate. However, there are some kinds of corruption it > cannot handle. These are the kinds of corruption that invalidate your > considerations. For example, when a chunk length is incorrect, the > location and length

The patch I submitted only reads AIFF files. I'm about to start the patch to write AIFF files. To do so, we need a command-line option to specify AIFF. My inclination is to add an option: -ff { raw | wav | aif } In some sense, "-ff" is silly since it probably stands for "format format". Still, I think it's better than just "-f", since the first

Dear Friends, I am new to this mailing list. I am with the National University of Rosario, Argentina, and I am writing a book on software-based acoustical measurements, which includes a chapter on FLAC for archival and streaming purposes from an remote embedded system including a sensor. I would like to ask why the seekpoint information in the seek table metadata block reserves 64 bit for

John Miles wrote: > You'll definitely have to encode in little-endian form. PCM audio is always > little-endian, to the best of my knowledge. I have never seen any > big-endian audio data files. Try AIFF, AU and IFF just for starters. They are definietly not the only ones. Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo

I built ogg vorbis from the rc1 cvs source on Irix 6.5.12 with gcc 2.95.2. Using oggenc I encoded about 8,000 aiff files but found about a dozen where oggenc would go into an infinite loop. I tracked the problem with Laguerre_With_Deflation() as far back as logmask being Inf in floor0_forward. I'm now building gcc 3.0 with the expectation this is a compiler issue. If not, I'll back

Hi, I am attempting to use flac to encode 96k/24-bit broadcast wav (BWF) files. BWFs are wav files with some extra meta-data chunks, and is the favoured archival format for many institutions around the world. These files are encoded successfully by flac, however the resulting flac file is not playable on all flac players - it plays successfully in foobar2000 but is silent in winamp, and when

On 05-06-13 00:27, Erik de Castro Lopo wrote: > Martijn van Beurden wrote: >> Considering flac.sourceforge.net, is this ever going to be updated? In >> case it should be redirected, I checked on my own sourceforge project >> webpage, adding the following two lines to .htaccess should redirect >> traffic to any resource on flac.sourceforge.net to xiph.org/flac >>

Is anyone working on making oggenc use some nice library that reads lots of different audio file formats so that people can encode from files other than WAV? For example, the 'AudioFile' library (just found it via a search on freshmeat) reads wav, aiff, aiff-c, .au, and .snd. Maybe there are other better libraries (OpenAL, maybe), but this library does what I want (read AIFF). -tim ---

I know this is probably blindingly obvious, but I can't find it anywhere. Is there a way to tell the Vorbis codec what endian your samples are or do I have to do my own byte reversal. AIFF is Be's 'standard' audio file format, and its encoded big endian, not little endian even on Intel boxes. I know for decoding ov_read has an endian option, but I can't see an equaivalent for

Hello, I'm on a Mac and I'm noticing that encoding via the flac command line is significantly slower with version 1.3.0 than 1.2.1. I'm encoding a 24/96 file to flac, both from wav and aiff and both formats are showing the same speed decrease when using 1.3.0. Also, 1.2.1 will not encode an aiff-c file, but 1.3.0 will. Is this change documented anywhere? Thanks, Scott --------------

Federico Miyara wrote: ... > The file format allows some unused fields for future use, such as the > padding block. It could include a flag to indicate a change in the > format adding one more streaminfo byte which would allow up to 256 > channels (actually, 256 + 8), or it could trigger a new byte when 11111111. > > There is also an invalid block identifier (127) which could be

Hello, I'm using flac for an application. The fact is that i'm trying to encode a file that is not necessary a wav file. So, at first, i would like to know if it's possible to encode a file different from the wav file with the flac encoder ? When i put my data into : ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need); after some rounds in my boucl i get : encoding: