similar to: FLAC QUESTION

Hello, in fact : this error(FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DAT) occurs when the sample size is bigger than the exepected size in the encoder. Now, can someone tell me how to model the signal, for example : *using a Fixed linear predictor for the input signal ? * 2012/4/13 Brian Willoughby <brianw at sounds.wa.com> > > On Apr 12, 2012, at 07:12, Rafael Velasquez

On Apr 12, 2012, at 07:12, Rafael Velasquez wrote: > 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 ? First of all, the FLAC API does not work with WAV format data -

On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote: > Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote: > >> - channels = 2; >> - bps = 16; >> + channels = ((unsigned)buffer[23] << 8) | buffer[22]; >> + bps = ((unsigned)buffer[35] << 8) | buffer[34]; >> total_samples = (((((((unsigned)buffer[43] << 8)

Hi, In the last days I've been taking as reference the example found in examples/c/encode/file/main.c. With it I've been able to encode a 2ch, 16 bps, 44100 sample rate input WAV file to a FLAC file. Now I've been trying to modify this example to encode a 2ch, 24 bps, 96000 sample rate WAV file. I have to say I'm a bit lost on how I should read the input file in this case, and

On Jan 2, 2007, at 5:15 AM, Arek Korbik wrote: > The binary frameworks distribution, as found on: > http://xiph.org/quicktime/download.html , hasn't really been tested > much outside the XiphQT project. I never had any problems with them, > but any improvement suggestions or help is welcome. Well, ok, then if the framework is intended to work, then let me describe closer the

My os is linux (Kubuntu). What do u mean by grabbing the whole tarball ? The whole tarball that is here : http://flac.cvs.sourceforge.net/viewvc/flac/flac/ ? 2012/4/4 Erik de Castro Lopo <mle+la at mega-nerd.com> > Rafael Velasquez wrote: > > > Hello, i'm new with Flac and C. > > What OS are you running on? > > > I download the example from your site ( >

Rafael Velasquez wrote: > I have installed libflac via synaptic, the version is 1.2.1-3. If you want to compile some other code against libflac, you also need to install libflac-dev. However, the code you posted a link to does not compile on it's own, it requires other code in the source code tarball. Erik -- ---------------------------------------------------------------------- Erik

Hi, I'm working on writing a FLAC encoding plugin for a personal cd ripping project of mine which uses paranoia for the raw audio extraction. My basic setup which follows gets me oddly high pitched audio with lots of noise (although the music IS somewhat recognizable). #include <FLAC/stream_encoder.h> FLAC__StreamEncoder *encoder; FILE *output_file_descriptor; encoder =

Hello everyone, I'm new to this flac thing (started about a week ago) but I have read a lot about flac and replaygain. As far as I understand it, replaygain is lossless in the sense that I can tell my player to ignore the settings or I can even use foobar2000 to remove the tags entirely, hence getting back to the original audio. If that is the case, why is there a warning in the foobar2000

I installed samba on an Intel based Server running Red Hat 5.2. I can access files on the shares I created but I run into a problem accessing a database program written in MS Access. It seems that only one person at a time can open the database. I get an error of "File is already in use" . What are the optimal settings for accessing MS Access in a multi-user application? Thanks for

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

Hello, i'm new with Flac and C. I download the example from your site ( http://flac.cvs.sourceforge.net/viewvc/flac/flac/examples/). I tried to compile the program in C by using : automake Makefile.am and i have the next message : *`configure.ac' or `configure.in' is required* So i would like to know how to do for compile the program. I tried to look in google, but i found no

Hello, I still having problem for the compilation. i do : make -f Makefile.lite This is the error : */usr/bin/ld: bitreader.release.o: relocation R_X86_64_32S against `FLAC__crc16_table' can not be used when making a shared object; recompile with -fPIC bitreader.release.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[2]: *** [../../obj/release/lib/libFLAC.so]

Hello, I manage to successfully write 8, 16 and 24 bit, all stereo, FLAC files. But when I try to write 32 bit FLACs my program crashes. *FLAC__stream_encoder_set_bits_per_sample *is called to match the desired bit depth (8, 16, 24, 32) It's the same code for all bit depths, i provide a fixed-size signed int buffer to the lib (size=16384), with values with appropriate ranges for each bit

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

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

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