similar to: [PATCH] treat negative qlp_shift as an error

Torsdag, 23 januar 2003, skrev Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch- link.de>: >According to the format description, the coding method has to be 0. > >I've been using libFLAC 1.0.4 to encode the stream. I've checked my interpretation against "flac -a" and it seems to read 17 bits for each warmup sample. Here is its output: frame=168 blocksize=4608

I have noticed lpc slowdown both in encoding and decoding, not related to new config.h stuff. It seems there is wrong choosing of fastest possible version of lpc function. Patch is attached. -- Miroslav Lichvar -------------- next part -------------- Index: src/libFLAC/stream_decoder.c =================================================================== RCS file:

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

Here's what I listed in that email. Merging doesn't appear to be necessary. If you have any build problems, let me know. Note that my detection code is Darwin-specific. It's a BSD call (sysctl()), so a change to the platform-detection macros should enable it to work on other BSDs. However, I don't know what that would be, and I couldn't determine any safe way to do the check

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)

I don't understand the comment in src/libFLAC/stream_decoder.c: /*@@@@@@ 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) << 32) ) */ if(bps + subframe->qlp_coeff_precision + FLAC__bitmath_ilog2(order) <= 32) see

Martijn van Beurden wrote: > Yes, but that MAX value is used to loop over the > qlp_coeff_precision values between MIN and MAX. So, if the > qlp_coeff_precision value is limited in the loop but MAX is not > limited, the loop does the exact same thing multiple times: a > waste of time. Therefore, only the MAX should be limited. > > I don't think the logic is needed

libFLAC limits the value of qlp_coeff_precision to make sure that 32-bit math is enough for decoding of 16-bit audio. subframe_bps can be equal to 17 for 16-bit input (for side channel). So the value of subframe_bps should be compared with 17, not 16 (see also http://git.xiph.org/?p=flac.git;a=blob;f=src/libFLAC/stream_encoder.c;hb=02591f6b4b09e7c6e26bd2ae7225ff2810b227bb#l3431 ) --------------

Martijn van Beurden wrote: > Or maybe the question is: why is this code in evaluate_lpc_subframe anyway, > i.e, why is this code duplicated? If process_subframe_ sets the > qlp_precision for evaluate_lpc_subframe, why should the latter ignore that? > > We can only speculate about this, but I think this code has been duplicated > by accident, and therefore it wasn't changed

On Thu, Jun 19, 2014 at 06:25:57PM +0400, lvqcl wrote: > Now I wonder why evaluate_lpc_subframe_() function in stream_encoder.c contains > almost the same code, but without any comments that it's not enough pessimistic: > evaluate_lpc_subframe_(): > > if(subframe_bps + qlp_coeff_precision + FLAC__bitmath_ilog2(order) <= 32) > if(subframe_bps <= 16 &&

In the precompute_partition_info_sums_ function, instead of selecting 64-bit accumulator when the signal bps is larger than 16, revert to the original approach based on partition size, but make room for few extra bits to not overflow with unusual signals where the average residual magnitude may be larger than bps. It slightly improves the performance with standard encoding levels and 16-bit files

--- configure.ac | 7 +++++ src/libFLAC/bitreader.c | 12 ++------- src/libFLAC/bitwriter.c | 8 ++---- src/libFLAC/fixed.c | 18 +++++-------- src/libFLAC/format.c | 8 ++---- src/libFLAC/include/private/macros.h | 29 ++++++++++++++++++++ src/libFLAC/metadata_iterators.c | 17 +++---------

On Mon, Apr 20, 2015 at 08:00:20PM +0300, lvqcl wrote: > I don't understand the comment in src/libFLAC/stream_decoder.c: > > /*@@@@@@ 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) << 32) ) > */ > if(bps +

>From the FLAC format description: <5> Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement). What happens when the shift is negative? Problem is that libFLAC decoder (and possibly all other decoders) uses always '>>' operator and this operator shifts only to right. (int)x >> -1 is the same as (int)x

I'm attempting to flac a 18GB raw file, and as per 1328191 I compiled CVS, when it gets to 4GB it doesn't continue. I've tried stracing and get no output after a certain point. I also have tried oggflac and it is no different. I configured with no options or special CFLAGS and this is on a x86 machine. Is there anything special I need to do to get this to work? I have also tried with

On Fri, Jul 20, 2001 at 08:14:55PM -0700, Josh Coalson wrote: > --- Matt Zimmerman <mdz@debian.org> wrote: > > On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > > > > > This version seems to work at least partially on ia64. I am able > > to encode my > > > usual test WAV file now, but I still get a segfault during the > >

On Sat, May 19, 2001 at 12:55:08AM -0400, Matt Zimmerman wrote: > On Sat, May 19, 2001 at 02:05:14AM +0000, Christian Weisgerber wrote: > > > Problems in FLAC 0.9: > > > > - On alpha, flac immediately dumps core for both encoding and > > decoding (FreeBSD/alpha). > > I have reproduced this on Debian/alpha as well. I will spend some time > debugging it

On 6/3/06, Avuton Olrich <avuton@gmail.com> wrote: ...stuff... I just ran gdb on it, when it got to the point that it goes no further I sent kill -11 and got the following (hope it helps): Core was generated by `flac -o hs.flac -0 --force-raw-format --channels=2 --bps=16 --sample-rate=44100'. Program terminated with signal 11, Segmentation fault. warning: Can't read pathname for

Hi all, I was investigating the behaviour of the -p switch of flac. This switch should enable exhaustive search for the optimal qlp coefficient precision, but the resulting files are usually 0.5% larger then when not using the switch. I stumbled upon this code in stream_encoder > if(encoder->protected_->do_qlp_coeff_prec_search) { > min_qlp_coeff_precision =

Yes, this is such a case. However, implementing this in a future encoder/decoder would break compatibility with most (likely all) existing decoders, and only in some very, very rare cases where the material is such that the encoder chooses to use negative shifts, which makes it even harder to troubleshoot. Furthermore, as this can only be used in very rare cases, there is no benefit from allowing