similar to: Encoder example for 24-bit files

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 List, I recently was assigned a task to port FLAC Encoder to our embedded platform. Thanks to OO-like design of the libFLAC and throught documentation, that porting went like a charm. I had some problems with chmod/chown like routines while porting but I was able to safely remove that piece of code without any trouble. I have observed that the my application FLAC Encoder failes in

samples in FLAC are always signed. they must be signed going into the encoder (flac converts unsigned samples to signed) and they come out of the decoder signed. Josh --- David Bishop <tech@bishop.dhs.org> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Followup to original question: how do I determine if a particular > flac file is > signed (and then, if

Hello. I create FLAC file decoding, processing and playing program and have the following question : how to convert FLAC 16 bit file data to 32 bit float buffer for CPU processing? I've already inplemented sound playing and tested it with sine wave - it works without problems; I even made writing into decoding buffer values of sine wave, instead of decoded FLAC file data, and it also works

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

lvqcl wrote: > Erik de Castro Lopo wrote: > > >> FLAC__int16 s16buffer[FLAC__MAX_BLOCK_SIZE * FLAC__MAX_CHANNELS * sizeof(FLAC__int32)/sizeof(FLAC__int16)]; > >> > >> instead of > >> > >> FLAC__int16 s16buffer[FLAC__MAX_BLOCK_SIZE * FLAC__MAX_CHANNELS * sizeof(FLAC__int16)]; > > > > Really? Would you also write this? : > > >

Hi all, I'm attempting to encode raw audio data using libFLAC++. My audio data is 16 bit, mono, 16000Hz. I set all the appropriate parameters on the encoder and then call init(). Everything appears to be ok. I don't know how to properly convert from char *data to the FLAC__int32 *[] requested by the process function. I think this is where my problem is. If I call process() like this:

When I was trying to find yesterday's xmms-plugin bug, i have noticed that seeking in stream without seek-table isn't very good. With attached patch it is much better. -- Miroslav Lichvar -------------- next part -------------- --- src/libFLAC/seekable_stream_decoder.c.orig 2003-02-26 19:41:51.000000000 +0100 +++ src/libFLAC/seekable_stream_decoder.c 2003-07-09 23:49:35.000000000 +0200

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 [meta: Thanks, Josh, for clearing that up] Continuing my quest to make everyone think I'm an idiot, could someone explain the following code to me? I.e., what the two for loops are accomplishing, and what, *exactly*, is in s16buffer when it's done. Assume I want to write a .wav file from this decoded flac*. Is it already in approximately

Ok, here is a patch waiting for new CVS :). It works fine for me, but please check it before commiting... -- Miroslav Lichvar -------------- next part -------------- --- src/libFLAC/bitbuffer.c.orig 2003-01-30 17:36:01.000000000 +0100 +++ src/libFLAC/bitbuffer.c 2003-01-30 21:53:18.000000000 +0100 @@ -51,6 +51,25 @@ */ static const unsigned FLAC__BITBUFFER_DEFAULT_CAPACITY = ((65536 - 64) *

I think I've gotten FLAC__lpc_restore_signal() about as good as I'm going to get it. Here's what I have: -a new file, lpc_asm.s, which has the assembly routines -changes to cpu.h, cpu.c, and stream_decoder.c to enable them -changes to configure.in to support the new cpu stuff -a preliminary Makefile.am -maybe something else I'm forgetting Now automake complains that configure.in

This commit will break OS/2's EMX 0.9d library (GCC 2.8.1) which has been been replaced by klibc. Considering the age of EMX and lack of testing and that klibc contains so many improvements I think this is exceptable. --- include/FLAC/ordinals.h | 17 +++++++++-------- src/flac/main.c | 2 +- src/libFLAC/metadata_iterators.c | 2 +-

Martijn van Beurden wrote: > Like I reported just before the release of 1.3.0 (mail of Fri, > 05 Apr 2013 08:25:10 +0200, to be specific), compiling on > Raspbian (Debian Wheezy, GCC 4.6) returns quite some warnings of > the type -Wcast-align. What happens if you change the definitions of s8buffer[] and ucbuffer_[] arrays as follows: static __attribute__((__aligned__(4))) FLAC__int8

Ok, the patch from 2003 about improving seeking still didn't make it to CVS, so here is another try. I made some benchmarking with the test_seeking utility from flac sources to show how bad the current seeking is, especially without seektable. Track used for the experiment had about 50 minutes. In the following table is average number of seeks and number of decoded frames required for one

On Sat, Mar 8, 2014 at 7:49 AM, ??????? ?????? <rodionkarimov at yandex.ru>wrote: > I create FLAC file decoding, processing and playing program and have the > following question : how to convert FLAC 16 bit file data to 32 bit float > buffer for CPU processing? I've already inplemented sound playing and > tested it with sine wave - it works without problems; I even made

On Thu, Jun 19, 2014 at 03:30:22PM +0400, lvqcl wrote: > BTW, what can you say about the following place in stream_decoder.c > in read_subframe_lpc_() function: > > /*@@@@@@ technically not pessimistic enough, should be more like > if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1)

Hi, This patch improves compression a very tiny bit on average, but up to 0.1 percentage point for classical music. I haven't found any tracks that show worsening compression with this patch. -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Improve-LPC-order-guess.patch Type: text/x-patch Size: 0 bytes Desc: not available Url :

Hi I've found bugs in vorbis-tools and wrote this patch to fix them: 1st part is to fix single-channel flac files. 2nd part fixes a problem with thread race causing lockup if you press Ctrl-C Ctrl-C sets sig_request.cancel, buffer thread exits if it sees sig_request.cancel However, if Ctrl-C was pressed here do { if (nthc-- == 0) { ----------------> Ctrl-C

Erik de Castro Lopo <mle+la <at> mega-nerd.com> writes: > > Collin wrote: > > > Has anyone encountered a similar problem, or are there any known issues that > > could explain this? > > No known issues of this kind, but I would like to see a repeatable test > case so I can investigate further. > > Cheers, > Erik It turns out it was an error