mirror of https://github.com/PCSX2/pcsx2.git
Merge pull request #790 from micove/Update_3rdparty
Update some of the 3rd party libraries.
This commit is contained in:
micove
commit
e65a223c91
File diff suppressed because it is too large
Load Diff
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|
@ -0,0 +1,327 @@
|
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------------------------------------------------------------------
|
||||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
||||
This program is released under the terms of the license contained
|
||||
in the file LICENSE.
|
||||
------------------------------------------------------------------
|
||||
|
||||
|
||||
0.9.0
|
||||
~~~~~
|
||||
First version.
|
||||
|
||||
|
||||
0.9.0a
|
||||
~~~~~~
|
||||
Removed 'ranlib' from Makefile, since most modern Unix-es
|
||||
don't need it, or even know about it.
|
||||
|
||||
|
||||
0.9.0b
|
||||
~~~~~~
|
||||
Fixed a problem with error reporting in bzip2.c. This does not effect
|
||||
the library in any way. Problem is: versions 0.9.0 and 0.9.0a (of the
|
||||
program proper) compress and decompress correctly, but give misleading
|
||||
error messages (internal panics) when an I/O error occurs, instead of
|
||||
reporting the problem correctly. This shouldn't give any data loss
|
||||
(as far as I can see), but is confusing.
|
||||
|
||||
Made the inline declarations disappear for non-GCC compilers.
|
||||
|
||||
|
||||
0.9.0c
|
||||
~~~~~~
|
||||
Fixed some problems in the library pertaining to some boundary cases.
|
||||
This makes the library behave more correctly in those situations. The
|
||||
fixes apply only to features (calls and parameters) not used by
|
||||
bzip2.c, so the non-fixedness of them in previous versions has no
|
||||
effect on reliability of bzip2.c.
|
||||
|
||||
In bzlib.c:
|
||||
* made zero-length BZ_FLUSH work correctly in bzCompress().
|
||||
* fixed bzWrite/bzRead to ignore zero-length requests.
|
||||
* fixed bzread to correctly handle read requests after EOF.
|
||||
* wrong parameter order in call to bzDecompressInit in
|
||||
bzBuffToBuffDecompress. Fixed.
|
||||
|
||||
In compress.c:
|
||||
* changed setting of nGroups in sendMTFValues() so as to
|
||||
do a bit better on small files. This _does_ effect
|
||||
bzip2.c.
|
||||
|
||||
|
||||
0.9.5a
|
||||
~~~~~~
|
||||
Major change: add a fallback sorting algorithm (blocksort.c)
|
||||
to give reasonable behaviour even for very repetitive inputs.
|
||||
Nuked --repetitive-best and --repetitive-fast since they are
|
||||
no longer useful.
|
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|
||||
Minor changes: mostly a whole bunch of small changes/
|
||||
bugfixes in the driver (bzip2.c). Changes pertaining to the
|
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user interface are:
|
||||
|
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allow decompression of symlink'd files to stdout
|
||||
decompress/test files even without .bz2 extension
|
||||
give more accurate error messages for I/O errors
|
||||
when compressing/decompressing to stdout, don't catch control-C
|
||||
read flags from BZIP2 and BZIP environment variables
|
||||
decline to break hard links to a file unless forced with -f
|
||||
allow -c flag even with no filenames
|
||||
preserve file ownerships as far as possible
|
||||
make -s -1 give the expected block size (100k)
|
||||
add a flag -q --quiet to suppress nonessential warnings
|
||||
stop decoding flags after --, so files beginning in - can be handled
|
||||
resolved inconsistent naming: bzcat or bz2cat ?
|
||||
bzip2 --help now returns 0
|
||||
|
||||
Programming-level changes are:
|
||||
|
||||
fixed syntax error in GET_LL4 for Borland C++ 5.02
|
||||
let bzBuffToBuffDecompress return BZ_DATA_ERROR{_MAGIC}
|
||||
fix overshoot of mode-string end in bzopen_or_bzdopen
|
||||
wrapped bzlib.h in #ifdef __cplusplus ... extern "C" { ... }
|
||||
close file handles under all error conditions
|
||||
added minor mods so it compiles with DJGPP out of the box
|
||||
fixed Makefile so it doesn't give problems with BSD make
|
||||
fix uninitialised memory reads in dlltest.c
|
||||
|
||||
0.9.5b
|
||||
~~~~~~
|
||||
Open stdin/stdout in binary mode for DJGPP.
|
||||
|
||||
0.9.5c
|
||||
~~~~~~
|
||||
Changed BZ_N_OVERSHOOT to be ... + 2 instead of ... + 1. The + 1
|
||||
version could cause the sorted order to be wrong in some extremely
|
||||
obscure cases. Also changed setting of quadrant in blocksort.c.
|
||||
|
||||
0.9.5d
|
||||
~~~~~~
|
||||
The only functional change is to make bzlibVersion() in the library
|
||||
return the correct string. This has no effect whatsoever on the
|
||||
functioning of the bzip2 program or library. Added a couple of casts
|
||||
so the library compiles without warnings at level 3 in MS Visual
|
||||
Studio 6.0. Included a Y2K statement in the file Y2K_INFO. All other
|
||||
changes are minor documentation changes.
|
||||
|
||||
1.0
|
||||
~~~
|
||||
Several minor bugfixes and enhancements:
|
||||
|
||||
* Large file support. The library uses 64-bit counters to
|
||||
count the volume of data passing through it. bzip2.c
|
||||
is now compiled with -D_FILE_OFFSET_BITS=64 to get large
|
||||
file support from the C library. -v correctly prints out
|
||||
file sizes greater than 4 gigabytes. All these changes have
|
||||
been made without assuming a 64-bit platform or a C compiler
|
||||
which supports 64-bit ints, so, except for the C library
|
||||
aspect, they are fully portable.
|
||||
|
||||
* Decompression robustness. The library/program should be
|
||||
robust to any corruption of compressed data, detecting and
|
||||
handling _all_ corruption, instead of merely relying on
|
||||
the CRCs. What this means is that the program should
|
||||
never crash, given corrupted data, and the library should
|
||||
always return BZ_DATA_ERROR.
|
||||
|
||||
* Fixed an obscure race-condition bug only ever observed on
|
||||
Solaris, in which, if you were very unlucky and issued
|
||||
control-C at exactly the wrong time, both input and output
|
||||
files would be deleted.
|
||||
|
||||
* Don't run out of file handles on test/decompression when
|
||||
large numbers of files have invalid magic numbers.
|
||||
|
||||
* Avoid library namespace pollution. Prefix all exported
|
||||
symbols with BZ2_.
|
||||
|
||||
* Minor sorting enhancements from my DCC2000 paper.
|
||||
|
||||
* Advance the version number to 1.0, so as to counteract the
|
||||
(false-in-this-case) impression some people have that programs
|
||||
with version numbers less than 1.0 are in some way, experimental,
|
||||
pre-release versions.
|
||||
|
||||
* Create an initial Makefile-libbz2_so to build a shared library.
|
||||
Yes, I know I should really use libtool et al ...
|
||||
|
||||
* Make the program exit with 2 instead of 0 when decompression
|
||||
fails due to a bad magic number (ie, an invalid bzip2 header).
|
||||
Also exit with 1 (as the manual claims :-) whenever a diagnostic
|
||||
message would have been printed AND the corresponding operation
|
||||
is aborted, for example
|
||||
bzip2: Output file xx already exists.
|
||||
When a diagnostic message is printed but the operation is not
|
||||
aborted, for example
|
||||
bzip2: Can't guess original name for wurble -- using wurble.out
|
||||
then the exit value 0 is returned, unless some other problem is
|
||||
also detected.
|
||||
|
||||
I think it corresponds more closely to what the manual claims now.
|
||||
|
||||
|
||||
1.0.1
|
||||
~~~~~
|
||||
* Modified dlltest.c so it uses the new BZ2_ naming scheme.
|
||||
* Modified makefile-msc to fix minor build probs on Win2k.
|
||||
* Updated README.COMPILATION.PROBLEMS.
|
||||
|
||||
There are no functionality changes or bug fixes relative to version
|
||||
1.0.0. This is just a documentation update + a fix for minor Win32
|
||||
build problems. For almost everyone, upgrading from 1.0.0 to 1.0.1 is
|
||||
utterly pointless. Don't bother.
|
||||
|
||||
|
||||
1.0.2
|
||||
~~~~~
|
||||
A bug fix release, addressing various minor issues which have appeared
|
||||
in the 18 or so months since 1.0.1 was released. Most of the fixes
|
||||
are to do with file-handling or documentation bugs. To the best of my
|
||||
knowledge, there have been no data-loss-causing bugs reported in the
|
||||
compression/decompression engine of 1.0.0 or 1.0.1.
|
||||
|
||||
Note that this release does not improve the rather crude build system
|
||||
for Unix platforms. The general plan here is to autoconfiscate/
|
||||
libtoolise 1.0.2 soon after release, and release the result as 1.1.0
|
||||
or perhaps 1.2.0. That, however, is still just a plan at this point.
|
||||
|
||||
Here are the changes in 1.0.2. Bug-reporters and/or patch-senders in
|
||||
parentheses.
|
||||
|
||||
* Fix an infinite segfault loop in 1.0.1 when a directory is
|
||||
encountered in -f (force) mode.
|
||||
(Trond Eivind Glomsrod, Nicholas Nethercote, Volker Schmidt)
|
||||
|
||||
* Avoid double fclose() of output file on certain I/O error paths.
|
||||
(Solar Designer)
|
||||
|
||||
* Don't fail with internal error 1007 when fed a long stream (> 48MB)
|
||||
of byte 251. Also print useful message suggesting that 1007s may be
|
||||
caused by bad memory.
|
||||
(noticed by Juan Pedro Vallejo, fixed by me)
|
||||
|
||||
* Fix uninitialised variable silly bug in demo prog dlltest.c.
|
||||
(Jorj Bauer)
|
||||
|
||||
* Remove 512-MB limitation on recovered file size for bzip2recover
|
||||
on selected platforms which support 64-bit ints. At the moment
|
||||
all GCC supported platforms, and Win32.
|
||||
(me, Alson van der Meulen)
|
||||
|
||||
* Hard-code header byte values, to give correct operation on platforms
|
||||
using EBCDIC as their native character set (IBM's OS/390).
|
||||
(Leland Lucius)
|
||||
|
||||
* Copy file access times correctly.
|
||||
(Marty Leisner)
|
||||
|
||||
* Add distclean and check targets to Makefile.
|
||||
(Michael Carmack)
|
||||
|
||||
* Parameterise use of ar and ranlib in Makefile. Also add $(LDFLAGS).
|
||||
(Rich Ireland, Bo Thorsen)
|
||||
|
||||
* Pass -p (create parent dirs as needed) to mkdir during make install.
|
||||
(Jeremy Fusco)
|
||||
|
||||
* Dereference symlinks when copying file permissions in -f mode.
|
||||
(Volker Schmidt)
|
||||
|
||||
* Majorly simplify implementation of uInt64_qrm10.
|
||||
(Bo Lindbergh)
|
||||
|
||||
* Check the input file still exists before deleting the output one,
|
||||
when aborting in cleanUpAndFail().
|
||||
(Joerg Prante, Robert Linden, Matthias Krings)
|
||||
|
||||
Also a bunch of patches courtesy of Philippe Troin, the Debian maintainer
|
||||
of bzip2:
|
||||
|
||||
* Wrapper scripts (with manpages): bzdiff, bzgrep, bzmore.
|
||||
|
||||
* Spelling changes and minor enhancements in bzip2.1.
|
||||
|
||||
* Avoid race condition between creating the output file and setting its
|
||||
interim permissions safely, by using fopen_output_safely().
|
||||
No changes to bzip2recover since there is no issue with file
|
||||
permissions there.
|
||||
|
||||
* do not print senseless report with -v when compressing an empty
|
||||
file.
|
||||
|
||||
* bzcat -f works on non-bzip2 files.
|
||||
|
||||
* do not try to escape shell meta-characters on unix (the shell takes
|
||||
care of these).
|
||||
|
||||
* added --fast and --best aliases for -1 -9 for gzip compatibility.
|
||||
|
||||
|
||||
1.0.3 (15 Feb 05)
|
||||
~~~~~~~~~~~~~~~~~
|
||||
Fixes some minor bugs since the last version, 1.0.2.
|
||||
|
||||
* Further robustification against corrupted compressed data.
|
||||
There are currently no known bitstreams which can cause the
|
||||
decompressor to crash, loop or access memory which does not
|
||||
belong to it. If you are using bzip2 or the library to
|
||||
decompress bitstreams from untrusted sources, an upgrade
|
||||
to 1.0.3 is recommended. This fixes CAN-2005-1260.
|
||||
|
||||
* The documentation has been converted to XML, from which html
|
||||
and pdf can be derived.
|
||||
|
||||
* Various minor bugs in the documentation have been fixed.
|
||||
|
||||
* Fixes for various compilation warnings with newer versions of
|
||||
gcc, and on 64-bit platforms.
|
||||
|
||||
* The BZ_NO_STDIO cpp symbol was not properly observed in 1.0.2.
|
||||
This has been fixed.
|
||||
|
||||
|
||||
1.0.4 (20 Dec 06)
|
||||
~~~~~~~~~~~~~~~~~
|
||||
Fixes some minor bugs since the last version, 1.0.3.
|
||||
|
||||
* Fix file permissions race problem (CAN-2005-0953).
|
||||
|
||||
* Avoid possible segfault in BZ2_bzclose. From Coverity's NetBSD
|
||||
scan.
|
||||
|
||||
* 'const'/prototype cleanups in the C code.
|
||||
|
||||
* Change default install location to /usr/local, and handle multiple
|
||||
'make install's without error.
|
||||
|
||||
* Sanitise file names more carefully in bzgrep. Fixes CAN-2005-0758
|
||||
to the extent that applies to bzgrep.
|
||||
|
||||
* Use 'mktemp' rather than 'tempfile' in bzdiff.
|
||||
|
||||
* Tighten up a couple of assertions in blocksort.c following automated
|
||||
analysis.
|
||||
|
||||
* Fix minor doc/comment bugs.
|
||||
|
||||
|
||||
1.0.5 (10 Dec 07)
|
||||
~~~~~~~~~~~~~~~~~
|
||||
Security fix only. Fixes CERT-FI 20469 as it applies to bzip2.
|
||||
|
||||
|
||||
1.0.6 (6 Sept 10)
|
||||
~~~~~~~~~~~~~~~~~
|
||||
|
||||
* Security fix for CVE-2010-0405. This was reported by Mikolaj
|
||||
Izdebski.
|
||||
|
||||
* Make the documentation build on Ubuntu 10.04
|
||||
|
|
@ -2,7 +2,7 @@
|
|||
--------------------------------------------------------------------------
|
||||
|
||||
This program, "bzip2", the associated library "libbzip2", and all
|
||||
documentation, are copyright (C) 1996-2006 Julian R Seward. All
|
||||
documentation, are copyright (C) 1996-2010 Julian R Seward. All
|
||||
rights reserved.
|
||||
|
||||
Redistribution and use in source and binary forms, with or without
|
||||
|
|
@ -36,8 +36,7 @@ WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
|
|||
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
|
||||
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
|
||||
Julian Seward, Cambridge, UK.
|
||||
jseward@bzip.org
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Julian Seward, jseward@bzip.org
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
|
||||
--------------------------------------------------------------------------
|
||||
|
|
|
|||
|
|
@ -6,8 +6,8 @@ This version is fully compatible with the previous public releases.
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in this file.
|
||||
|
||||
|
|
@ -177,6 +177,14 @@ WHAT'S NEW IN 1.0.4 ?
|
|||
|
||||
See the CHANGES file.
|
||||
|
||||
WHAT'S NEW IN 1.0.5 ?
|
||||
|
||||
See the CHANGES file.
|
||||
|
||||
WHAT'S NEW IN 1.0.6 ?
|
||||
|
||||
See the CHANGES file.
|
||||
|
||||
|
||||
I hope you find bzip2 useful. Feel free to contact me at
|
||||
jseward@bzip.org
|
||||
|
|
@ -203,3 +211,5 @@ Cambridge, UK.
|
|||
30 December 2001 (bzip2, version 1.0.2pre1)
|
||||
15 February 2005 (bzip2, version 1.0.3)
|
||||
20 December 2006 (bzip2, version 1.0.4)
|
||||
10 December 2007 (bzip2, version 1.0.5)
|
||||
6 Sept 2010 (bzip2, version 1.0.6)
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
@ -48,7 +48,7 @@ void BZ2_bz__AssertH__fail ( int errcode )
|
|||
"component, you should also report this bug to the author(s)\n"
|
||||
"of that program. Please make an effort to report this bug;\n"
|
||||
"timely and accurate bug reports eventually lead to higher\n"
|
||||
"quality software. Thanks. Julian Seward, 15 February 2005.\n\n",
|
||||
"quality software. Thanks. Julian Seward, 10 December 2007.\n\n",
|
||||
errcode,
|
||||
BZ2_bzlibVersion()
|
||||
);
|
||||
|
|
@ -598,6 +598,7 @@ Bool unRLE_obuf_to_output_FAST ( DState* s )
|
|||
UInt32 c_tPos = s->tPos;
|
||||
char* cs_next_out = s->strm->next_out;
|
||||
unsigned int cs_avail_out = s->strm->avail_out;
|
||||
Int32 ro_blockSize100k = s->blockSize100k;
|
||||
/* end restore */
|
||||
|
||||
UInt32 avail_out_INIT = cs_avail_out;
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
@ -36,7 +36,7 @@
|
|||
|
||||
/*-- General stuff. --*/
|
||||
|
||||
#define BZ_VERSION "1.0.4, 20-Dec-2006"
|
||||
#define BZ_VERSION "1.0.6, 6-Sept-2010"
|
||||
|
||||
typedef char Char;
|
||||
typedef unsigned char Bool;
|
||||
|
|
@ -442,11 +442,15 @@ typedef
|
|||
/*-- Macros for decompression. --*/
|
||||
|
||||
#define BZ_GET_FAST(cccc) \
|
||||
/* c_tPos is unsigned, hence test < 0 is pointless. */ \
|
||||
if (s->tPos >= (UInt32)100000 * (UInt32)s->blockSize100k) return True; \
|
||||
s->tPos = s->tt[s->tPos]; \
|
||||
cccc = (UChar)(s->tPos & 0xff); \
|
||||
s->tPos >>= 8;
|
||||
|
||||
#define BZ_GET_FAST_C(cccc) \
|
||||
/* c_tPos is unsigned, hence test < 0 is pointless. */ \
|
||||
if (c_tPos >= (UInt32)100000 * (UInt32)ro_blockSize100k) return True; \
|
||||
c_tPos = c_tt[c_tPos]; \
|
||||
cccc = (UChar)(c_tPos & 0xff); \
|
||||
c_tPos >>= 8;
|
||||
|
|
@ -469,8 +473,10 @@ typedef
|
|||
(((UInt32)s->ll16[i]) | (GET_LL4(i) << 16))
|
||||
|
||||
#define BZ_GET_SMALL(cccc) \
|
||||
cccc = BZ2_indexIntoF ( s->tPos, s->cftab ); \
|
||||
s->tPos = GET_LL(s->tPos);
|
||||
/* c_tPos is unsigned, hence test < 0 is pointless. */ \
|
||||
if (s->tPos >= (UInt32)100000 * (UInt32)s->blockSize100k) return True; \
|
||||
cccc = BZ2_indexIntoF ( s->tPos, s->cftab ); \
|
||||
s->tPos = GET_LL(s->tPos);
|
||||
|
||||
|
||||
/*-- externs for decompression. --*/
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
@ -381,6 +381,13 @@ Int32 BZ2_decompress ( DState* s )
|
|||
es = -1;
|
||||
N = 1;
|
||||
do {
|
||||
/* Check that N doesn't get too big, so that es doesn't
|
||||
go negative. The maximum value that can be
|
||||
RUNA/RUNB encoded is equal to the block size (post
|
||||
the initial RLE), viz, 900k, so bounding N at 2
|
||||
million should guard against overflow without
|
||||
rejecting any legitimate inputs. */
|
||||
if (N >= 2*1024*1024) RETURN(BZ_DATA_ERROR);
|
||||
if (nextSym == BZ_RUNA) es = es + (0+1) * N; else
|
||||
if (nextSym == BZ_RUNB) es = es + (1+1) * N;
|
||||
N = N * 2;
|
||||
|
|
@ -485,15 +492,28 @@ Int32 BZ2_decompress ( DState* s )
|
|||
RETURN(BZ_DATA_ERROR);
|
||||
|
||||
/*-- Set up cftab to facilitate generation of T^(-1) --*/
|
||||
/* Check: unzftab entries in range. */
|
||||
for (i = 0; i <= 255; i++) {
|
||||
if (s->unzftab[i] < 0 || s->unzftab[i] > nblock)
|
||||
RETURN(BZ_DATA_ERROR);
|
||||
}
|
||||
/* Actually generate cftab. */
|
||||
s->cftab[0] = 0;
|
||||
for (i = 1; i <= 256; i++) s->cftab[i] = s->unzftab[i-1];
|
||||
for (i = 1; i <= 256; i++) s->cftab[i] += s->cftab[i-1];
|
||||
/* Check: cftab entries in range. */
|
||||
for (i = 0; i <= 256; i++) {
|
||||
if (s->cftab[i] < 0 || s->cftab[i] > nblock) {
|
||||
/* s->cftab[i] can legitimately be == nblock */
|
||||
RETURN(BZ_DATA_ERROR);
|
||||
}
|
||||
}
|
||||
/* Check: cftab entries non-descending. */
|
||||
for (i = 1; i <= 256; i++) {
|
||||
if (s->cftab[i-1] > s->cftab[i]) {
|
||||
RETURN(BZ_DATA_ERROR);
|
||||
}
|
||||
}
|
||||
|
||||
s->state_out_len = 0;
|
||||
s->state_out_ch = 0;
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -8,8 +8,8 @@
|
|||
This file is part of bzip2/libbzip2, a program and library for
|
||||
lossless, block-sorting data compression.
|
||||
|
||||
bzip2/libbzip2 version 1.0.4 of 20 December 2006
|
||||
Copyright (C) 1996-2006 Julian Seward <jseward@bzip.org>
|
||||
bzip2/libbzip2 version 1.0.6 of 6 September 2010
|
||||
Copyright (C) 1996-2010 Julian Seward <jseward@bzip.org>
|
||||
|
||||
Please read the WARNING, DISCLAIMER and PATENTS sections in the
|
||||
README file.
|
||||
|
|
|
|||
|
|
@ -1,349 +0,0 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Win32 version of the AMD 3DNow! optimized routines for AMD K6-2/Athlon
|
||||
/// processors. All 3DNow! optimized functions have been gathered into this
|
||||
/// single source code file, regardless to their class or original source code
|
||||
/// file, in order to ease porting the library to other compiler and processor
|
||||
/// platforms.
|
||||
///
|
||||
/// By the way; the performance gain depends heavily on the CPU generation: On
|
||||
/// K6-2 these routines provided speed-up of even 2.4 times, while on Athlon the
|
||||
/// difference to the original routines stayed at unremarkable 8%! Such a small
|
||||
/// improvement on Athlon is due to 3DNow can perform only two operations in
|
||||
/// parallel, and obviously also the Athlon FPU is doing a very good job with
|
||||
/// the standard C floating point routines! Here these routines are anyway,
|
||||
/// although it might not be worth the effort to convert these to GCC platform,
|
||||
/// for Athlon CPU at least. The situation is different regarding the SSE
|
||||
/// optimizations though, thanks to the four parallel operations of SSE that
|
||||
/// already make a difference.
|
||||
///
|
||||
/// This file is to be compiled in Windows platform with Microsoft Visual C++
|
||||
/// Compiler. Please see '3dnow_gcc.cpp' for the gcc compiler version for all
|
||||
/// GNU platforms (if file supplied).
|
||||
///
|
||||
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
|
||||
/// 6.0 processor pack" update to support 3DNow! instruction set. The update is
|
||||
/// available for download at Microsoft Developers Network, see here:
|
||||
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
|
||||
///
|
||||
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
|
||||
/// perform a search with keywords "processor pack".
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: 3dnow_win.cpp 63 2009-02-21 16:00:14Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include "cpu_detect.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
#ifndef WIN32
|
||||
#error "wrong platform - this source code file is exclusively for Win32 platform"
|
||||
#endif
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
#ifdef ALLOW_3DNOW
|
||||
// 3DNow! routines available only with float sample type
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// implementation of 3DNow! optimized functions of class 'TDStretch3DNow'
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include "TDStretch.h"
|
||||
|
||||
|
||||
// Calculates cross correlation of two buffers
|
||||
double TDStretch3DNow::calcCrossCorrStereo(const float *pV1, const float *pV2) const
|
||||
{
|
||||
int overlapLengthLocal = overlapLength;
|
||||
float corr = 0;
|
||||
|
||||
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
|
||||
/*
|
||||
c-pseudocode:
|
||||
|
||||
corr = 0;
|
||||
for (i = 0; i < overlapLength / 4; i ++)
|
||||
{
|
||||
corr += pV1[0] * pV2[0];
|
||||
pV1[1] * pV2[1];
|
||||
pV1[2] * pV2[2];
|
||||
pV1[3] * pV2[3];
|
||||
pV1[4] * pV2[4];
|
||||
pV1[5] * pV2[5];
|
||||
pV1[6] * pV2[6];
|
||||
pV1[7] * pV2[7];
|
||||
|
||||
pV1 += 8;
|
||||
pV2 += 8;
|
||||
}
|
||||
*/
|
||||
|
||||
_asm
|
||||
{
|
||||
// give prefetch hints to CPU of what data are to be needed soonish.
|
||||
// give more aggressive hints on pV1 as that changes more between different calls
|
||||
// while pV2 stays the same.
|
||||
prefetch [pV1]
|
||||
prefetch [pV2]
|
||||
prefetch [pV1 + 32]
|
||||
|
||||
mov eax, dword ptr pV2
|
||||
mov ebx, dword ptr pV1
|
||||
|
||||
pxor mm0, mm0
|
||||
|
||||
mov ecx, overlapLengthLocal
|
||||
shr ecx, 2 // div by four
|
||||
|
||||
loop1:
|
||||
movq mm1, [eax]
|
||||
prefetch [eax + 32] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
pfmul mm1, [ebx]
|
||||
prefetch [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
|
||||
movq mm2, [eax + 8]
|
||||
pfadd mm0, mm1
|
||||
pfmul mm2, [ebx + 8]
|
||||
|
||||
movq mm3, [eax + 16]
|
||||
pfadd mm0, mm2
|
||||
pfmul mm3, [ebx + 16]
|
||||
|
||||
movq mm4, [eax + 24]
|
||||
pfadd mm0, mm3
|
||||
pfmul mm4, [ebx + 24]
|
||||
|
||||
add eax, 32
|
||||
pfadd mm0, mm4
|
||||
add ebx, 32
|
||||
|
||||
dec ecx
|
||||
jnz loop1
|
||||
|
||||
// add halfs of mm0 together and return the result.
|
||||
// note: mm1 is used as a dummy parameter only, we actually don't care about it's value
|
||||
pfacc mm0, mm1
|
||||
movd corr, mm0
|
||||
femms
|
||||
}
|
||||
|
||||
return corr;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// implementation of 3DNow! optimized functions of class 'FIRFilter'
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include "FIRFilter.h"
|
||||
|
||||
FIRFilter3DNow::FIRFilter3DNow() : FIRFilter()
|
||||
{
|
||||
filterCoeffsUnalign = NULL;
|
||||
filterCoeffsAlign = NULL;
|
||||
}
|
||||
|
||||
|
||||
FIRFilter3DNow::~FIRFilter3DNow()
|
||||
{
|
||||
delete[] filterCoeffsUnalign;
|
||||
filterCoeffsUnalign = NULL;
|
||||
filterCoeffsAlign = NULL;
|
||||
}
|
||||
|
||||
|
||||
// (overloaded) Calculates filter coefficients for 3DNow! routine
|
||||
void FIRFilter3DNow::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
|
||||
{
|
||||
uint i;
|
||||
float fDivider;
|
||||
|
||||
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
|
||||
|
||||
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
|
||||
// also rearrange coefficients suitably for 3DNow!
|
||||
// Ensure that filter coeffs array is aligned to 16-byte boundary
|
||||
delete[] filterCoeffsUnalign;
|
||||
filterCoeffsUnalign = new float[2 * newLength + 4];
|
||||
filterCoeffsAlign = (float *)(((uint)filterCoeffsUnalign + 15) & (uint)-16);
|
||||
|
||||
fDivider = (float)resultDivider;
|
||||
|
||||
// rearrange the filter coefficients for mmx routines
|
||||
for (i = 0; i < newLength; i ++)
|
||||
{
|
||||
filterCoeffsAlign[2 * i + 0] =
|
||||
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// 3DNow!-optimized version of the filter routine for stereo sound
|
||||
uint FIRFilter3DNow::evaluateFilterStereo(float *dest, const float *src, uint numSamples) const
|
||||
{
|
||||
float *filterCoeffsLocal = filterCoeffsAlign;
|
||||
uint count = (numSamples - length) & (uint)-2;
|
||||
uint lengthLocal = length / 4;
|
||||
|
||||
assert(length != 0);
|
||||
assert(count % 2 == 0);
|
||||
|
||||
/* original code:
|
||||
|
||||
double suml1, suml2;
|
||||
double sumr1, sumr2;
|
||||
uint i, j;
|
||||
|
||||
for (j = 0; j < count; j += 2)
|
||||
{
|
||||
const float *ptr;
|
||||
|
||||
suml1 = sumr1 = 0.0;
|
||||
suml2 = sumr2 = 0.0;
|
||||
ptr = src;
|
||||
filterCoeffsLocal = filterCoeffs;
|
||||
for (i = 0; i < lengthLocal; i ++)
|
||||
{
|
||||
// unroll loop for efficiency.
|
||||
|
||||
suml1 += ptr[0] * filterCoeffsLocal[0] +
|
||||
ptr[2] * filterCoeffsLocal[2] +
|
||||
ptr[4] * filterCoeffsLocal[4] +
|
||||
ptr[6] * filterCoeffsLocal[6];
|
||||
|
||||
sumr1 += ptr[1] * filterCoeffsLocal[1] +
|
||||
ptr[3] * filterCoeffsLocal[3] +
|
||||
ptr[5] * filterCoeffsLocal[5] +
|
||||
ptr[7] * filterCoeffsLocal[7];
|
||||
|
||||
suml2 += ptr[8] * filterCoeffsLocal[0] +
|
||||
ptr[10] * filterCoeffsLocal[2] +
|
||||
ptr[12] * filterCoeffsLocal[4] +
|
||||
ptr[14] * filterCoeffsLocal[6];
|
||||
|
||||
sumr2 += ptr[9] * filterCoeffsLocal[1] +
|
||||
ptr[11] * filterCoeffsLocal[3] +
|
||||
ptr[13] * filterCoeffsLocal[5] +
|
||||
ptr[15] * filterCoeffsLocal[7];
|
||||
|
||||
ptr += 16;
|
||||
filterCoeffsLocal += 8;
|
||||
}
|
||||
dest[0] = (float)suml1;
|
||||
dest[1] = (float)sumr1;
|
||||
dest[2] = (float)suml2;
|
||||
dest[3] = (float)sumr2;
|
||||
|
||||
src += 4;
|
||||
dest += 4;
|
||||
}
|
||||
|
||||
*/
|
||||
_asm
|
||||
{
|
||||
mov eax, dword ptr dest
|
||||
mov ebx, dword ptr src
|
||||
mov edx, count
|
||||
shr edx, 1
|
||||
|
||||
loop1:
|
||||
// "outer loop" : during each round 2*2 output samples are calculated
|
||||
prefetch [ebx] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
prefetch [filterCoeffsLocal] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
|
||||
mov esi, ebx
|
||||
mov edi, filterCoeffsLocal
|
||||
pxor mm0, mm0
|
||||
pxor mm1, mm1
|
||||
mov ecx, lengthLocal
|
||||
|
||||
loop2:
|
||||
// "inner loop" : during each round four FIR filter taps are evaluated for 2*2 output samples
|
||||
movq mm2, [edi]
|
||||
movq mm3, mm2
|
||||
prefetch [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
pfmul mm2, [esi]
|
||||
prefetch [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
|
||||
pfmul mm3, [esi + 8]
|
||||
|
||||
movq mm4, [edi + 8]
|
||||
movq mm5, mm4
|
||||
pfadd mm0, mm2
|
||||
pfmul mm4, [esi + 8]
|
||||
pfadd mm1, mm3
|
||||
pfmul mm5, [esi + 16]
|
||||
|
||||
movq mm2, [edi + 16]
|
||||
movq mm6, mm2
|
||||
pfadd mm0, mm4
|
||||
pfmul mm2, [esi + 16]
|
||||
pfadd mm1, mm5
|
||||
pfmul mm6, [esi + 24]
|
||||
|
||||
movq mm3, [edi + 24]
|
||||
movq mm7, mm3
|
||||
pfadd mm0, mm2
|
||||
pfmul mm3, [esi + 24]
|
||||
pfadd mm1, mm6
|
||||
pfmul mm7, [esi + 32]
|
||||
add esi, 32
|
||||
pfadd mm0, mm3
|
||||
add edi, 32
|
||||
pfadd mm1, mm7
|
||||
|
||||
dec ecx
|
||||
jnz loop2
|
||||
|
||||
movq [eax], mm0
|
||||
add ebx, 16
|
||||
movq [eax + 8], mm1
|
||||
add eax, 16
|
||||
|
||||
dec edx
|
||||
jnz loop1
|
||||
|
||||
femms
|
||||
}
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
|
||||
#endif // ALLOW_3DNOW
|
||||
|
|
@ -0,0 +1,458 @@
|
|||
GNU LESSER GENERAL PUBLIC LICENSE
|
||||
Version 2.1, February 1999
|
||||
|
||||
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
|
||||
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
Everyone is permitted to copy and distribute verbatim copies
|
||||
of this license document, but changing it is not allowed.
|
||||
|
||||
[This is the first released version of the Lesser GPL. It also counts
|
||||
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|
||||
the version number 2.1.]
|
||||
|
||||
Preamble
|
||||
|
||||
The licenses for most software are designed to take away your
|
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|
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Licenses are intended to guarantee your freedom to share and change
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|
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|
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|
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|
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|
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Thus, it is not the intent of this section to claim rights or contest
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|
||||
12. If the distribution and/or use of the Library is restricted in
|
||||
certain countries either by patents or by copyrighted interfaces, the
|
||||
original copyright holder who places the Library under this License may add
|
||||
an explicit geographical distribution limitation excluding those countries,
|
||||
so that distribution is permitted only in or among countries not thus
|
||||
excluded. In such case, this License incorporates the limitation as if
|
||||
written in the body of this License.
|
||||
|
||||
13. The Free Software Foundation may publish revised and/or new
|
||||
versions of the Lesser General Public License from time to time.
|
||||
Such new versions will be similar in spirit to the present version,
|
||||
but may differ in detail to address new problems or concerns.
|
||||
|
||||
Each version is given a distinguishing version number. If the Library
|
||||
specifies a version number of this License which applies to it and
|
||||
"any later version", you have the option of following the terms and
|
||||
conditions either of that version or of any later version published by
|
||||
the Free Software Foundation. If the Library does not specify a
|
||||
license version number, you may choose any version ever published by
|
||||
the Free Software Foundation.
|
||||
|
||||
14. If you wish to incorporate parts of the Library into other free
|
||||
programs whose distribution conditions are incompatible with these,
|
||||
write to the author to ask for permission. For software which is
|
||||
copyrighted by the Free Software Foundation, write to the Free
|
||||
Software Foundation; we sometimes make exceptions for this. Our
|
||||
decision will be guided by the two goals of preserving the free status
|
||||
of all derivatives of our free software and of promoting the sharing
|
||||
and reuse of software generally.
|
||||
|
||||
NO WARRANTY
|
||||
|
||||
15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
|
||||
WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
|
||||
EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR
|
||||
OTHER PARTIES PROVIDE THE LIBRARY "AS IS" WITHOUT WARRANTY OF ANY
|
||||
KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE
|
||||
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
|
||||
PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE
|
||||
LIBRARY IS WITH YOU. SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME
|
||||
THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
|
||||
|
||||
16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN
|
||||
WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY
|
||||
AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU
|
||||
FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR
|
||||
CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE
|
||||
LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING
|
||||
RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A
|
||||
FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
|
||||
SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
|
||||
DAMAGES.
|
||||
|
||||
END OF TERMS AND CONDITIONS
|
||||
|
|
@ -1,71 +0,0 @@
|
|||
## Process this file with automake to create Makefile.in
|
||||
##
|
||||
## $Id: Makefile.am 138 2012-04-01 20:00:09Z oparviai $
|
||||
##
|
||||
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
|
||||
##
|
||||
## SoundTouch is free software; you can redistribute it and/or modify it under the
|
||||
## terms of the GNU General Public License as published by the Free Software
|
||||
## Foundation; either version 2 of the License, or (at your option) any later
|
||||
## version.
|
||||
##
|
||||
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
|
||||
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
|
||||
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
|
||||
##
|
||||
## You should have received a copy of the GNU General Public License along with
|
||||
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
|
||||
## Place - Suite 330, Boston, MA 02111-1307, USA
|
||||
|
||||
|
||||
include $(top_srcdir)/config/am_include.mk
|
||||
|
||||
|
||||
# set to something if you want other stuff to be included in the distribution tarball
|
||||
EXTRA_DIST=SoundTouch.dsp SoundTouch.dsw SoundTouch.sln SoundTouch.vcproj
|
||||
|
||||
noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.h
|
||||
|
||||
lib_LTLIBRARIES=libSoundTouch.la
|
||||
#
|
||||
libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp BPMDetect.cpp PeakFinder.cpp
|
||||
|
||||
|
||||
# Compiler flags
|
||||
AM_CXXFLAGS=-O3 -fcheck-new -I../../include
|
||||
|
||||
# Compile the files that need MMX and SSE individually.
|
||||
libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
|
||||
noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
|
||||
libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
|
||||
libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
|
||||
|
||||
# We enable optimizations by default.
|
||||
# If MMX is supported compile with -mmmx.
|
||||
# Do not assume -msse is also supported.
|
||||
if HAVE_MMX
|
||||
libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
|
||||
else
|
||||
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
|
||||
endif
|
||||
|
||||
# We enable optimizations by default.
|
||||
# If SSE is supported compile with -msse.
|
||||
if HAVE_SSE
|
||||
libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
|
||||
else
|
||||
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
|
||||
endif
|
||||
|
||||
# Let the user disable optimizations if he wishes to.
|
||||
if !X86_OPTIMIZATIONS
|
||||
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
|
||||
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
|
||||
endif
|
||||
|
||||
|
||||
# other linking flags to add
|
||||
# noinst_LTLIBRARIES = libSoundTouchOpt.la
|
||||
# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
|
||||
# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp
|
||||
# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include
|
||||
|
|
@ -8,15 +8,13 @@
|
|||
<meta name="author" content="Olli Parviainen">
|
||||
<meta name="description"
|
||||
content="Readme file for SoundTouch audio processing library">
|
||||
<meta name="GENERATOR" content="Microsoft FrontPage 4.0">
|
||||
<meta name="ProgId" content="FrontPage.Editor.Document">
|
||||
<style> <!-- .normal { font-family: Arial }
|
||||
--></style>
|
||||
</head>
|
||||
<body class="normal">
|
||||
<hr>
|
||||
<h1>SoundTouch audio processing library v1.7.1</h1>
|
||||
<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2012 </p>
|
||||
<h1>SoundTouch audio processing library v1.9</h1>
|
||||
<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2015</p>
|
||||
<hr>
|
||||
<h2>1. Introduction </h2>
|
||||
<p>SoundTouch is an open-source audio processing library that allows
|
||||
|
|
@ -24,35 +22,31 @@ changing the sound tempo, pitch and playback rate parameters
|
|||
independently from each other, i.e.:</p>
|
||||
<ul>
|
||||
<li> Sound tempo can be increased or decreased while maintaining the
|
||||
original pitch </li>
|
||||
original pitch</li>
|
||||
<li> Sound pitch can be increased or decreased while maintaining the
|
||||
original tempo </li>
|
||||
original tempo</li>
|
||||
<li> Change playback rate that affects both tempo and pitch at the
|
||||
same time </li>
|
||||
same time</li>
|
||||
<li> Choose any combination of tempo/pitch/rate</li>
|
||||
</ul>
|
||||
<h3>1.1 Contact information </h3>
|
||||
<p>Author email: oparviai 'at' iki.fi </p>
|
||||
<p>SoundTouch WWW page: <a href="http://www.surina.net/soundtouch">http://www.surina.net/soundtouch</a></p>
|
||||
<p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p>
|
||||
<hr>
|
||||
<h2>2. Compiling SoundTouch</h2>
|
||||
<p>Before compiling, notice that you can choose the sample data format
|
||||
if it's desirable to use floating point sample data instead of 16bit
|
||||
integers. See section "sample data format" for more information.</p>
|
||||
<p>Before compiling, notice that you can choose the sample data format if it's
|
||||
desirable to use floating point sample data instead of 16bit integers. See
|
||||
section "sample data format" for more information.</p>
|
||||
<p>Also notice that SoundTouch can use OpenMP instructions for parallel
|
||||
computation to accelerate the runtime processing speed in multi-core systems,
|
||||
however, these improvements need to be separately enabled before compiling. See
|
||||
OpenMP notes in Chapter 3 below.</p>
|
||||
<h3>2.1. Building in Microsoft Windows</h3>
|
||||
<p>Project files for Microsoft Visual C++ 6.0 and Visual C++ .NET are
|
||||
supplied with the source code package.<br>
|
||||
<p>Project files for Microsoft Visual C++ are supplied with the source
|
||||
code package. Go to Microsoft WWW page to download
|
||||
<a href="http://www.visualstudio.com/en-US/products/visual-studio-express-vs">
|
||||
Microsoft Visual Studio Express version for free</a>.
|
||||
</p>
|
||||
<p> Please notice that SoundTouch library uses processor-specific
|
||||
optimizations for Pentium III and AMD processors. Visual Studio .NET
|
||||
and later versions supports the required instructions by default, but
|
||||
Visual Studio 6.0 requires a processor pack upgrade to be installed in
|
||||
order to support these optimizations. The processor pack upgrade can be
|
||||
downloaded from Microsoft site at this URL:</p>
|
||||
<p><a href="http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx">http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx</a></p>
|
||||
<p>If the above URL is unavailable or removed, go to <a
|
||||
href="http://msdn.microsoft.com/"> http://msdn.microsoft.com</a> and
|
||||
perform a search with keywords "processor pack". </p>
|
||||
<p>To build the binaries with Visual C++ compiler, either run
|
||||
"make-win.bat" script, or open the appropriate project files in source
|
||||
code directories with Visual Studio. The final executable will appear
|
||||
|
|
@ -61,6 +55,22 @@ instead of the make-win.bat script, directories bin and lib may need to
|
|||
be created manually to the SoundTouch package root for the final
|
||||
executables. The make-win.bat script creates these directories
|
||||
automatically. </p>
|
||||
<p><strong>OpenMP NOTE</strong>: If activating the OpenMP parallel computing in
|
||||
the compilation, the target program will require additional vcomp dll library to
|
||||
properly run. In Visual C++ 9.0 these libraries can be found in the following
|
||||
folders.</p>
|
||||
<ul>
|
||||
<li>x86 32bit: C:\Program Files (x86)\Microsoft Visual Studio
|
||||
9.0\VC\redist\x86\Microsoft.VC90.OPENMP\vcomp90.dll</li>
|
||||
<li>x64 64bit: C:\Program Files (x86)\Microsoft Visual Studio
|
||||
9.0\VC\redist\amd64\Microsoft.VC90.OPENMP\vcomp90.dll</li>
|
||||
</ul>
|
||||
<p>In Visual Studio 2008, a SP1 version may be required for these libraries. In
|
||||
other VC++ versions the required library will be expectedly found in similar
|
||||
"redist" location.</p>
|
||||
<p>Notice that as minor demonstration of a "dll hell" phenomenon both the 32-bit
|
||||
and 64-bit version of vcomp90.dll have the same filename but different contents,
|
||||
thus choose the proper version to allow the program start.</p>
|
||||
<h3>2.2. Building in Gnu platforms</h3>
|
||||
<p>The SoundTouch library compiles in practically any platform
|
||||
supporting GNU compiler (GCC) tools. SoundTouch requires GCC version 4.3 or later.</p>
|
||||
|
|
@ -92,7 +102,9 @@ Notice that "configure" file is not available before running the
|
|||
<pre>make -</pre>
|
||||
</td>
|
||||
<td>
|
||||
<p>Builds the SoundTouch library & SoundStretch utility.</p>
|
||||
<p>Builds the SoundTouch library & SoundStretch utility. You can
|
||||
optionally add "-j" switch after "make" to speed up the compilation in
|
||||
multi-core systems.</p>
|
||||
</td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
|
|
@ -133,7 +145,7 @@ directly and remove the following definition:<blockquote>
|
|||
<pre>#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1</pre>
|
||||
</blockquote>
|
||||
|
||||
<h4><b>2.2.3 Compiling Shared Library / DLL version</b></h4>
|
||||
<h4><b>2.2.3 Compiling Shared Library / DLL version in Cygwin</b></h4>
|
||||
<p>
|
||||
The GNU compilation does not automatically create a shared-library version of
|
||||
SoundTouch (.so or .dll). If such is desired, then you can create it as follows
|
||||
|
|
@ -147,7 +159,15 @@ sstrip SoundTouch.dll</pre>
|
|||
<h3>2.1. Building in Android</h3>
|
||||
<p>Android compilation instructions are within the
|
||||
source code package, see file "<b>source/Android-lib/README-SoundTouch-Android.html</b>"
|
||||
in the package.</p>
|
||||
in the source code package. </p>
|
||||
<p>The Android compilation automatically builds separate .so library binaries
|
||||
for ARM, X86 and MIPS processor architectures. For optimal device support,
|
||||
include all these .so library binaries into the Android .apk application
|
||||
package, so the target Android device can automatically choose the proper
|
||||
library binary version to use.</p>
|
||||
<p>The <strong>source/Android-lib</strong> folder includes also an Android
|
||||
example application that processes WAV audio files using SoundTouch library in
|
||||
Android devices.</p>
|
||||
|
||||
<hr>
|
||||
<h2>3. About implementation & Usage tips <h3>3.1. Supported sample data formats</h3>
|
||||
|
|
@ -157,7 +177,7 @@ and 32bit floating point values, the default is 32bit floating point. </p>
|
|||
"STTypes.h" by choosing one of the following defines:</p>
|
||||
<ul>
|
||||
<li> <span style="font-weight: bold;">#define
|
||||
SOUNDTOUCH_INTEGER_SAMPLES</span> for 16bit signed integer </li>
|
||||
SOUNDTOUCH_INTEGER_SAMPLES</span> for 16bit signed integer</li>
|
||||
<li> <span style="font-weight: bold;">#define </span><span
|
||||
style="font-weight: bold;">SOUNDTOUCH_</span><span
|
||||
style="font-weight: bold;">FLOAT_SAMPLES</span> for 32bit floating
|
||||
|
|
@ -183,7 +203,7 @@ the channels, which consequently would ruin the stereo effect.</p>
|
|||
<li> Input/output processing latency for the SoundTouch processor is
|
||||
around 100 ms. This is when time-stretching is used. If the rate
|
||||
transposing effect alone is used, the latency requirement is much
|
||||
shorter, see section 'About algorithms'. </li>
|
||||
shorter, see section 'About algorithms'.</li>
|
||||
<li> Processing CD-quality sound (16bit stereo sound with 44100H
|
||||
sample rate) in real-time or faster is possible starting from
|
||||
processors equivalent to Intel Pentium 133Mh or better, if using the
|
||||
|
|
@ -217,9 +237,9 @@ is around 100 ms.</p>
|
|||
to produce the tempo, pitch and rate controls:</p>
|
||||
<ul>
|
||||
<li> <strong>'Tempo'</strong> control is implemented purely by
|
||||
time-stretching. </li>
|
||||
time-stretching.</li>
|
||||
<li> <strong>'Rate</strong>' control is implemented purely by sample
|
||||
rate transposing. </li>
|
||||
rate transposing.</li>
|
||||
<li> <strong>'Pitch</strong>' control is implemented as a
|
||||
combination of time-stretching and sample rate transposing. For
|
||||
example, to increase pitch the audio stream is first time-stretched to
|
||||
|
|
@ -251,7 +271,7 @@ when increasing the tempo and vice versa. <br>
|
|||
<br>
|
||||
By default, this setting value is calculated automatically according to
|
||||
tempo value.<br>
|
||||
</li>
|
||||
</li>
|
||||
<li> <strong>DEFAULT_SEEKWINDOW_MS</strong>: The seeking window
|
||||
default length in milliseconds is for the algorithm that seeks the best
|
||||
possible overlapping location. This determines from how wide a sample
|
||||
|
|
@ -267,7 +287,7 @@ this setting.<br>
|
|||
<br>
|
||||
By default, this setting value is calculated automatically according to
|
||||
tempo value.<br>
|
||||
</li>
|
||||
</li>
|
||||
<li> <strong>DEFAULT_OVERLAP_MS</strong>: Overlap length in
|
||||
milliseconds. When the sound sequences are mixed back together to form
|
||||
again a continuous sound stream, this parameter defines how much the
|
||||
|
|
@ -343,28 +363,55 @@ function with parameter id of SETTING_USE_QUICKSEEK and value
|
|||
<p>setSetting(SETTING_USE_QUICKSEEK, 1);</p>
|
||||
</blockquote>
|
||||
<p><strong>CPU-specific optimizations:</strong></p>
|
||||
<p>Intel x86 specific SIMD optimizations are implemented using compiler
|
||||
intrinsics, providing about a 3x processing speedup for x86 compatible
|
||||
processors vs. non-SIMD implementation:</p>
|
||||
<ul>
|
||||
<li> Intel MMX optimized routines are used with compatible CPUs when
|
||||
16bit integer sample type is used. MMX optimizations are available both
|
||||
in Win32 and Gnu/x86 platforms. Compatible processors are Intel
|
||||
PentiumMMX and later; AMD K6-2, Athlon and later. </li>
|
||||
<li> Intel SSE optimized routines are used with compatible CPUs when
|
||||
floating point sample type is used. SSE optimizations are currently
|
||||
implemented for Win32 platform only. Processors compatible with SSE
|
||||
extension are Intel processors starting from Pentium-III, and AMD
|
||||
processors starting from Athlon XP. </li>
|
||||
<li> AMD 3DNow! optimized routines are used with compatible CPUs when
|
||||
floating point sample type is used, but SSE extension isn't supported .
|
||||
3DNow! optimizations are currently implemented for Win32 platform only.
|
||||
These optimizations are used in AMD K6-2 and Athlon (classic) CPU's;
|
||||
better performing SSE routines are used with AMD processor starting
|
||||
from Athlon XP. </li>
|
||||
<li> Intel MMX optimized routines are used with x86 CPUs when 16bit integer
|
||||
sample type is used</li>
|
||||
<li> Intel SSE optimized routines are used with x86 CPUs when 32bit floating
|
||||
point sample type is used</li>
|
||||
</ul>
|
||||
<h3>3.5 OpenMP parallel computation</h3>
|
||||
<p>SoundTouch 1.9 onwards support running the algorithms parallel in several CPU
|
||||
cores. Based on benchmark the experienced multi-core processing speed-up gain
|
||||
ranges between +30% (on a high-spec dual-core x86 Windows PC) to 215% (on a moderately low-spec
|
||||
quad-core ARM of Raspberry Pi2).</p>
|
||||
<p>The parallel computing support is implemented using OpenMP spec 3.0
|
||||
instructions. These instructions are supported by Visual C++ 2008 and later, and
|
||||
GCC v4.2 and later. Compilers that do not supporting OpenMP will ignore these
|
||||
optimizations and routines will still work properly. Possible warnings about
|
||||
unknown #pragmas are related to OpenMP support and can be safely ignored.</p>
|
||||
<p>The OpenMP improvements are disabled by default, and need to be enabled by
|
||||
developer during compile-time. Reason for this is that parallel processing adds
|
||||
moderate runtime overhead in managing the multi-threading, so it may not be
|
||||
necessary nor desirable in all applications. For example real-time processing
|
||||
that is not constrained by CPU power will not benefit of speed-up provided by
|
||||
the parallel processing, in the contrary it may increase power consumption due
|
||||
to the increased overhead.</p>
|
||||
<p>However, applications that run on low-spec multi-core CPUs and may otherwise
|
||||
have possibly constrained performance will benefit of the OpenMP improvements.
|
||||
This include for example multi-core embedded devices.</p>
|
||||
<p>OpenMP parallel computation can be enabled before compiling SoundTouch
|
||||
library as follows:</p>
|
||||
<ul>
|
||||
<li><strong>Visual Studio</strong>: Open properties for the <strong>SoundTouch
|
||||
</strong>sub-project, browse to <strong>C/C++</strong> and <strong>Language
|
||||
</strong>settings. Set
|
||||
there "<strong>OpenMP support</strong>" to "<strong>Yes</strong>". Alternatively add
|
||||
<strong>/openmp</strong> switch to command-line
|
||||
parameters</li>
|
||||
<li><strong>GNU</strong>: Run the configure script with "<strong>./configure
|
||||
--enable-openmp</strong>" switch, then run make as usually</li>
|
||||
<li><strong>Android</strong>: Add "<strong>-fopenmp</strong>" switches to compiler & linker
|
||||
options, see README-SoundTouch-Android.html in the source code package for
|
||||
more detailed instructions.</li>
|
||||
</ul>
|
||||
<hr>
|
||||
<h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility
|
||||
</h2>
|
||||
<p>SoundStretch audio processing utility<br>
|
||||
Copyright (c) Olli Parviainen 2002-2012</p>
|
||||
Copyright (c) Olli Parviainen 2002-2015</p>
|
||||
<p>SoundStretch is a simple command-line application that can change
|
||||
tempo, pitch and playback rates of WAV sound files. This program is
|
||||
intended primarily to demonstrate how the "SoundTouch" library can be
|
||||
|
|
@ -464,15 +511,15 @@ transposing. Gains speed but loses sound quality. </td>
|
|||
<li> To use standard input/output pipes for processing, give "stdin"
|
||||
and "stdout" as input/output filenames correspondingly. The standard
|
||||
input/output pipes will still carry the audio data in .wav audio file
|
||||
format. </li>
|
||||
format.</li>
|
||||
<li> The numerical switches allow both integer (e.g. "-tempo=123")
|
||||
and decimal (e.g. "-tempo=123.45") numbers. </li>
|
||||
and decimal (e.g. "-tempo=123.45") numbers.</li>
|
||||
<li> The "-naa" and/or "-quick" switches can be used to reduce CPU
|
||||
usage while compromising some sound quality </li>
|
||||
usage while compromising some sound quality</li>
|
||||
<li> The BPM detection algorithm works by detecting repeating bass or
|
||||
drum patterns at low frequencies of <250Hz. A lower-than-expected
|
||||
BPM figure may be reported for music with uneven or complex bass
|
||||
patterns. </li>
|
||||
patterns.</li>
|
||||
</ul>
|
||||
<h3>4.2. SoundStretch usage examples </h3>
|
||||
<p><strong>Example 1</strong></p>
|
||||
|
|
@ -510,9 +557,42 @@ and estimates the BPM rate:</p>
|
|||
<blockquote>
|
||||
<pre>soundstretch stdin -bpm</pre>
|
||||
</blockquote>
|
||||
<p><strong>Example 6</strong></p>
|
||||
<p>The following command tunes song from original 440Hz tuning to 432Hz tuning:
|
||||
this corresponds to lowering the pitch by -0.318 semitones:</p>
|
||||
<blockquote>
|
||||
<pre>soundstretch original.wav output.wav -pitch=-0.318</pre>
|
||||
</blockquote>
|
||||
<hr>
|
||||
<h2>5. Change History</h2>
|
||||
<h3>5.1. SoundTouch library Change History </h3>
|
||||
<p><b>1.9:</b></p>
|
||||
<ul>
|
||||
<li>Added support for parallel computation support via OpenMP primitives for better performance in multicore systems.
|
||||
Benchmarks show that achieved parallel processing speedup improvement
|
||||
typically range from +30% (x86 dual-core) to +180% (ARM quad-core). The
|
||||
OpenMP optimizations are disabled by default, see OpenMP notes above in this
|
||||
readme file how to enabled these optimizations.</li>
|
||||
<li>Android: Added support for Android devices featuring X86 and MIPS CPUs,
|
||||
in addition to ARM CPUs.</li>
|
||||
<li>Android: More versatile Android example application that processes WAV
|
||||
audio files with SoundTouch library</li>
|
||||
<li>Replaced Windows-like 'BOOL' types with native 'bool'</li>
|
||||
<li>Changed documentation token to "dist_doc_DATA" in Makefile.am file</li>
|
||||
<li>Miscellaneous small fixes and improvements</li>
|
||||
</ul>
|
||||
<p><b>1.8.0:</b></p>
|
||||
<ul>
|
||||
<li>Added support for multi-channel audio processing</li>
|
||||
<li>Added support for <b>cubic</b> and <b>shannon</b> interpolation for rate and pitch shift effects besides
|
||||
the original <b>linear</b> interpolation, to reduce aliasing at high frequencies due to interpolation.
|
||||
Cubic interpolation is used as default for floating point processing, and linear interpolation for integer
|
||||
processing.</li>
|
||||
<li>Fixed bug in anti-alias filtering that limited stop-band attenuation to -10 dB instead of <-50dB, and
|
||||
increased filter length from 32 to 64 taps to further reduce aliasing due to frequency folding.</li>
|
||||
<li>Performance improvements in cross-correlation algorithm</li>
|
||||
<li>Other bug and compatibility fixes</li>
|
||||
</ul>
|
||||
<p><b>1.7.1:</b></p>
|
||||
<ul>
|
||||
<li>Added files for Android compilation
|
||||
|
|
@ -532,7 +612,7 @@ and estimates the BPM rate:</p>
|
|||
<p><b>1.6.0:</b></p>
|
||||
<ul>
|
||||
<li> Added automatic cutoff threshold adaptation to beat detection
|
||||
routine to better adapt BPM calculation to different types of music </li>
|
||||
routine to better adapt BPM calculation to different types of music</li>
|
||||
<li> Retired 3DNow! optimization support as 3DNow! is nowadays
|
||||
obsoleted and assembler code is nuisance to maintain</li>
|
||||
<li>Retired "configure" file from source code package due to
|
||||
|
|
@ -542,7 +622,7 @@ toolchain version for generating the "configure" file</li>
|
|||
<li>Resolved namespace/label naming conflicts with other libraries by
|
||||
replacing global labels such as INTEGER_SAMPLES with more specific
|
||||
SOUNDTOUCH_INTEGER_SAMPLES etc.<br>
|
||||
</li>
|
||||
</li>
|
||||
<li>Updated windows build scripts & project files for Visual
|
||||
Studio 2008 support</li>
|
||||
<li> Updated SoundTouch.dll API for .NET compatibility</li>
|
||||
|
|
@ -552,22 +632,22 @@ sample batch sizes</li>
|
|||
<p><strong>1.5.0:</strong></p>
|
||||
<ul>
|
||||
<li> Added normalization to correlation calculation and improvement
|
||||
automatic seek/sequence parameter calculation to improve sound quality </li>
|
||||
automatic seek/sequence parameter calculation to improve sound quality</li>
|
||||
<li> Bugfixes:
|
||||
<ul>
|
||||
<li> Fixed negative array indexing in quick seek algorithm </li>
|
||||
<li> FIR autoalias filter running too far in processing buffer </li>
|
||||
<li> Check against zero sample count in rate transposing </li>
|
||||
<li> Fixed negative array indexing in quick seek algorithm</li>
|
||||
<li> FIR autoalias filter running too far in processing buffer</li>
|
||||
<li> Check against zero sample count in rate transposing</li>
|
||||
<li> Fix for x86-64 support: Removed pop/push instructions from
|
||||
the cpu detection algorithm. </li>
|
||||
<li> Check against empty buffers in FIFOSampleBuffer </li>
|
||||
the cpu detection algorithm. </li>
|
||||
<li> Check against empty buffers in FIFOSampleBuffer</li>
|
||||
<li> Other minor fixes & code cleanup</li>
|
||||
</ul>
|
||||
</li>
|
||||
<li> Fixes in compilation scripts for non-Intel platforms </li>
|
||||
</li>
|
||||
<li> Fixes in compilation scripts for non-Intel platforms</li>
|
||||
<li> Added Dynamic-Link-Library (DLL) version of SoundTouch library
|
||||
build, provided with Delphi/Pascal wrapper for calling the dll routines
|
||||
</li>
|
||||
</li>
|
||||
<li> Added #define PREVENT_CLICK_AT_RATE_CROSSOVER that prevents a
|
||||
click artifact when crossing the nominal pitch from either positive to
|
||||
negative side or vice versa</li>
|
||||
|
|
@ -580,86 +660,91 @@ processing more than 2048 samples at one call </li>
|
|||
<p><strong>1.4.0:</strong></p>
|
||||
<ul>
|
||||
<li> Improved sound quality by automatic calculation of time stretch
|
||||
algorithm processing parameters according to tempo setting </li>
|
||||
algorithm processing parameters according to tempo setting</li>
|
||||
<li> Moved BPM detection routines from SoundStretch application into
|
||||
SoundTouch library </li>
|
||||
SoundTouch library</li>
|
||||
<li> Bugfixes: Usage of uninitialied variables, GNU build scripts,
|
||||
compiler errors due to 'const' keyword mismatch. </li>
|
||||
compiler errors due to 'const' keyword mismatch.</li>
|
||||
<li> Source code cleanup</li>
|
||||
</ul>
|
||||
<p><strong>1.3.1: </strong> </p>
|
||||
<ul>
|
||||
<li> Changed static class declaration to GCC 4.x compiler compatible
|
||||
syntax. </li>
|
||||
syntax.</li>
|
||||
<li> Enabled MMX/SSE-optimized routines also for GCC compilers.
|
||||
Earlier the MMX/SSE-optimized routines were written in
|
||||
compiler-specific inline assembler, now these routines are migrated to
|
||||
use compiler intrinsic syntax which allows compiling the same
|
||||
MMX/SSE-optimized source code with both Visual C++ and GCC compilers. </li>
|
||||
MMX/SSE-optimized source code with both Visual C++ and GCC compilers.</li>
|
||||
<li> Set floating point as the default sample format and added switch
|
||||
to the GNU configure script for selecting the other sample format.</li>
|
||||
</ul>
|
||||
<p><strong>1.3.0: </strong> </p>
|
||||
<ul>
|
||||
<li> Fixed tempo routine output duration inaccuracy due to rounding
|
||||
error </li>
|
||||
error</li>
|
||||
<li> Implemented separate processing routines for integer and
|
||||
floating arithmetic to allow improvements to floating point routines
|
||||
(earlier used algorithms mostly optimized for integer arithmetic also
|
||||
for floating point samples) </li>
|
||||
for floating point samples)</li>
|
||||
<li> Fixed a bug that distorts sound if sample rate changes during
|
||||
the sound stream </li>
|
||||
the sound stream</li>
|
||||
<li> Fixed a memory leak that appeared in MMX/SSE/3DNow! optimized
|
||||
routines </li>
|
||||
routines</li>
|
||||
<li> Reduced redundant code pieces in MMX/SSE/3DNow! optimized
|
||||
routines vs. the standard C routines. </li>
|
||||
<li> MMX routine incompatibility with new gcc compiler versions </li>
|
||||
<li> Other miscellaneous bug fixes </li>
|
||||
routines vs. the standard C routines.</li>
|
||||
<li> MMX routine incompatibility with new gcc compiler versions</li>
|
||||
<li> Other miscellaneous bug fixes</li>
|
||||
</ul>
|
||||
<p><strong>1.2.1: </strong> </p>
|
||||
<ul>
|
||||
<li> Added automake/autoconf scripts for GNU platforms (in courtesy
|
||||
of David Durham) </li>
|
||||
<li> Fixed SCALE overflow bug in rate transposer routine. </li>
|
||||
<li> Fixed 64bit address space bugs. </li>
|
||||
of David Durham)</li>
|
||||
<li> Fixed SCALE overflow bug in rate transposer routine.</li>
|
||||
<li> Fixed 64bit address space bugs.</li>
|
||||
<li> Created a 'soundtouch' namespace for SAMPLETYPE definitions.</li>
|
||||
</ul>
|
||||
<p><strong>1.2.0: </strong> </p>
|
||||
<ul>
|
||||
<li> Added support for 32bit floating point sample data type with
|
||||
SSE/3DNow! optimizations for Win32 platform (SSE/3DNow! optimizations
|
||||
currently not supported in GCC environment) </li>
|
||||
currently not supported in GCC environment)</li>
|
||||
<li> Replaced 'make-gcc' script for GNU environment by master
|
||||
Makefile </li>
|
||||
Makefile</li>
|
||||
<li> Added time-stretch routine configurability to SoundTouch main
|
||||
class </li>
|
||||
class</li>
|
||||
<li> Bugfixes</li>
|
||||
</ul>
|
||||
<p><strong>1.1.1: </strong> </p>
|
||||
<ul>
|
||||
<li> Moved SoundTouch under lesser GPL license (LGPL). This allows
|
||||
using SoundTouch library in programs that aren't released under GPL
|
||||
license. </li>
|
||||
license.</li>
|
||||
<li> Changed MMX routine organiation so that MMX optimized routines
|
||||
are now implemented in classes that are derived from the basic classes
|
||||
having the standard non-mmx routines. </li>
|
||||
<li> MMX routines to support gcc version 3. </li>
|
||||
<li> Replaced windows makefiles by script using the .dsw files </li>
|
||||
having the standard non-mmx routines.</li>
|
||||
<li> MMX routines to support gcc version 3.</li>
|
||||
<li> Replaced windows makefiles by script using the .dsw files</li>
|
||||
</ul>
|
||||
<p><strong>1.0.1: </strong> </p>
|
||||
<ul>
|
||||
<li> "mmx_gcc.cpp": Added "using namespace std" and removed "return
|
||||
0" from a function with void return value to fix compiler errors when
|
||||
compiling the library in Solaris environment. </li>
|
||||
compiling the library in Solaris environment.</li>
|
||||
<li> Moved file "FIFOSampleBuffer.h" to "include" directory to allow
|
||||
accessing the FIFOSampleBuffer class from external files. </li>
|
||||
accessing the FIFOSampleBuffer class from external files.</li>
|
||||
</ul>
|
||||
<p><strong>1.0: </strong> </p>
|
||||
<ul>
|
||||
<li> Initial release </li>
|
||||
<li> Initial release</li>
|
||||
</ul>
|
||||
<p> </p>
|
||||
<h3>5.2. SoundStretch application Change History </h3>
|
||||
<p><b>1.9:</b></p>
|
||||
<ul>
|
||||
<li>Added support for WAV file 'fact' information chunk.</li>
|
||||
</ul>
|
||||
|
||||
<p><b>1.7.0:</b></p>
|
||||
<ul>
|
||||
<li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer
|
||||
|
|
@ -674,14 +759,14 @@ music processing.</li>
|
|||
<p><strong>1.4.0:</strong></p>
|
||||
<ul>
|
||||
<li> Moved BPM detection routines from SoundStretch application into
|
||||
SoundTouch library </li>
|
||||
SoundTouch library</li>
|
||||
<li> Allow using standard input/output pipes as audio processing
|
||||
input/output streams</li>
|
||||
</ul>
|
||||
<p><strong>1.3.0:</strong></p>
|
||||
<ul>
|
||||
<li> Simplified accessing WAV files with floating point sample
|
||||
format. </li>
|
||||
format.</li>
|
||||
</ul>
|
||||
<p><strong>1.2.1: </strong> </p>
|
||||
<ul>
|
||||
|
|
@ -689,69 +774,66 @@ format. </li>
|
|||
</ul>
|
||||
<p><strong>1.2.0: </strong> </p>
|
||||
<ul>
|
||||
<li> Added support for 32bit floating point sample data type </li>
|
||||
<li> Restructured the BPM routines into separate library </li>
|
||||
<li> Added support for 32bit floating point sample data type</li>
|
||||
<li> Restructured the BPM routines into separate library</li>
|
||||
<li> Fixed big-endian conversion bugs in WAV file routines (hopefully
|
||||
:)</li>
|
||||
</ul>
|
||||
<p><strong>1.1.1: </strong> </p>
|
||||
<ul>
|
||||
<li> Fixed bugs in WAV file reading & added byte-order conversion
|
||||
for big-endian processors. </li>
|
||||
for big-endian processors.</li>
|
||||
<li> Moved SoundStretch source code under 'example' directory to
|
||||
highlight difference from SoundTouch stuff. </li>
|
||||
<li> Replaced windows makefiles by script using the .dsw files </li>
|
||||
highlight difference from SoundTouch stuff.</li>
|
||||
<li> Replaced windows makefiles by script using the .dsw files</li>
|
||||
<li> Output file name isn't required if output isn't desired (e.g. if
|
||||
using the switch '-bpm' in plain format only) </li>
|
||||
using the switch '-bpm' in plain format only)</li>
|
||||
</ul>
|
||||
<p><strong>1.1:</strong></p>
|
||||
<ul>
|
||||
<li> Fixed "Release" settings in Microsoft Visual C++ project file
|
||||
(.dsp) </li>
|
||||
(.dsp)</li>
|
||||
<li> Added beats-per-minute (BPM) detection routine and command-line
|
||||
switch "-bpm" </li>
|
||||
switch "-bpm"</li>
|
||||
</ul>
|
||||
<p><strong>1.01: </strong> </p>
|
||||
<ul>
|
||||
<li> Initial release </li>
|
||||
<li> Initial release</li>
|
||||
</ul>
|
||||
<hr>
|
||||
<h2>6. Acknowledgements </h2>
|
||||
<p>Kudos for these people who have contributed to development or
|
||||
submitted bugfixes since SoundTouch v1.3.1: </p>
|
||||
submitted bugfixes:</p>
|
||||
<ul>
|
||||
<li> Arthur A</li>
|
||||
<li> Arthur A</li>
|
||||
<li> Richard Ash</li>
|
||||
<li> Stanislav Brabec</li>
|
||||
<li> Christian Budde</li>
|
||||
<li> Chris Bryan</li>
|
||||
<li> Jacek Caban</li>
|
||||
<li> Brian Cameron</li>
|
||||
<li> Jason Champion</li>
|
||||
<li> David Clark</li>
|
||||
<li> Patrick Colis</li>
|
||||
<li> Miquel Colon</li>
|
||||
<li> Jim Credland</li>
|
||||
<li> Sandro Cumerlato</li>
|
||||
<li> Justin Frankel</li>
|
||||
<li> Masa H.</li>
|
||||
<li> Jason Garland</li>
|
||||
<li> Takashi Iwai</li>
|
||||
<li> Thomas Klausner</li>
|
||||
<li> Mathias Möhl</li>
|
||||
<li> Yuval Naveh</li>
|
||||
<li> Paulo Pizarro</li>
|
||||
<li> Blaise Potard</li>
|
||||
<li> RJ Ryan</li>
|
||||
<li> Patrick Colis </li>
|
||||
<li> Miquel Colon </li>
|
||||
<li> Sandro Cumerlato</li>
|
||||
<li> Justin Frankel </li>
|
||||
<li> Jason Garland </li>
|
||||
<li> Takashi Iwai </li>
|
||||
<li> Mathias Möhl</li>
|
||||
<li> Yuval Naveh </li>
|
||||
<li> Paulo Pizarro </li>
|
||||
<li> Blaise Potard</li>
|
||||
<li> RJ Ryan </li>
|
||||
<li> John Sheehy</li>
|
||||
<li> Michael Pruett</li>
|
||||
<li> Rajeev Puran</li>
|
||||
<li> RJ Ryan</li>
|
||||
<li> John Sheehy</li>
|
||||
<li> Tim Shuttleworth</li>
|
||||
<li> Albert Sirvent</li>
|
||||
<li> John Stumpo</li>
|
||||
<li> Tim Shuttleworth</li>
|
||||
<li> Katja Vetter</li>
|
||||
</ul>
|
||||
<p>Moral greetings to all other contributors and users also!</p>
|
||||
|
|
@ -770,8 +852,8 @@ General Public License for more details.</p>
|
|||
License along with this library; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA</p>
|
||||
<hr><!--
|
||||
$Id: README.html 168 2012-12-28 20:55:19Z oparviai $
-->
|
||||
$Id: README.html 220 2015-05-18 17:39:26Z oparviai $
-->
|
||||
<p>
|
||||
<i>RREADME.html file updated on 28-Dec-2012</i></p>
|
||||
<i>README.html file updated in May-2015</i></p>
|
||||
</body>
|
||||
</html>
|
||||
|
|
|
|||
|
|
@ -1,626 +0,0 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Sample rate transposer. Changes sample rate by using linear interpolation
|
||||
/// together with anti-alias filtering (first order interpolation with anti-
|
||||
/// alias filtering should be quite adequate for this application)
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <memory.h>
|
||||
#include <assert.h>
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include "RateTransposer.h"
|
||||
#include "AAFilter.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
|
||||
/// A linear samplerate transposer class that uses integer arithmetics.
|
||||
/// for the transposing.
|
||||
class RateTransposerInteger : public RateTransposer
|
||||
{
|
||||
protected:
|
||||
int iSlopeCount;
|
||||
int iRate;
|
||||
SAMPLETYPE sPrevSampleL, sPrevSampleR;
|
||||
|
||||
virtual void resetRegisters();
|
||||
|
||||
virtual uint transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
virtual uint transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
public:
|
||||
RateTransposerInteger();
|
||||
virtual ~RateTransposerInteger();
|
||||
|
||||
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
|
||||
/// rate, larger faster rates.
|
||||
virtual void setRate(float newRate);
|
||||
|
||||
};
|
||||
|
||||
|
||||
/// A linear samplerate transposer class that uses floating point arithmetics
|
||||
/// for the transposing.
|
||||
class RateTransposerFloat : public RateTransposer
|
||||
{
|
||||
protected:
|
||||
float fSlopeCount;
|
||||
SAMPLETYPE sPrevSampleL, sPrevSampleR;
|
||||
|
||||
virtual void resetRegisters();
|
||||
|
||||
virtual uint transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
virtual uint transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
public:
|
||||
RateTransposerFloat();
|
||||
virtual ~RateTransposerFloat();
|
||||
};
|
||||
|
||||
|
||||
|
||||
|
||||
// Operator 'new' is overloaded so that it automatically creates a suitable instance
|
||||
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
|
||||
void * RateTransposer::operator new(size_t s)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
|
||||
return newInstance();
|
||||
}
|
||||
|
||||
|
||||
RateTransposer *RateTransposer::newInstance()
|
||||
{
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
return ::new RateTransposerInteger;
|
||||
#else
|
||||
return ::new RateTransposerFloat;
|
||||
#endif
|
||||
}
|
||||
|
||||
|
||||
// Constructor
|
||||
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
|
||||
{
|
||||
numChannels = 2;
|
||||
bUseAAFilter = TRUE;
|
||||
fRate = 0;
|
||||
|
||||
// Instantiates the anti-alias filter with default tap length
|
||||
// of 32
|
||||
pAAFilter = new AAFilter(32);
|
||||
}
|
||||
|
||||
|
||||
|
||||
RateTransposer::~RateTransposer()
|
||||
{
|
||||
delete pAAFilter;
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
|
||||
void RateTransposer::enableAAFilter(BOOL newMode)
|
||||
{
|
||||
bUseAAFilter = newMode;
|
||||
}
|
||||
|
||||
|
||||
/// Returns nonzero if anti-alias filter is enabled.
|
||||
BOOL RateTransposer::isAAFilterEnabled() const
|
||||
{
|
||||
return bUseAAFilter;
|
||||
}
|
||||
|
||||
|
||||
AAFilter *RateTransposer::getAAFilter()
|
||||
{
|
||||
return pAAFilter;
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void RateTransposer::setRate(float newRate)
|
||||
{
|
||||
double fCutoff;
|
||||
|
||||
fRate = newRate;
|
||||
|
||||
// design a new anti-alias filter
|
||||
if (newRate > 1.0f)
|
||||
{
|
||||
fCutoff = 0.5f / newRate;
|
||||
}
|
||||
else
|
||||
{
|
||||
fCutoff = 0.5f * newRate;
|
||||
}
|
||||
pAAFilter->setCutoffFreq(fCutoff);
|
||||
}
|
||||
|
||||
|
||||
// Outputs as many samples of the 'outputBuffer' as possible, and if there's
|
||||
// any room left, outputs also as many of the incoming samples as possible.
|
||||
// The goal is to drive the outputBuffer empty.
|
||||
//
|
||||
// It's allowed for 'output' and 'input' parameters to point to the same
|
||||
// memory position.
|
||||
/*
|
||||
void RateTransposer::flushStoreBuffer()
|
||||
{
|
||||
if (storeBuffer.isEmpty()) return;
|
||||
|
||||
outputBuffer.moveSamples(storeBuffer);
|
||||
}
|
||||
*/
|
||||
|
||||
|
||||
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
|
||||
// the input of the object.
|
||||
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
||||
{
|
||||
processSamples(samples, nSamples);
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Transposes up the sample rate, causing the observed playback 'rate' of the
|
||||
// sound to decrease
|
||||
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count, sizeTemp, num;
|
||||
|
||||
// If the parameter 'uRate' value is smaller than 'SCALE', first transpose
|
||||
// the samples and then apply the anti-alias filter to remove aliasing.
|
||||
|
||||
// First check that there's enough room in 'storeBuffer'
|
||||
// (+16 is to reserve some slack in the destination buffer)
|
||||
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
|
||||
|
||||
// Transpose the samples, store the result into the end of "storeBuffer"
|
||||
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
|
||||
storeBuffer.putSamples(count);
|
||||
|
||||
// Apply the anti-alias filter to samples in "store output", output the
|
||||
// result to "dest"
|
||||
num = storeBuffer.numSamples();
|
||||
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
|
||||
storeBuffer.ptrBegin(), num, (uint)numChannels);
|
||||
outputBuffer.putSamples(count);
|
||||
|
||||
// Remove the processed samples from "storeBuffer"
|
||||
storeBuffer.receiveSamples(count);
|
||||
}
|
||||
|
||||
|
||||
// Transposes down the sample rate, causing the observed playback 'rate' of the
|
||||
// sound to increase
|
||||
void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count, sizeTemp;
|
||||
|
||||
// If the parameter 'uRate' value is larger than 'SCALE', first apply the
|
||||
// anti-alias filter to remove high frequencies (prevent them from folding
|
||||
// over the lover frequencies), then transpose.
|
||||
|
||||
// Add the new samples to the end of the storeBuffer
|
||||
storeBuffer.putSamples(src, nSamples);
|
||||
|
||||
// Anti-alias filter the samples to prevent folding and output the filtered
|
||||
// data to tempBuffer. Note : because of the FIR filter length, the
|
||||
// filtering routine takes in 'filter_length' more samples than it outputs.
|
||||
assert(tempBuffer.isEmpty());
|
||||
sizeTemp = storeBuffer.numSamples();
|
||||
|
||||
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
|
||||
storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
|
||||
|
||||
if (count == 0) return;
|
||||
|
||||
// Remove the filtered samples from 'storeBuffer'
|
||||
storeBuffer.receiveSamples(count);
|
||||
|
||||
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
|
||||
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
|
||||
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
|
||||
outputBuffer.putSamples(count);
|
||||
}
|
||||
|
||||
|
||||
// Transposes sample rate by applying anti-alias filter to prevent folding.
|
||||
// Returns amount of samples returned in the "dest" buffer.
|
||||
// The maximum amount of samples that can be returned at a time is set by
|
||||
// the 'set_returnBuffer_size' function.
|
||||
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count;
|
||||
uint sizeReq;
|
||||
|
||||
if (nSamples == 0) return;
|
||||
assert(pAAFilter);
|
||||
|
||||
// If anti-alias filter is turned off, simply transpose without applying
|
||||
// the filter
|
||||
if (bUseAAFilter == FALSE)
|
||||
{
|
||||
sizeReq = (uint)((float)nSamples / fRate + 1.0f);
|
||||
count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
|
||||
outputBuffer.putSamples(count);
|
||||
return;
|
||||
}
|
||||
|
||||
// Transpose with anti-alias filter
|
||||
if (fRate < 1.0f)
|
||||
{
|
||||
upsample(src, nSamples);
|
||||
}
|
||||
else
|
||||
{
|
||||
downsample(src, nSamples);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// Returns the number of samples returned in the "dest" buffer
|
||||
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
if (numChannels == 2)
|
||||
{
|
||||
return transposeStereo(dest, src, nSamples);
|
||||
}
|
||||
else
|
||||
{
|
||||
return transposeMono(dest, src, nSamples);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Sets the number of channels, 1 = mono, 2 = stereo
|
||||
void RateTransposer::setChannels(int nChannels)
|
||||
{
|
||||
assert(nChannels > 0);
|
||||
if (numChannels == nChannels) return;
|
||||
|
||||
assert(nChannels == 1 || nChannels == 2);
|
||||
numChannels = nChannels;
|
||||
|
||||
storeBuffer.setChannels(numChannels);
|
||||
tempBuffer.setChannels(numChannels);
|
||||
outputBuffer.setChannels(numChannels);
|
||||
|
||||
// Inits the linear interpolation registers
|
||||
resetRegisters();
|
||||
}
|
||||
|
||||
|
||||
// Clears all the samples in the object
|
||||
void RateTransposer::clear()
|
||||
{
|
||||
outputBuffer.clear();
|
||||
storeBuffer.clear();
|
||||
}
|
||||
|
||||
|
||||
// Returns nonzero if there aren't any samples available for outputting.
|
||||
int RateTransposer::isEmpty() const
|
||||
{
|
||||
int res;
|
||||
|
||||
res = FIFOProcessor::isEmpty();
|
||||
if (res == 0) return 0;
|
||||
return storeBuffer.isEmpty();
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// RateTransposerInteger - integer arithmetic implementation
|
||||
//
|
||||
|
||||
/// fixed-point interpolation routine precision
|
||||
#define SCALE 65536
|
||||
|
||||
// Constructor
|
||||
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
RateTransposerInteger::resetRegisters();
|
||||
RateTransposerInteger::setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
RateTransposerInteger::~RateTransposerInteger()
|
||||
{
|
||||
}
|
||||
|
||||
|
||||
void RateTransposerInteger::resetRegisters()
|
||||
{
|
||||
iSlopeCount = 0;
|
||||
sPrevSampleL =
|
||||
sPrevSampleR = 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
unsigned int i, used;
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
|
||||
if (nSamples == 0) return 0; // no samples, no work
|
||||
|
||||
used = 0;
|
||||
i = 0;
|
||||
|
||||
// Process the last sample saved from the previous call first...
|
||||
while (iSlopeCount <= SCALE)
|
||||
{
|
||||
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
|
||||
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
|
||||
dest[i] = (SAMPLETYPE)(temp / SCALE);
|
||||
i++;
|
||||
iSlopeCount += iRate;
|
||||
}
|
||||
// now always (iSlopeCount > SCALE)
|
||||
iSlopeCount -= SCALE;
|
||||
|
||||
while (1)
|
||||
{
|
||||
while (iSlopeCount > SCALE)
|
||||
{
|
||||
iSlopeCount -= SCALE;
|
||||
used ++;
|
||||
if (used >= nSamples - 1) goto end;
|
||||
}
|
||||
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
|
||||
temp = src[used] * vol1 + iSlopeCount * src[used + 1];
|
||||
dest[i] = (SAMPLETYPE)(temp / SCALE);
|
||||
|
||||
i++;
|
||||
iSlopeCount += iRate;
|
||||
}
|
||||
end:
|
||||
// Store the last sample for the next round
|
||||
sPrevSampleL = src[nSamples - 1];
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Stereo' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
unsigned int srcPos, i, used;
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
|
||||
if (nSamples == 0) return 0; // no samples, no work
|
||||
|
||||
used = 0;
|
||||
i = 0;
|
||||
|
||||
// Process the last sample saved from the sPrevSampleLious call first...
|
||||
while (iSlopeCount <= SCALE)
|
||||
{
|
||||
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
|
||||
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
|
||||
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
|
||||
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
|
||||
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
|
||||
i++;
|
||||
iSlopeCount += iRate;
|
||||
}
|
||||
// now always (iSlopeCount > SCALE)
|
||||
iSlopeCount -= SCALE;
|
||||
|
||||
while (1)
|
||||
{
|
||||
while (iSlopeCount > SCALE)
|
||||
{
|
||||
iSlopeCount -= SCALE;
|
||||
used ++;
|
||||
if (used >= nSamples - 1) goto end;
|
||||
}
|
||||
srcPos = 2 * used;
|
||||
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
|
||||
temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
|
||||
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
|
||||
temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
|
||||
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
|
||||
|
||||
i++;
|
||||
iSlopeCount += iRate;
|
||||
}
|
||||
end:
|
||||
// Store the last sample for the next round
|
||||
sPrevSampleL = src[2 * nSamples - 2];
|
||||
sPrevSampleR = src[2 * nSamples - 1];
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void RateTransposerInteger::setRate(float newRate)
|
||||
{
|
||||
iRate = (int)(newRate * SCALE + 0.5f);
|
||||
RateTransposer::setRate(newRate);
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// RateTransposerFloat - floating point arithmetic implementation
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Constructor
|
||||
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
RateTransposerFloat::resetRegisters();
|
||||
RateTransposerFloat::setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
RateTransposerFloat::~RateTransposerFloat()
|
||||
{
|
||||
}
|
||||
|
||||
|
||||
void RateTransposerFloat::resetRegisters()
|
||||
{
|
||||
fSlopeCount = 0;
|
||||
sPrevSampleL =
|
||||
sPrevSampleR = 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
unsigned int i, used;
|
||||
|
||||
used = 0;
|
||||
i = 0;
|
||||
|
||||
// Process the last sample saved from the previous call first...
|
||||
while (fSlopeCount <= 1.0f)
|
||||
{
|
||||
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
}
|
||||
fSlopeCount -= 1.0f;
|
||||
|
||||
if (nSamples > 1)
|
||||
{
|
||||
while (1)
|
||||
{
|
||||
while (fSlopeCount > 1.0f)
|
||||
{
|
||||
fSlopeCount -= 1.0f;
|
||||
used ++;
|
||||
if (used >= nSamples - 1) goto end;
|
||||
}
|
||||
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
}
|
||||
}
|
||||
end:
|
||||
// Store the last sample for the next round
|
||||
sPrevSampleL = src[nSamples - 1];
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
unsigned int srcPos, i, used;
|
||||
|
||||
if (nSamples == 0) return 0; // no samples, no work
|
||||
|
||||
used = 0;
|
||||
i = 0;
|
||||
|
||||
// Process the last sample saved from the sPrevSampleLious call first...
|
||||
while (fSlopeCount <= 1.0f)
|
||||
{
|
||||
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
|
||||
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
}
|
||||
// now always (iSlopeCount > 1.0f)
|
||||
fSlopeCount -= 1.0f;
|
||||
|
||||
if (nSamples > 1)
|
||||
{
|
||||
while (1)
|
||||
{
|
||||
while (fSlopeCount > 1.0f)
|
||||
{
|
||||
fSlopeCount -= 1.0f;
|
||||
used ++;
|
||||
if (used >= nSamples - 1) goto end;
|
||||
}
|
||||
srcPos = 2 * used;
|
||||
|
||||
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
|
||||
+ fSlopeCount * src[srcPos + 2]);
|
||||
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
|
||||
+ fSlopeCount * src[srcPos + 3]);
|
||||
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
}
|
||||
}
|
||||
end:
|
||||
// Store the last sample for the next round
|
||||
sPrevSampleL = src[2 * nSamples - 2];
|
||||
sPrevSampleR = src[2 * nSamples - 1];
|
||||
|
||||
return i;
|
||||
}
|
||||
|
|
@ -5,112 +5,88 @@
|
|||
<Configuration>Debug</Configuration>
|
||||
<Platform>Win32</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Debug|x64">
|
||||
<Configuration>Debug</Configuration>
|
||||
<Platform>x64</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Devel|Win32">
|
||||
<Configuration>Devel</Configuration>
|
||||
<Platform>Win32</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Devel|x64">
|
||||
<Configuration>Devel</Configuration>
|
||||
<Platform>x64</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Release|Win32">
|
||||
<Configuration>Release</Configuration>
|
||||
<Platform>Win32</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Release|x64">
|
||||
<Configuration>Release</Configuration>
|
||||
<Platform>x64</Platform>
|
||||
</ProjectConfiguration>
|
||||
</ItemGroup>
|
||||
<PropertyGroup Label="Globals">
|
||||
<ProjectGuid>{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}</ProjectGuid>
|
||||
<RootNamespace>SoundTouch</RootNamespace>
|
||||
<ProjectName>SoundTouch</ProjectName>
|
||||
<RootNamespace>SoundTouch</RootNamespace>
|
||||
</PropertyGroup>
|
||||
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
|
||||
<PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'" Label="Configuration">
|
||||
<PropertyGroup Label="Configuration">
|
||||
<ConfigurationType>StaticLibrary</ConfigurationType>
|
||||
<CharacterSet>MultiByte</CharacterSet>
|
||||
<WholeProgramOptimization>true</WholeProgramOptimization>
|
||||
<PlatformToolset>$(DefaultPlatformToolset)_xp</PlatformToolset>
|
||||
</PropertyGroup>
|
||||
<PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
|
||||
<ConfigurationType>StaticLibrary</ConfigurationType>
|
||||
<CharacterSet>MultiByte</CharacterSet>
|
||||
<WholeProgramOptimization>true</WholeProgramOptimization>
|
||||
<PlatformToolset>$(DefaultPlatformToolset)_xp</PlatformToolset>
|
||||
</PropertyGroup>
|
||||
<PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
|
||||
<ConfigurationType>StaticLibrary</ConfigurationType>
|
||||
<CharacterSet>MultiByte</CharacterSet>
|
||||
<PlatformToolset>$(DefaultPlatformToolset)_xp</PlatformToolset>
|
||||
<UseDebugLibraries Condition="'$(Configuration)'=='Debug'">true</UseDebugLibraries>
|
||||
<UseDebugLibraries Condition="'$(Configuration)'!='Debug'">false</UseDebugLibraries>
|
||||
<WholeProgramOptimization Condition="'$(Configuration)'=='Release'">true</WholeProgramOptimization>
|
||||
</PropertyGroup>
|
||||
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
|
||||
<ImportGroup Label="ExtensionSettings">
|
||||
</ImportGroup>
|
||||
<ImportGroup Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'" Label="PropertySheets">
|
||||
<ImportGroup Label="PropertySheets">
|
||||
<Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
|
||||
<Import Project="..\DefaultProjectRootDir.props" />
|
||||
<Import Project="..\3rdparty.props" />
|
||||
<Import Project="..\..\common\vsprops\CodeGen_Devel.props" />
|
||||
<Import Project="..\..\common\vsprops\IncrementalLinking.props" />
|
||||
<Import Condition="'$(Configuration)'!='Release'" Project="..\..\common\vsprops\IncrementalLinking.props" />
|
||||
<Import Condition="'$(Configuration)'=='Debug'" Project="..\..\common\vsprops\CodeGen_Debug.props" />
|
||||
<Import Condition="'$(Configuration)'=='Devel'" Project="..\..\common\vsprops\CodeGen_Devel.props" />
|
||||
<Import Condition="'$(Configuration)'=='Release'" Project="..\..\common\vsprops\CodeGen_Release.props" />
|
||||
</ImportGroup>
|
||||
<ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="PropertySheets">
|
||||
<Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
|
||||
<Import Project="..\DefaultProjectRootDir.props" />
|
||||
<Import Project="..\3rdparty.props" />
|
||||
<Import Project="..\..\common\vsprops\CodeGen_Release.props" />
|
||||
</ImportGroup>
|
||||
<ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="PropertySheets">
|
||||
<Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
|
||||
<Import Project="..\DefaultProjectRootDir.props" />
|
||||
<Import Project="..\3rdparty.props" />
|
||||
<Import Project="..\..\common\vsprops\CodeGen_Debug.props" />
|
||||
<Import Project="..\..\common\vsprops\IncrementalLinking.props" />
|
||||
</ImportGroup>
|
||||
<PropertyGroup Label="UserMacros" />
|
||||
<PropertyGroup>
|
||||
<_ProjectFileVersion>10.0.30319.1</_ProjectFileVersion>
|
||||
<CodeAnalysisRuleSet Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">AllRules.ruleset</CodeAnalysisRuleSet>
|
||||
<CodeAnalysisRules Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" />
|
||||
<CodeAnalysisRuleAssemblies Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" />
|
||||
<CodeAnalysisRuleSet Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'">AllRules.ruleset</CodeAnalysisRuleSet>
|
||||
<CodeAnalysisRules Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'" />
|
||||
<CodeAnalysisRuleAssemblies Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'" />
|
||||
<CodeAnalysisRuleSet Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">AllRules.ruleset</CodeAnalysisRuleSet>
|
||||
<CodeAnalysisRules Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" />
|
||||
<CodeAnalysisRuleAssemblies Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" />
|
||||
<TargetName Condition="'$(Configuration)|$(Platform)'=='Devel|Win32'">$(ProjectName)-dev</TargetName>
|
||||
</PropertyGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
|
||||
<ItemDefinitionGroup>
|
||||
<ClCompile>
|
||||
<WarningLevel>Level3</WarningLevel>
|
||||
</ClCompile>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
|
||||
<ClCompile>
|
||||
<WarningLevel>Level3</WarningLevel>
|
||||
<AdditionalIncludeDirectories>$(ProjectDir)soundtouch;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemGroup>
|
||||
<ClCompile Include="3dnow_win.cpp" />
|
||||
<ClCompile Include="AAFilter.cpp" />
|
||||
<ClCompile Include="cpu_detect_x86_win.cpp" />
|
||||
<ClCompile Include="FIFOSampleBuffer.cpp" />
|
||||
<ClCompile Include="FIRFilter.cpp" />
|
||||
<ClCompile Include="mmx_optimized.cpp" />
|
||||
<ClCompile Include="RateTransposer.cpp" />
|
||||
<ClCompile Include="SoundTouch.cpp" />
|
||||
<ClCompile Include="sse_optimized.cpp" />
|
||||
<ClCompile Include="TDStretch.cpp" />
|
||||
<ClCompile Include="WavFile.cpp" />
|
||||
<ClCompile Include="source\SoundStretch\WavFile.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\AAFilter.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\BPMDetect.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\cpu_detect_x86.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\FIFOSampleBuffer.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\FIRFilter.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\InterpolateCubic.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\InterpolateLinear.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\InterpolateShannon.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\mmx_optimized.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\PeakFinder.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\RateTransposer.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\SoundTouch.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\sse_optimized.cpp" />
|
||||
<ClCompile Include="source\SoundTouch\TDStretch.cpp" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="AAFilter.h" />
|
||||
<ClInclude Include="BPMDetect.h" />
|
||||
<ClInclude Include="cpu_detect.h" />
|
||||
<ClInclude Include="FIFOSampleBuffer.h" />
|
||||
<ClInclude Include="FIFOSamplePipe.h" />
|
||||
<ClInclude Include="FIRFilter.h" />
|
||||
<ClInclude Include="RateTransposer.h" />
|
||||
<ClInclude Include="SoundTouch.h" />
|
||||
<ClInclude Include="STTypes.h" />
|
||||
<ClInclude Include="TDStretch.h" />
|
||||
<ClInclude Include="WavFile.h" />
|
||||
<ClInclude Include="soundtouch\BPMDetect.h" />
|
||||
<ClInclude Include="soundtouch\FIFOSampleBuffer.h" />
|
||||
<ClInclude Include="soundtouch\FIFOSamplePipe.h" />
|
||||
<ClInclude Include="soundtouch\SoundTouch.h" />
|
||||
<ClInclude Include="soundtouch\STTypes.h" />
|
||||
<ClInclude Include="source\SoundStretch\WavFile.h" />
|
||||
<ClInclude Include="source\SoundTouch\AAFilter.h" />
|
||||
<ClInclude Include="source\SoundTouch\cpu_detect.h" />
|
||||
<ClInclude Include="source\SoundTouch\FIRFilter.h" />
|
||||
<ClInclude Include="source\SoundTouch\InterpolateCubic.h" />
|
||||
<ClInclude Include="source\SoundTouch\InterpolateLinear.h" />
|
||||
<ClInclude Include="source\SoundTouch\InterpolateShannon.h" />
|
||||
<ClInclude Include="source\SoundTouch\PeakFinder.h" />
|
||||
<ClInclude Include="source\SoundTouch\RateTransposer.h" />
|
||||
<ClInclude Include="source\SoundTouch\TDStretch.h" />
|
||||
</ItemGroup>
|
||||
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
|
||||
<ImportGroup Label="ExtensionTargets">
|
||||
</ImportGroup>
|
||||
</Project>
|
||||
|
|
@ -11,72 +11,96 @@
|
|||
</Filter>
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClCompile Include="3dnow_win.cpp">
|
||||
<ClCompile Include="source\SoundTouch\AAFilter.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="AAFilter.cpp">
|
||||
<ClCompile Include="source\SoundTouch\BPMDetect.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="cpu_detect_x86_win.cpp">
|
||||
<ClCompile Include="source\SoundTouch\cpu_detect_x86.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="FIFOSampleBuffer.cpp">
|
||||
<ClCompile Include="source\SoundTouch\FIFOSampleBuffer.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="FIRFilter.cpp">
|
||||
<ClCompile Include="source\SoundTouch\FIRFilter.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="mmx_optimized.cpp">
|
||||
<ClCompile Include="source\SoundTouch\InterpolateCubic.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="RateTransposer.cpp">
|
||||
<ClCompile Include="source\SoundTouch\InterpolateLinear.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="SoundTouch.cpp">
|
||||
<ClCompile Include="source\SoundTouch\InterpolateShannon.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="sse_optimized.cpp">
|
||||
<ClCompile Include="source\SoundTouch\mmx_optimized.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="TDStretch.cpp">
|
||||
<ClCompile Include="source\SoundTouch\PeakFinder.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="WavFile.cpp">
|
||||
<ClCompile Include="source\SoundTouch\RateTransposer.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="source\SoundTouch\SoundTouch.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="source\SoundTouch\sse_optimized.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="source\SoundTouch\TDStretch.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="source\SoundStretch\WavFile.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="AAFilter.h">
|
||||
<ClInclude Include="soundtouch\BPMDetect.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="BPMDetect.h">
|
||||
<ClInclude Include="soundtouch\FIFOSampleBuffer.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="cpu_detect.h">
|
||||
<ClInclude Include="soundtouch\FIFOSamplePipe.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="FIFOSampleBuffer.h">
|
||||
<ClInclude Include="soundtouch\SoundTouch.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="FIFOSamplePipe.h">
|
||||
<ClInclude Include="soundtouch\STTypes.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="FIRFilter.h">
|
||||
<ClInclude Include="source\SoundTouch\AAFilter.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="RateTransposer.h">
|
||||
<ClInclude Include="source\SoundTouch\cpu_detect.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="SoundTouch.h">
|
||||
<ClInclude Include="source\SoundTouch\FIRFilter.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="STTypes.h">
|
||||
<ClInclude Include="source\SoundTouch\InterpolateCubic.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="TDStretch.h">
|
||||
<ClInclude Include="source\SoundTouch\InterpolateLinear.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="WavFile.h">
|
||||
<ClInclude Include="source\SoundTouch\InterpolateShannon.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="source\SoundTouch\PeakFinder.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="source\SoundTouch\RateTransposer.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="source\SoundTouch\TDStretch.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="source\SoundStretch\WavFile.h">
|
||||
<Filter>Header Files</Filter>
|
||||
</ClInclude>
|
||||
</ItemGroup>
|
||||
|
|
|
|||
|
|
@ -1,745 +0,0 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Classes for easy reading & writing of WAV sound files.
|
||||
///
|
||||
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
|
||||
/// parse the WAV files with such processors.
|
||||
///
|
||||
/// Admittingly, more complete WAV reader routines may exist in public domain,
|
||||
/// but the reason for 'yet another' one is that those generic WAV reader
|
||||
/// libraries are exhaustingly large and cumbersome! Wanted to have something
|
||||
/// simpler here, i.e. something that's not already larger than rest of the
|
||||
/// SoundTouch/SoundStretch program...
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: WavFile.cpp 63 2009-02-21 16:00:14Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdexcept>
|
||||
#include <string>
|
||||
#include <cstring>
|
||||
#include <assert.h>
|
||||
#include <limits.h>
|
||||
|
||||
#include "WavFile.h"
|
||||
|
||||
using namespace std;
|
||||
|
||||
static const char riffStr[] = "RIFF";
|
||||
static const char waveStr[] = "WAVE";
|
||||
static const char fmtStr[] = "fmt ";
|
||||
static const char dataStr[] = "data";
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Helper functions for swapping byte order to correctly read/write WAV files
|
||||
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
|
||||
// turn-on the conversion if it appears necessary.
|
||||
//
|
||||
// For example, Intel x86 is little-endian and doesn't require conversion,
|
||||
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
|
||||
|
||||
#ifdef BYTE_ORDER
|
||||
// In gcc compiler detect the byte order automatically
|
||||
#if BYTE_ORDER == BIG_ENDIAN
|
||||
// big-endian platform.
|
||||
#define _BIG_ENDIAN_
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifdef _BIG_ENDIAN_
|
||||
// big-endian CPU, swap bytes in 16 & 32 bit words
|
||||
|
||||
// helper-function to swap byte-order of 32bit integer
|
||||
static inline void _swap32(unsigned int &dwData)
|
||||
{
|
||||
dwData = ((dwData >> 24) & 0x000000FF) |
|
||||
((dwData >> 8) & 0x0000FF00) |
|
||||
((dwData << 8) & 0x00FF0000) |
|
||||
((dwData << 24) & 0xFF000000);
|
||||
}
|
||||
|
||||
// helper-function to swap byte-order of 16bit integer
|
||||
static inline void _swap16(unsigned short &wData)
|
||||
{
|
||||
wData = ((wData >> 8) & 0x00FF) |
|
||||
((wData << 8) & 0xFF00);
|
||||
}
|
||||
|
||||
// helper-function to swap byte-order of buffer of 16bit integers
|
||||
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumWords)
|
||||
{
|
||||
unsigned long i;
|
||||
|
||||
for (i = 0; i < dwNumWords; i ++)
|
||||
{
|
||||
_swap16(pData[i]);
|
||||
}
|
||||
}
|
||||
|
||||
#else // BIG_ENDIAN
|
||||
// little-endian CPU, WAV file is ok as such
|
||||
|
||||
// dummy helper-function
|
||||
static inline void _swap32(unsigned int &dwData)
|
||||
{
|
||||
// do nothing
|
||||
}
|
||||
|
||||
// dummy helper-function
|
||||
static inline void _swap16(unsigned short &wData)
|
||||
{
|
||||
// do nothing
|
||||
}
|
||||
|
||||
// dummy helper-function
|
||||
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumBytes)
|
||||
{
|
||||
// do nothing
|
||||
}
|
||||
|
||||
#endif // BIG_ENDIAN
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Class WavInFile
|
||||
//
|
||||
|
||||
WavInFile::WavInFile(const char *fileName)
|
||||
{
|
||||
// Try to open the file for reading
|
||||
fptr = fopen(fileName, "rb");
|
||||
if (fptr == NULL)
|
||||
{
|
||||
// didn't succeed
|
||||
string msg = "Error : Unable to open file \"";
|
||||
msg += fileName;
|
||||
msg += "\" for reading.";
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
init();
|
||||
}
|
||||
|
||||
|
||||
WavInFile::WavInFile(FILE *file)
|
||||
{
|
||||
// Try to open the file for reading
|
||||
fptr = file;
|
||||
if (!file)
|
||||
{
|
||||
// didn't succeed
|
||||
string msg = "Error : Unable to access input stream for reading";
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
init();
|
||||
}
|
||||
|
||||
|
||||
/// Init the WAV file stream
|
||||
void WavInFile::init()
|
||||
{
|
||||
int hdrsOk;
|
||||
|
||||
// assume file stream is already open
|
||||
assert(fptr);
|
||||
|
||||
// Read the file headers
|
||||
hdrsOk = readWavHeaders();
|
||||
if (hdrsOk != 0)
|
||||
{
|
||||
// Something didn't match in the wav file headers
|
||||
string msg = "Input file is corrupt or not a WAV file";
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
if (header.format.fixed != 1)
|
||||
{
|
||||
string msg = "Input file uses unsupported encoding.";
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
dataRead = 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
WavInFile::~WavInFile()
|
||||
{
|
||||
if (fptr) fclose(fptr);
|
||||
fptr = NULL;
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavInFile::rewind()
|
||||
{
|
||||
int hdrsOk;
|
||||
|
||||
fseek(fptr, 0, SEEK_SET);
|
||||
hdrsOk = readWavHeaders();
|
||||
assert(hdrsOk == 0);
|
||||
dataRead = 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::checkCharTags() const
|
||||
{
|
||||
// header.format.fmt should equal to 'fmt '
|
||||
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
|
||||
// header.data.data_field should equal to 'data'
|
||||
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::read(char *buffer, int maxElems)
|
||||
{
|
||||
int numBytes;
|
||||
uint afterDataRead;
|
||||
|
||||
// ensure it's 8 bit format
|
||||
if (header.format.bits_per_sample != 8)
|
||||
{
|
||||
throw runtime_error("Error: WavInFile::read(char*, int) works only with 8bit samples.");
|
||||
}
|
||||
assert(sizeof(char) == 1);
|
||||
|
||||
numBytes = maxElems;
|
||||
afterDataRead = dataRead + numBytes;
|
||||
if (afterDataRead > header.data.data_len)
|
||||
{
|
||||
// Don't read more samples than are marked available in header
|
||||
numBytes = (int)header.data.data_len - (int)dataRead;
|
||||
assert(numBytes >= 0);
|
||||
}
|
||||
|
||||
assert(buffer);
|
||||
numBytes = fread(buffer, 1, numBytes, fptr);
|
||||
dataRead += numBytes;
|
||||
|
||||
return numBytes;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::read(short *buffer, int maxElems)
|
||||
{
|
||||
unsigned int afterDataRead;
|
||||
int numBytes;
|
||||
int numElems;
|
||||
|
||||
assert(buffer);
|
||||
if (header.format.bits_per_sample == 8)
|
||||
{
|
||||
// 8 bit format
|
||||
char *temp = new char[maxElems];
|
||||
int i;
|
||||
|
||||
numElems = read(temp, maxElems);
|
||||
// convert from 8 to 16 bit
|
||||
for (i = 0; i < numElems; i ++)
|
||||
{
|
||||
buffer[i] = temp[i] << 8;
|
||||
}
|
||||
delete[] temp;
|
||||
}
|
||||
else
|
||||
{
|
||||
// 16 bit format
|
||||
assert(header.format.bits_per_sample == 16);
|
||||
assert(sizeof(short) == 2);
|
||||
|
||||
numBytes = maxElems * 2;
|
||||
afterDataRead = dataRead + numBytes;
|
||||
if (afterDataRead > header.data.data_len)
|
||||
{
|
||||
// Don't read more samples than are marked available in header
|
||||
numBytes = (int)header.data.data_len - (int)dataRead;
|
||||
assert(numBytes >= 0);
|
||||
}
|
||||
|
||||
numBytes = fread(buffer, 1, numBytes, fptr);
|
||||
dataRead += numBytes;
|
||||
numElems = numBytes / 2;
|
||||
|
||||
// 16bit samples, swap byte order if necessary
|
||||
_swap16Buffer((unsigned short *)buffer, numElems);
|
||||
}
|
||||
|
||||
return numElems;
|
||||
}
|
||||
|
||||
|
||||
|
||||
int WavInFile::read(float *buffer, int maxElems)
|
||||
{
|
||||
short *temp = new short[maxElems];
|
||||
int num;
|
||||
int i;
|
||||
double fscale;
|
||||
|
||||
num = read(temp, maxElems);
|
||||
|
||||
fscale = 1.0 / 32768.0;
|
||||
// convert to floats, scale to range [-1..+1[
|
||||
for (i = 0; i < num; i ++)
|
||||
{
|
||||
buffer[i] = (float)(fscale * (double)temp[i]);
|
||||
}
|
||||
|
||||
delete[] temp;
|
||||
|
||||
return num;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::eof() const
|
||||
{
|
||||
// return true if all data has been read or file eof has reached
|
||||
return (dataRead == header.data.data_len || feof(fptr));
|
||||
}
|
||||
|
||||
|
||||
|
||||
// test if character code is between a white space ' ' and little 'z'
|
||||
static int isAlpha(char c)
|
||||
{
|
||||
return (c >= ' ' && c <= 'z') ? 1 : 0;
|
||||
}
|
||||
|
||||
|
||||
// test if all characters are between a white space ' ' and little 'z'
|
||||
static int isAlphaStr(const char *str)
|
||||
{
|
||||
char c;
|
||||
|
||||
c = str[0];
|
||||
while (c)
|
||||
{
|
||||
if (isAlpha(c) == 0) return 0;
|
||||
str ++;
|
||||
c = str[0];
|
||||
}
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::readRIFFBlock()
|
||||
{
|
||||
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
|
||||
|
||||
// swap 32bit data byte order if necessary
|
||||
_swap32((unsigned int &)header.riff.package_len);
|
||||
|
||||
// header.riff.riff_char should equal to 'RIFF');
|
||||
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
|
||||
// header.riff.wave should equal to 'WAVE'
|
||||
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
int WavInFile::readHeaderBlock()
|
||||
{
|
||||
char label[5];
|
||||
string sLabel;
|
||||
|
||||
// lead label string
|
||||
if (fread(label, 1, 4, fptr) !=4) return -1;
|
||||
label[4] = 0;
|
||||
|
||||
if (isAlphaStr(label) == 0) return -1; // not a valid label
|
||||
|
||||
// Decode blocks according to their label
|
||||
if (strcmp(label, fmtStr) == 0)
|
||||
{
|
||||
int nLen, nDump;
|
||||
|
||||
// 'fmt ' block
|
||||
memcpy(header.format.fmt, fmtStr, 4);
|
||||
|
||||
// read length of the format field
|
||||
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
|
||||
// swap byte order if necessary
|
||||
_swap32((unsigned int &)nLen); // int format_len;
|
||||
header.format.format_len = nLen;
|
||||
|
||||
// calculate how much length differs from expected
|
||||
nDump = nLen - ((int)sizeof(header.format) - 8);
|
||||
|
||||
// if format_len is larger than expected, read only as much data as we've space for
|
||||
if (nDump > 0)
|
||||
{
|
||||
nLen = sizeof(header.format) - 8;
|
||||
}
|
||||
|
||||
// read data
|
||||
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
|
||||
|
||||
// swap byte order if necessary
|
||||
_swap16((unsigned short &)header.format.fixed); // short int fixed;
|
||||
_swap16((unsigned short &)header.format.channel_number); // short int channel_number;
|
||||
_swap32((unsigned int &)header.format.sample_rate); // int sample_rate;
|
||||
_swap32((unsigned int &)header.format.byte_rate); // int byte_rate;
|
||||
_swap16((unsigned short &)header.format.byte_per_sample); // short int byte_per_sample;
|
||||
_swap16((unsigned short &)header.format.bits_per_sample); // short int bits_per_sample;
|
||||
|
||||
// if format_len is larger than expected, skip the extra data
|
||||
if (nDump > 0)
|
||||
{
|
||||
fseek(fptr, nDump, SEEK_CUR);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
else if (strcmp(label, dataStr) == 0)
|
||||
{
|
||||
// 'data' block
|
||||
memcpy(header.data.data_field, dataStr, 4);
|
||||
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
|
||||
|
||||
// swap byte order if necessary
|
||||
_swap32((unsigned int &)header.data.data_len);
|
||||
|
||||
return 1;
|
||||
}
|
||||
else
|
||||
{
|
||||
uint len, i;
|
||||
uint temp;
|
||||
// unknown block
|
||||
|
||||
// read length
|
||||
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
|
||||
// scan through the block
|
||||
for (i = 0; i < len; i ++)
|
||||
{
|
||||
if (fread(&temp, 1, 1, fptr) != 1) return -1;
|
||||
if (feof(fptr)) return -1; // unexpected eof
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::readWavHeaders()
|
||||
{
|
||||
int res;
|
||||
|
||||
memset(&header, 0, sizeof(header));
|
||||
|
||||
res = readRIFFBlock();
|
||||
if (res) return 1;
|
||||
// read header blocks until data block is found
|
||||
do
|
||||
{
|
||||
// read header blocks
|
||||
res = readHeaderBlock();
|
||||
if (res < 0) return 1; // error in file structure
|
||||
} while (res == 0);
|
||||
// check that all required tags are legal
|
||||
return checkCharTags();
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumChannels() const
|
||||
{
|
||||
return header.format.channel_number;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumBits() const
|
||||
{
|
||||
return header.format.bits_per_sample;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getBytesPerSample() const
|
||||
{
|
||||
return getNumChannels() * getNumBits() / 8;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getSampleRate() const
|
||||
{
|
||||
return header.format.sample_rate;
|
||||
}
|
||||
|
||||
|
||||
|
||||
uint WavInFile::getDataSizeInBytes() const
|
||||
{
|
||||
return header.data.data_len;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumSamples() const
|
||||
{
|
||||
if (header.format.byte_per_sample == 0) return 0;
|
||||
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getLengthMS() const
|
||||
{
|
||||
uint numSamples;
|
||||
uint sampleRate;
|
||||
|
||||
numSamples = getNumSamples();
|
||||
sampleRate = getSampleRate();
|
||||
|
||||
assert(numSamples < UINT_MAX / 1000);
|
||||
return (1000 * numSamples / sampleRate);
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Class WavOutFile
|
||||
//
|
||||
|
||||
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
|
||||
{
|
||||
bytesWritten = 0;
|
||||
fptr = fopen(fileName, "wb");
|
||||
if (fptr == NULL)
|
||||
{
|
||||
string msg = "Error : Unable to open file \"";
|
||||
msg += fileName;
|
||||
msg += "\" for writing.";
|
||||
//pmsg = msg.c_str;
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
fillInHeader(sampleRate, bits, channels);
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
|
||||
{
|
||||
bytesWritten = 0;
|
||||
fptr = file;
|
||||
if (fptr == NULL)
|
||||
{
|
||||
string msg = "Error : Unable to access output file stream.";
|
||||
throw runtime_error(msg);
|
||||
}
|
||||
|
||||
fillInHeader(sampleRate, bits, channels);
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
|
||||
WavOutFile::~WavOutFile()
|
||||
{
|
||||
finishHeader();
|
||||
if (fptr) fclose(fptr);
|
||||
fptr = NULL;
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
|
||||
{
|
||||
// fill in the 'riff' part..
|
||||
|
||||
// copy string 'RIFF' to riff_char
|
||||
memcpy(&(header.riff.riff_char), riffStr, 4);
|
||||
// package_len unknown so far
|
||||
header.riff.package_len = 0;
|
||||
// copy string 'WAVE' to wave
|
||||
memcpy(&(header.riff.wave), waveStr, 4);
|
||||
|
||||
|
||||
// fill in the 'format' part..
|
||||
|
||||
// copy string 'fmt ' to fmt
|
||||
memcpy(&(header.format.fmt), fmtStr, 4);
|
||||
|
||||
header.format.format_len = 0x10;
|
||||
header.format.fixed = 1;
|
||||
header.format.channel_number = (short)channels;
|
||||
header.format.sample_rate = (int)sampleRate;
|
||||
header.format.bits_per_sample = (short)bits;
|
||||
header.format.byte_per_sample = (short)(bits * channels / 8);
|
||||
header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
|
||||
header.format.sample_rate = (int)sampleRate;
|
||||
|
||||
// fill in the 'data' part..
|
||||
|
||||
// copy string 'data' to data_field
|
||||
memcpy(&(header.data.data_field), dataStr, 4);
|
||||
// data_len unknown so far
|
||||
header.data.data_len = 0;
|
||||
}
|
||||
|
||||
|
||||
void WavOutFile::finishHeader()
|
||||
{
|
||||
// supplement the file length into the header structure
|
||||
header.riff.package_len = bytesWritten + 36;
|
||||
header.data.data_len = bytesWritten;
|
||||
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::writeHeader()
|
||||
{
|
||||
WavHeader hdrTemp;
|
||||
int res;
|
||||
|
||||
// swap byte order if necessary
|
||||
hdrTemp = header;
|
||||
_swap32((unsigned int &)hdrTemp.riff.package_len);
|
||||
_swap32((unsigned int &)hdrTemp.format.format_len);
|
||||
_swap16((unsigned short &)hdrTemp.format.fixed);
|
||||
_swap16((unsigned short &)hdrTemp.format.channel_number);
|
||||
_swap32((unsigned int &)hdrTemp.format.sample_rate);
|
||||
_swap32((unsigned int &)hdrTemp.format.byte_rate);
|
||||
_swap16((unsigned short &)hdrTemp.format.byte_per_sample);
|
||||
_swap16((unsigned short &)hdrTemp.format.bits_per_sample);
|
||||
_swap32((unsigned int &)hdrTemp.data.data_len);
|
||||
|
||||
// write the supplemented header in the beginning of the file
|
||||
fseek(fptr, 0, SEEK_SET);
|
||||
res = fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
|
||||
if (res != 1)
|
||||
{
|
||||
throw runtime_error("Error while writing to a wav file.");
|
||||
}
|
||||
|
||||
// jump back to the end of the file
|
||||
fseek(fptr, 0, SEEK_END);
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::write(const char *buffer, int numElems)
|
||||
{
|
||||
int res;
|
||||
|
||||
if (header.format.bits_per_sample != 8)
|
||||
{
|
||||
throw runtime_error("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
|
||||
}
|
||||
assert(sizeof(char) == 1);
|
||||
|
||||
res = fwrite(buffer, 1, numElems, fptr);
|
||||
if (res != numElems)
|
||||
{
|
||||
throw runtime_error("Error while writing to a wav file.");
|
||||
}
|
||||
|
||||
bytesWritten += numElems;
|
||||
}
|
||||
|
||||
|
||||
void WavOutFile::write(const short *buffer, int numElems)
|
||||
{
|
||||
int res;
|
||||
|
||||
// 16 bit samples
|
||||
if (numElems < 1) return; // nothing to do
|
||||
|
||||
if (header.format.bits_per_sample == 8)
|
||||
{
|
||||
int i;
|
||||
char *temp = new char[numElems];
|
||||
// convert from 16bit format to 8bit format
|
||||
for (i = 0; i < numElems; i ++)
|
||||
{
|
||||
temp[i] = buffer[i] >> 8;
|
||||
}
|
||||
// write in 8bit format
|
||||
write(temp, numElems);
|
||||
delete[] temp;
|
||||
}
|
||||
else
|
||||
{
|
||||
// 16bit format
|
||||
unsigned short *pTemp = new unsigned short[numElems];
|
||||
|
||||
assert(header.format.bits_per_sample == 16);
|
||||
|
||||
// allocate temp buffer to swap byte order if necessary
|
||||
memcpy(pTemp, buffer, numElems * 2);
|
||||
_swap16Buffer(pTemp, numElems);
|
||||
|
||||
res = fwrite(pTemp, 2, numElems, fptr);
|
||||
|
||||
delete[] pTemp;
|
||||
|
||||
if (res != numElems)
|
||||
{
|
||||
throw runtime_error("Error while writing to a wav file.");
|
||||
}
|
||||
bytesWritten += 2 * numElems;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
void WavOutFile::write(const float *buffer, int numElems)
|
||||
{
|
||||
int i;
|
||||
short *temp = new short[numElems];
|
||||
int iTemp;
|
||||
|
||||
// convert to 16 bit integer
|
||||
for (i = 0; i < numElems; i ++)
|
||||
{
|
||||
// convert to integer
|
||||
iTemp = (int)(32768.0f * buffer[i]);
|
||||
|
||||
// saturate
|
||||
if (iTemp < -32768) iTemp = -32768;
|
||||
if (iTemp > 32767) iTemp = 32767;
|
||||
temp[i] = (short)iTemp;
|
||||
}
|
||||
|
||||
write(temp, numElems);
|
||||
|
||||
delete[] temp;
|
||||
}
|
||||
|
|
@ -1,28 +0,0 @@
|
|||
#!/bin/sh
|
||||
|
||||
curdir=`pwd`
|
||||
|
||||
echo -----------------
|
||||
echo Building SoundTouch
|
||||
echo -----------------
|
||||
|
||||
if [ $# -gt 0 ] && [ $1 = "all" ]
|
||||
then
|
||||
|
||||
aclocal
|
||||
automake -a
|
||||
autoconf
|
||||
./configure
|
||||
make clean
|
||||
make install
|
||||
|
||||
else
|
||||
make $@
|
||||
fi
|
||||
|
||||
if [ $? -ne 0 ]
|
||||
then
|
||||
exit 1
|
||||
fi
|
||||
|
||||
#cp libZeroSPU2*.so* ${PCSX2PLUGINS}
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
# -*- Autoconf -*-
|
||||
# Process this file with autoconf to produce a configure script.
|
||||
|
||||
#AC_PREREQ([2.63])
|
||||
AC_INIT([FULL-PACKAGE-NAME], [VERSION], [BUG-REPORT-ADDRESS])
|
||||
AM_INIT_AUTOMAKE
|
||||
AC_CONFIG_SRCDIR([BPMDetect.h])
|
||||
|
||||
# Checks for programs.
|
||||
AC_PROG_CXX
|
||||
AC_PROG_CC
|
||||
AC_PROG_RANLIB
|
||||
|
||||
CFLAGS=
|
||||
CPPFLAGS=
|
||||
CXXFLAGS=
|
||||
CCASFLAGS=
|
||||
|
||||
CFLAGS+=" -m32 "
|
||||
CPPFLAGS+=" -m32 "
|
||||
CXXFLAGS+=" -m32 "
|
||||
CCASFLAGS+=" -m32 "
|
||||
|
||||
# Checks for header files.
|
||||
AC_CHECK_HEADERS([limits.h memory.h stdlib.h string.h])
|
||||
|
||||
# Checks for typedefs, structures, and compiler characteristics.
|
||||
AC_C_INLINE
|
||||
AC_C_RESTRICT
|
||||
AC_TYPE_SIZE_T
|
||||
AC_HEADER_STDBOOL
|
||||
|
||||
# Checks for library functions.
|
||||
AC_CHECK_FUNCS([memmove memset])
|
||||
|
||||
AC_CONFIG_FILES([Makefile])
|
||||
AC_OUTPUT
|
||||
|
|
@ -1,134 +0,0 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Generic version of the x86 CPU extension detection routine.
|
||||
///
|
||||
/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
|
||||
/// for the Microsoft compiler version.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect_x86_gcc.cpp 131 2011-09-02 18:56:11Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include "cpu_detect.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// processor instructions extension detection routines
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Flag variable indicating whick ISA extensions are disabled (for debugging)
|
||||
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
|
||||
|
||||
// Disables given set of instruction extensions. See SUPPORT_... defines.
|
||||
void disableExtensions(uint dwDisableMask)
|
||||
{
|
||||
_dwDisabledISA = dwDisableMask;
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Checks which instruction set extensions are supported by the CPU.
|
||||
uint detectCPUextensions(void)
|
||||
{
|
||||
#if (!(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS) || !(__GNUC__))
|
||||
|
||||
return 0; // always disable extensions on non-x86 platforms.
|
||||
|
||||
#else
|
||||
uint res = 0;
|
||||
|
||||
if (_dwDisabledISA == 0xffffffff) return 0;
|
||||
|
||||
asm volatile(
|
||||
#ifndef __x86_64__
|
||||
// Check if 'cpuid' instructions is available by toggling eflags bit 21.
|
||||
// Skip this for x86-64 as they always have cpuid while stack manipulation
|
||||
// differs from 16/32bit ISA.
|
||||
"\n\txor %%esi, %%esi" // clear %%esi = result register
|
||||
|
||||
"\n\tpushf" // save eflags to stack
|
||||
"\n\tmovl (%%esp), %%eax" // load eax from stack (with eflags)
|
||||
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
|
||||
"\n\txor $0x00200000, %%eax" // toggle bit 21
|
||||
"\n\tmovl %%eax, (%%esp)" // store toggled eflags to stack
|
||||
"\n\tpopf" // load eflags from stack
|
||||
"\n\tpushf" // save updated eflags to stack
|
||||
"\n\tmovl (%%esp), %%eax" // load eax from stack
|
||||
"\n\tpopf" // pop stack to restore esp
|
||||
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
|
||||
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
|
||||
"\n\tjz end" // jumps to 'end' if cpuid not present
|
||||
#endif // __x86_64__
|
||||
|
||||
// cpuid instruction available, test for presence of mmx instructions
|
||||
|
||||
"\n\tmovl $1, %%eax"
|
||||
"\n\tcpuid"
|
||||
"\n\ttest $0x00800000, %%edx"
|
||||
"\n\tjz end" // branch if MMX not available
|
||||
|
||||
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
|
||||
|
||||
"\n\ttest $0x02000000, %%edx"
|
||||
"\n\tjz test3DNow" // branch if SSE not available
|
||||
|
||||
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
|
||||
|
||||
"\n\ttest3DNow:"
|
||||
// test for precense of AMD extensions
|
||||
"\n\tmov $0x80000000, %%eax"
|
||||
"\n\tcpuid"
|
||||
"\n\tcmp $0x80000000, %%eax"
|
||||
"\n\tjbe end" // branch if no AMD extensions detected
|
||||
|
||||
// test for precense of 3DNow! extension
|
||||
"\n\tmov $0x80000001, %%eax"
|
||||
"\n\tcpuid"
|
||||
"\n\ttest $0x80000000, %%edx"
|
||||
"\n\tjz end" // branch if 3DNow! not detected
|
||||
|
||||
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
|
||||
|
||||
"\n\tend:"
|
||||
|
||||
"\n\tmov %%esi, %0"
|
||||
|
||||
: "=r" (res)
|
||||
: /* no inputs */
|
||||
: "%edx", "%eax", "%ecx", "%esi" );
|
||||
|
||||
return res & ~_dwDisabledISA;
|
||||
#endif
|
||||
}
|
||||
|
|
@ -1,137 +0,0 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Win32 version of the x86 CPU detect routine.
|
||||
///
|
||||
/// This file is to be compiled in Windows platform with Microsoft Visual C++
|
||||
/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version
|
||||
/// for all GNU platforms.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-07-17 07:58:40 -0300 (dom, 17 jul 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect_x86_win.cpp 127 2011-07-17 10:58:40Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include "cpu_detect.h"
|
||||
|
||||
#include "STTypes.h"
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// processor instructions extension detection routines
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Flag variable indicating whick ISA extensions are disabled (for debugging)
|
||||
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
|
||||
|
||||
|
||||
// Disables given set of instruction extensions. See SUPPORT_... defines.
|
||||
void disableExtensions(uint dwDisableMask)
|
||||
{
|
||||
_dwDisabledISA = dwDisableMask;
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Checks which instruction set extensions are supported by the CPU.
|
||||
uint detectCPUextensions(void)
|
||||
{
|
||||
uint res = 0;
|
||||
|
||||
if (_dwDisabledISA == 0xffffffff) return 0;
|
||||
|
||||
#ifndef _M_X64
|
||||
// 32bit compilation, detect CPU capabilities with inline assembler.
|
||||
__asm
|
||||
{
|
||||
; check if 'cpuid' instructions is available by toggling eflags bit 21
|
||||
;
|
||||
xor esi, esi ; clear esi = result register
|
||||
|
||||
pushfd ; save eflags to stack
|
||||
mov eax,dword ptr [esp] ; load eax from stack (with eflags)
|
||||
mov ecx, eax ; save the original eflags values to ecx
|
||||
xor eax, 0x00200000 ; toggle bit 21
|
||||
mov dword ptr [esp],eax ; store toggled eflags to stack
|
||||
popfd ; load eflags from stack
|
||||
|
||||
pushfd ; save updated eflags to stack
|
||||
mov eax,dword ptr [esp] ; load eax from stack
|
||||
popfd ; pop stack to restore stack pointer
|
||||
|
||||
xor edx, edx ; clear edx for defaulting no mmx
|
||||
cmp eax, ecx ; compare to original eflags values
|
||||
jz end ; jumps to 'end' if cpuid not present
|
||||
|
||||
; cpuid instruction available, test for presence of mmx instructions
|
||||
mov eax, 1
|
||||
cpuid
|
||||
test edx, 0x00800000
|
||||
jz end ; branch if MMX not available
|
||||
|
||||
or esi, SUPPORT_MMX ; otherwise add MMX support bit
|
||||
|
||||
test edx, 0x02000000
|
||||
jz test3DNow ; branch if SSE not available
|
||||
|
||||
or esi, SUPPORT_SSE ; otherwise add SSE support bit
|
||||
|
||||
test3DNow:
|
||||
; test for precense of AMD extensions
|
||||
mov eax, 0x80000000
|
||||
cpuid
|
||||
cmp eax, 0x80000000
|
||||
jbe end ; branch if no AMD extensions detected
|
||||
|
||||
; test for precense of 3DNow! extension
|
||||
mov eax, 0x80000001
|
||||
cpuid
|
||||
test edx, 0x80000000
|
||||
jz end ; branch if 3DNow! not detected
|
||||
|
||||
or esi, SUPPORT_3DNOW ; otherwise add 3DNow support bit
|
||||
|
||||
end:
|
||||
|
||||
mov res, esi
|
||||
}
|
||||
|
||||
#else
|
||||
|
||||
// Visual C++ 64bit compilation doesn't support inline assembler. However,
|
||||
// all x64 compatible CPUs support MMX & SSE extensions.
|
||||
res = SUPPORT_MMX | SUPPORT_SSE | SUPPORT_SSE2;
|
||||
|
||||
#endif
|
||||
|
||||
return res & ~_dwDisabledISA;
|
||||
}
|
||||
|
|
@ -1 +0,0 @@
|
|||
/usr/share/automake-1.10/depcomp
|
||||
|
|
@ -1 +0,0 @@
|
|||
/usr/share/automake-1.10/install-sh
|
||||
|
|
@ -1 +0,0 @@
|
|||
/usr/share/automake-1.10/missing
|
||||
|
|
@ -26,7 +26,7 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-08-30 16:53:44 -0300 (qui, 30 ago 2012) $
|
||||
// Last changed : $Date: 2012-08-30 19:53:44 +0000 (Thu, 30 Aug 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
|
||||
|
|
@ -15,10 +15,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
|
||||
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $
|
||||
// $Id: FIFOSampleBuffer.h 177 2014-01-05 21:40:22Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -162,6 +162,12 @@ public:
|
|||
/// Sets number of channels, 1 = mono, 2 = stereo.
|
||||
void setChannels(int numChannels);
|
||||
|
||||
/// Get number of channels
|
||||
int getChannels()
|
||||
{
|
||||
return channels;
|
||||
}
|
||||
|
||||
/// Returns nonzero if there aren't any samples available for outputting.
|
||||
virtual int isEmpty() const;
|
||||
|
||||
|
|
@ -17,7 +17,7 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
|
||||
// Last changed : $Date: 2012-06-13 19:29:53 +0000 (Wed, 13 Jun 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
|
||||
|
|
@ -8,10 +8,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-12-28 12:53:56 -0200 (sex, 28 dez 2012) $
|
||||
// Last changed : $Date: 2015-05-18 15:25:07 +0000 (Mon, 18 May 2015) $
|
||||
// File revision : $Revision: 3 $
|
||||
//
|
||||
// $Id: STTypes.h 162 2012-12-28 14:53:56Z oparviai $
|
||||
// $Id: STTypes.h 215 2015-05-18 15:25:07Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -60,16 +60,6 @@ typedef unsigned long ulong;
|
|||
#include "soundtouch_config.h"
|
||||
#endif
|
||||
|
||||
#ifndef _WINDEF_
|
||||
// if these aren't defined already by Windows headers, define now
|
||||
|
||||
typedef int BOOL;
|
||||
|
||||
#define FALSE 0
|
||||
#define TRUE 1
|
||||
|
||||
#endif // _WINDEF_
|
||||
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
|
|
@ -78,7 +68,14 @@ namespace soundtouch
|
|||
//#undef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
//#undef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
|
||||
#if (defined(__SOFTFP__))
|
||||
/// If following flag is defined, always uses multichannel processing
|
||||
/// routines also for mono and stero sound. This is for routine testing
|
||||
/// purposes; output should be same with either routines, yet disabling
|
||||
/// the dedicated mono/stereo processing routines will result in slower
|
||||
/// runtime performance so recommendation is to keep this off.
|
||||
// #define USE_MULTICH_ALWAYS
|
||||
|
||||
#if (defined(__SOFTFP__) && defined(ANDROID))
|
||||
// For Android compilation: Force use of Integer samples in case that
|
||||
// compilation uses soft-floating point emulation - soft-fp is way too slow
|
||||
#undef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
|
|
@ -175,6 +172,7 @@ namespace soundtouch
|
|||
#else
|
||||
// use c++ standard exceptions
|
||||
#include <stdexcept>
|
||||
#include <string>
|
||||
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
|
||||
#endif
|
||||
|
||||
|
|
@ -41,10 +41,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-12-28 17:32:59 -0200 (sex, 28 dez 2012) $
|
||||
// Last changed : $Date: 2015-05-18 15:28:41 +0000 (Mon, 18 May 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: SoundTouch.h 163 2012-12-28 19:32:59Z oparviai $
|
||||
// $Id: SoundTouch.h 216 2015-05-18 15:28:41Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -79,10 +79,10 @@ namespace soundtouch
|
|||
{
|
||||
|
||||
/// Soundtouch library version string
|
||||
#define SOUNDTOUCH_VERSION "1.7.1"
|
||||
#define SOUNDTOUCH_VERSION "1.9.0"
|
||||
|
||||
/// SoundTouch library version id
|
||||
#define SOUNDTOUCH_VERSION_ID (10701)
|
||||
#define SOUNDTOUCH_VERSION_ID (10900)
|
||||
|
||||
//
|
||||
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
|
||||
|
|
@ -160,7 +160,7 @@ private:
|
|||
float virtualPitch;
|
||||
|
||||
/// Flag: Has sample rate been set?
|
||||
BOOL bSrateSet;
|
||||
bool bSrateSet;
|
||||
|
||||
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
|
||||
/// 'virtualPitch' parameters.
|
||||
|
|
@ -247,8 +247,8 @@ public:
|
|||
/// Changes a setting controlling the processing system behaviour. See the
|
||||
/// 'SETTING_...' defines for available setting ID's.
|
||||
///
|
||||
/// \return 'TRUE' if the setting was succesfully changed
|
||||
BOOL setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
|
||||
/// \return 'true' if the setting was succesfully changed
|
||||
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
|
||||
int value ///< New setting value.
|
||||
);
|
||||
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
|
||||
#ifndef SOUNDTOUCH_CONFIG_H_INCLUDED
|
||||
#define SOUNDTOUCH_CONFIG_H_INCLUDED
|
||||
|
||||
|
||||
|
||||
#endif // SOUNDTOUCH_CONFIG_H_INCLUDED
|
||||
|
|
@ -0,0 +1,997 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Classes for easy reading & writing of WAV sound files.
|
||||
///
|
||||
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
|
||||
/// parse the WAV files with such processors.
|
||||
///
|
||||
/// Admittingly, more complete WAV reader routines may exist in public domain,
|
||||
/// but the reason for 'yet another' one is that those generic WAV reader
|
||||
/// libraries are exhaustingly large and cumbersome! Wanted to have something
|
||||
/// simpler here, i.e. something that's not already larger than rest of the
|
||||
/// SoundTouch/SoundStretch program...
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2014-10-05 16:20:24 +0000 (Sun, 05 Oct 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: WavFile.cpp 200 2014-10-05 16:20:24Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <stdio.h>
|
||||
#include <string>
|
||||
#include <sstream>
|
||||
#include <cstring>
|
||||
#include <assert.h>
|
||||
#include <limits.h>
|
||||
|
||||
#include "WavFile.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
using namespace std;
|
||||
|
||||
static const char riffStr[] = "RIFF";
|
||||
static const char waveStr[] = "WAVE";
|
||||
static const char fmtStr[] = "fmt ";
|
||||
static const char factStr[] = "fact";
|
||||
static const char dataStr[] = "data";
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Helper functions for swapping byte order to correctly read/write WAV files
|
||||
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
|
||||
// turn-on the conversion if it appears necessary.
|
||||
//
|
||||
// For example, Intel x86 is little-endian and doesn't require conversion,
|
||||
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
|
||||
|
||||
#ifdef BYTE_ORDER
|
||||
// In gcc compiler detect the byte order automatically
|
||||
#if BYTE_ORDER == BIG_ENDIAN
|
||||
// big-endian platform.
|
||||
#define _BIG_ENDIAN_
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifdef _BIG_ENDIAN_
|
||||
// big-endian CPU, swap bytes in 16 & 32 bit words
|
||||
|
||||
// helper-function to swap byte-order of 32bit integer
|
||||
static inline int _swap32(int &dwData)
|
||||
{
|
||||
dwData = ((dwData >> 24) & 0x000000FF) |
|
||||
((dwData >> 8) & 0x0000FF00) |
|
||||
((dwData << 8) & 0x00FF0000) |
|
||||
((dwData << 24) & 0xFF000000);
|
||||
return dwData;
|
||||
}
|
||||
|
||||
// helper-function to swap byte-order of 16bit integer
|
||||
static inline short _swap16(short &wData)
|
||||
{
|
||||
wData = ((wData >> 8) & 0x00FF) |
|
||||
((wData << 8) & 0xFF00);
|
||||
return wData;
|
||||
}
|
||||
|
||||
// helper-function to swap byte-order of buffer of 16bit integers
|
||||
static inline void _swap16Buffer(short *pData, int numWords)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < numWords; i ++)
|
||||
{
|
||||
pData[i] = _swap16(pData[i]);
|
||||
}
|
||||
}
|
||||
|
||||
#else // BIG_ENDIAN
|
||||
// little-endian CPU, WAV file is ok as such
|
||||
|
||||
// dummy helper-function
|
||||
static inline int _swap32(int &dwData)
|
||||
{
|
||||
// do nothing
|
||||
return dwData;
|
||||
}
|
||||
|
||||
// dummy helper-function
|
||||
static inline short _swap16(short &wData)
|
||||
{
|
||||
// do nothing
|
||||
return wData;
|
||||
}
|
||||
|
||||
// dummy helper-function
|
||||
static inline void _swap16Buffer(short *pData, int numBytes)
|
||||
{
|
||||
// do nothing
|
||||
}
|
||||
|
||||
#endif // BIG_ENDIAN
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Class WavFileBase
|
||||
//
|
||||
|
||||
WavFileBase::WavFileBase()
|
||||
{
|
||||
convBuff = NULL;
|
||||
convBuffSize = 0;
|
||||
}
|
||||
|
||||
|
||||
WavFileBase::~WavFileBase()
|
||||
{
|
||||
delete[] convBuff;
|
||||
convBuffSize = 0;
|
||||
}
|
||||
|
||||
|
||||
/// Get pointer to conversion buffer of at min. given size
|
||||
void *WavFileBase::getConvBuffer(int sizeBytes)
|
||||
{
|
||||
if (convBuffSize < sizeBytes)
|
||||
{
|
||||
delete[] convBuff;
|
||||
|
||||
convBuffSize = (sizeBytes + 15) & -8; // round up to following 8-byte bounday
|
||||
convBuff = new char[convBuffSize];
|
||||
}
|
||||
return convBuff;
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Class WavInFile
|
||||
//
|
||||
|
||||
WavInFile::WavInFile(const char *fileName)
|
||||
{
|
||||
// Try to open the file for reading
|
||||
fptr = fopen(fileName, "rb");
|
||||
if (fptr == NULL)
|
||||
{
|
||||
// didn't succeed
|
||||
string msg = "Error : Unable to open file \"";
|
||||
msg += fileName;
|
||||
msg += "\" for reading.";
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
|
||||
init();
|
||||
}
|
||||
|
||||
|
||||
WavInFile::WavInFile(FILE *file)
|
||||
{
|
||||
// Try to open the file for reading
|
||||
fptr = file;
|
||||
if (!file)
|
||||
{
|
||||
// didn't succeed
|
||||
string msg = "Error : Unable to access input stream for reading";
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
|
||||
init();
|
||||
}
|
||||
|
||||
|
||||
/// Init the WAV file stream
|
||||
void WavInFile::init()
|
||||
{
|
||||
int hdrsOk;
|
||||
|
||||
// assume file stream is already open
|
||||
assert(fptr);
|
||||
|
||||
// Read the file headers
|
||||
hdrsOk = readWavHeaders();
|
||||
if (hdrsOk != 0)
|
||||
{
|
||||
// Something didn't match in the wav file headers
|
||||
string msg = "Input file is corrupt or not a WAV file";
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
|
||||
/* Ignore 'fixed' field value as 32bit signed linear data can have other value than 1.
|
||||
if (header.format.fixed != 1)
|
||||
{
|
||||
string msg = "Input file uses unsupported encoding.";
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
*/
|
||||
|
||||
dataRead = 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
WavInFile::~WavInFile()
|
||||
{
|
||||
if (fptr) fclose(fptr);
|
||||
fptr = NULL;
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavInFile::rewind()
|
||||
{
|
||||
int hdrsOk;
|
||||
|
||||
fseek(fptr, 0, SEEK_SET);
|
||||
hdrsOk = readWavHeaders();
|
||||
assert(hdrsOk == 0);
|
||||
dataRead = 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::checkCharTags() const
|
||||
{
|
||||
// header.format.fmt should equal to 'fmt '
|
||||
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
|
||||
// header.data.data_field should equal to 'data'
|
||||
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::read(unsigned char *buffer, int maxElems)
|
||||
{
|
||||
int numBytes;
|
||||
uint afterDataRead;
|
||||
|
||||
// ensure it's 8 bit format
|
||||
if (header.format.bits_per_sample != 8)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error: WavInFile::read(char*, int) works only with 8bit samples.");
|
||||
}
|
||||
assert(sizeof(char) == 1);
|
||||
|
||||
numBytes = maxElems;
|
||||
afterDataRead = dataRead + numBytes;
|
||||
if (afterDataRead > header.data.data_len)
|
||||
{
|
||||
// Don't read more samples than are marked available in header
|
||||
numBytes = (int)header.data.data_len - (int)dataRead;
|
||||
assert(numBytes >= 0);
|
||||
}
|
||||
|
||||
assert(buffer);
|
||||
numBytes = (int)fread(buffer, 1, numBytes, fptr);
|
||||
dataRead += numBytes;
|
||||
|
||||
return numBytes;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::read(short *buffer, int maxElems)
|
||||
{
|
||||
unsigned int afterDataRead;
|
||||
int numBytes;
|
||||
int numElems;
|
||||
|
||||
assert(buffer);
|
||||
switch (header.format.bits_per_sample)
|
||||
{
|
||||
case 8:
|
||||
{
|
||||
// 8 bit format
|
||||
unsigned char *temp = (unsigned char*)getConvBuffer(maxElems);
|
||||
int i;
|
||||
|
||||
numElems = read(temp, maxElems);
|
||||
// convert from 8 to 16 bit
|
||||
for (i = 0; i < numElems; i ++)
|
||||
{
|
||||
buffer[i] = (short)(((short)temp[i] - 128) * 256);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 16:
|
||||
{
|
||||
// 16 bit format
|
||||
|
||||
assert(sizeof(short) == 2);
|
||||
|
||||
numBytes = maxElems * 2;
|
||||
afterDataRead = dataRead + numBytes;
|
||||
if (afterDataRead > header.data.data_len)
|
||||
{
|
||||
// Don't read more samples than are marked available in header
|
||||
numBytes = (int)header.data.data_len - (int)dataRead;
|
||||
assert(numBytes >= 0);
|
||||
}
|
||||
|
||||
numBytes = (int)fread(buffer, 1, numBytes, fptr);
|
||||
dataRead += numBytes;
|
||||
numElems = numBytes / 2;
|
||||
|
||||
// 16bit samples, swap byte order if necessary
|
||||
_swap16Buffer((short *)buffer, numElems);
|
||||
break;
|
||||
}
|
||||
|
||||
default:
|
||||
{
|
||||
stringstream ss;
|
||||
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
|
||||
ss << (int)header.format.bits_per_sample;
|
||||
ss << " bit sample format. ";
|
||||
ST_THROW_RT_ERROR(ss.str().c_str());
|
||||
}
|
||||
};
|
||||
|
||||
return numElems;
|
||||
}
|
||||
|
||||
|
||||
/// Read data in float format. Notice that when reading in float format
|
||||
/// 8/16/24/32 bit sample formats are supported
|
||||
int WavInFile::read(float *buffer, int maxElems)
|
||||
{
|
||||
unsigned int afterDataRead;
|
||||
int numBytes;
|
||||
int numElems;
|
||||
int bytesPerSample;
|
||||
|
||||
assert(buffer);
|
||||
|
||||
bytesPerSample = header.format.bits_per_sample / 8;
|
||||
if ((bytesPerSample < 1) || (bytesPerSample > 4))
|
||||
{
|
||||
stringstream ss;
|
||||
ss << "\nOnly 8/16/24/32 bit sample WAV files supported. Can't open WAV file with ";
|
||||
ss << (int)header.format.bits_per_sample;
|
||||
ss << " bit sample format. ";
|
||||
ST_THROW_RT_ERROR(ss.str().c_str());
|
||||
}
|
||||
|
||||
numBytes = maxElems * bytesPerSample;
|
||||
afterDataRead = dataRead + numBytes;
|
||||
if (afterDataRead > header.data.data_len)
|
||||
{
|
||||
// Don't read more samples than are marked available in header
|
||||
numBytes = (int)header.data.data_len - (int)dataRead;
|
||||
assert(numBytes >= 0);
|
||||
}
|
||||
|
||||
// read raw data into temporary buffer
|
||||
char *temp = (char*)getConvBuffer(numBytes);
|
||||
numBytes = (int)fread(temp, 1, numBytes, fptr);
|
||||
dataRead += numBytes;
|
||||
|
||||
numElems = numBytes / bytesPerSample;
|
||||
|
||||
// swap byte ordert & convert to float, depending on sample format
|
||||
switch (bytesPerSample)
|
||||
{
|
||||
case 1:
|
||||
{
|
||||
unsigned char *temp2 = (unsigned char*)temp;
|
||||
double conv = 1.0 / 128.0;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
buffer[i] = (float)(temp2[i] * conv - 1.0);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 2:
|
||||
{
|
||||
short *temp2 = (short*)temp;
|
||||
double conv = 1.0 / 32768.0;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
short value = temp2[i];
|
||||
buffer[i] = (float)(_swap16(value) * conv);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 3:
|
||||
{
|
||||
char *temp2 = (char *)temp;
|
||||
double conv = 1.0 / 8388608.0;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
int value = *((int*)temp2);
|
||||
value = _swap32(value) & 0x00ffffff; // take 24 bits
|
||||
value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
|
||||
buffer[i] = (float)(value * conv);
|
||||
temp2 += 3;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 4:
|
||||
{
|
||||
int *temp2 = (int *)temp;
|
||||
double conv = 1.0 / 2147483648.0;
|
||||
assert(sizeof(int) == 4);
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
int value = temp2[i];
|
||||
buffer[i] = (float)(_swap32(value) * conv);
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
return numElems;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::eof() const
|
||||
{
|
||||
// return true if all data has been read or file eof has reached
|
||||
return (dataRead == header.data.data_len || feof(fptr));
|
||||
}
|
||||
|
||||
|
||||
|
||||
// test if character code is between a white space ' ' and little 'z'
|
||||
static int isAlpha(char c)
|
||||
{
|
||||
return (c >= ' ' && c <= 'z') ? 1 : 0;
|
||||
}
|
||||
|
||||
|
||||
// test if all characters are between a white space ' ' and little 'z'
|
||||
static int isAlphaStr(const char *str)
|
||||
{
|
||||
char c;
|
||||
|
||||
c = str[0];
|
||||
while (c)
|
||||
{
|
||||
if (isAlpha(c) == 0) return 0;
|
||||
str ++;
|
||||
c = str[0];
|
||||
}
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::readRIFFBlock()
|
||||
{
|
||||
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
|
||||
|
||||
// swap 32bit data byte order if necessary
|
||||
_swap32((int &)header.riff.package_len);
|
||||
|
||||
// header.riff.riff_char should equal to 'RIFF');
|
||||
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
|
||||
// header.riff.wave should equal to 'WAVE'
|
||||
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
int WavInFile::readHeaderBlock()
|
||||
{
|
||||
char label[5];
|
||||
string sLabel;
|
||||
|
||||
// lead label string
|
||||
if (fread(label, 1, 4, fptr) !=4) return -1;
|
||||
label[4] = 0;
|
||||
|
||||
if (isAlphaStr(label) == 0) return -1; // not a valid label
|
||||
|
||||
// Decode blocks according to their label
|
||||
if (strcmp(label, fmtStr) == 0)
|
||||
{
|
||||
int nLen, nDump;
|
||||
|
||||
// 'fmt ' block
|
||||
memcpy(header.format.fmt, fmtStr, 4);
|
||||
|
||||
// read length of the format field
|
||||
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
|
||||
// swap byte order if necessary
|
||||
_swap32(nLen); // int format_len;
|
||||
header.format.format_len = nLen;
|
||||
|
||||
// calculate how much length differs from expected
|
||||
nDump = nLen - ((int)sizeof(header.format) - 8);
|
||||
|
||||
// if format_len is larger than expected, read only as much data as we've space for
|
||||
if (nDump > 0)
|
||||
{
|
||||
nLen = sizeof(header.format) - 8;
|
||||
}
|
||||
|
||||
// read data
|
||||
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
|
||||
|
||||
// swap byte order if necessary
|
||||
_swap16(header.format.fixed); // short int fixed;
|
||||
_swap16(header.format.channel_number); // short int channel_number;
|
||||
_swap32((int &)header.format.sample_rate); // int sample_rate;
|
||||
_swap32((int &)header.format.byte_rate); // int byte_rate;
|
||||
_swap16(header.format.byte_per_sample); // short int byte_per_sample;
|
||||
_swap16(header.format.bits_per_sample); // short int bits_per_sample;
|
||||
|
||||
// if format_len is larger than expected, skip the extra data
|
||||
if (nDump > 0)
|
||||
{
|
||||
fseek(fptr, nDump, SEEK_CUR);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
else if (strcmp(label, factStr) == 0)
|
||||
{
|
||||
int nLen, nDump;
|
||||
|
||||
// 'fact' block
|
||||
memcpy(header.fact.fact_field, factStr, 4);
|
||||
|
||||
// read length of the fact field
|
||||
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
|
||||
// swap byte order if necessary
|
||||
_swap32(nLen); // int fact_len;
|
||||
header.fact.fact_len = nLen;
|
||||
|
||||
// calculate how much length differs from expected
|
||||
nDump = nLen - ((int)sizeof(header.fact) - 8);
|
||||
|
||||
// if format_len is larger than expected, read only as much data as we've space for
|
||||
if (nDump > 0)
|
||||
{
|
||||
nLen = sizeof(header.fact) - 8;
|
||||
}
|
||||
|
||||
// read data
|
||||
if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1;
|
||||
|
||||
// swap byte order if necessary
|
||||
_swap32((int &)header.fact.fact_sample_len); // int sample_length;
|
||||
|
||||
// if fact_len is larger than expected, skip the extra data
|
||||
if (nDump > 0)
|
||||
{
|
||||
fseek(fptr, nDump, SEEK_CUR);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
else if (strcmp(label, dataStr) == 0)
|
||||
{
|
||||
// 'data' block
|
||||
memcpy(header.data.data_field, dataStr, 4);
|
||||
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
|
||||
|
||||
// swap byte order if necessary
|
||||
_swap32((int &)header.data.data_len);
|
||||
|
||||
return 1;
|
||||
}
|
||||
else
|
||||
{
|
||||
uint len, i;
|
||||
uint temp;
|
||||
// unknown block
|
||||
|
||||
// read length
|
||||
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
|
||||
// scan through the block
|
||||
for (i = 0; i < len; i ++)
|
||||
{
|
||||
if (fread(&temp, 1, 1, fptr) != 1) return -1;
|
||||
if (feof(fptr)) return -1; // unexpected eof
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
int WavInFile::readWavHeaders()
|
||||
{
|
||||
int res;
|
||||
|
||||
memset(&header, 0, sizeof(header));
|
||||
|
||||
res = readRIFFBlock();
|
||||
if (res) return 1;
|
||||
// read header blocks until data block is found
|
||||
do
|
||||
{
|
||||
// read header blocks
|
||||
res = readHeaderBlock();
|
||||
if (res < 0) return 1; // error in file structure
|
||||
} while (res == 0);
|
||||
// check that all required tags are legal
|
||||
return checkCharTags();
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumChannels() const
|
||||
{
|
||||
return header.format.channel_number;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumBits() const
|
||||
{
|
||||
return header.format.bits_per_sample;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getBytesPerSample() const
|
||||
{
|
||||
return getNumChannels() * getNumBits() / 8;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getSampleRate() const
|
||||
{
|
||||
return header.format.sample_rate;
|
||||
}
|
||||
|
||||
|
||||
|
||||
uint WavInFile::getDataSizeInBytes() const
|
||||
{
|
||||
return header.data.data_len;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getNumSamples() const
|
||||
{
|
||||
if (header.format.byte_per_sample == 0) return 0;
|
||||
if (header.format.fixed > 1) return header.fact.fact_sample_len;
|
||||
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
|
||||
}
|
||||
|
||||
|
||||
uint WavInFile::getLengthMS() const
|
||||
{
|
||||
double numSamples;
|
||||
double sampleRate;
|
||||
|
||||
numSamples = (double)getNumSamples();
|
||||
sampleRate = (double)getSampleRate();
|
||||
|
||||
return (uint)(1000.0 * numSamples / sampleRate + 0.5);
|
||||
}
|
||||
|
||||
|
||||
/// Returns how many milliseconds of audio have so far been read from the file
|
||||
uint WavInFile::getElapsedMS() const
|
||||
{
|
||||
return (uint)(1000.0 * (double)dataRead / (double)header.format.byte_rate);
|
||||
}
|
||||
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Class WavOutFile
|
||||
//
|
||||
|
||||
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
|
||||
{
|
||||
bytesWritten = 0;
|
||||
fptr = fopen(fileName, "wb");
|
||||
if (fptr == NULL)
|
||||
{
|
||||
string msg = "Error : Unable to open file \"";
|
||||
msg += fileName;
|
||||
msg += "\" for writing.";
|
||||
//pmsg = msg.c_str;
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
|
||||
fillInHeader(sampleRate, bits, channels);
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
|
||||
{
|
||||
bytesWritten = 0;
|
||||
fptr = file;
|
||||
if (fptr == NULL)
|
||||
{
|
||||
string msg = "Error : Unable to access output file stream.";
|
||||
ST_THROW_RT_ERROR(msg.c_str());
|
||||
}
|
||||
|
||||
fillInHeader(sampleRate, bits, channels);
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
|
||||
WavOutFile::~WavOutFile()
|
||||
{
|
||||
finishHeader();
|
||||
if (fptr) fclose(fptr);
|
||||
fptr = NULL;
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
|
||||
{
|
||||
// fill in the 'riff' part..
|
||||
|
||||
// copy string 'RIFF' to riff_char
|
||||
memcpy(&(header.riff.riff_char), riffStr, 4);
|
||||
// package_len unknown so far
|
||||
header.riff.package_len = 0;
|
||||
// copy string 'WAVE' to wave
|
||||
memcpy(&(header.riff.wave), waveStr, 4);
|
||||
|
||||
// fill in the 'format' part..
|
||||
|
||||
// copy string 'fmt ' to fmt
|
||||
memcpy(&(header.format.fmt), fmtStr, 4);
|
||||
|
||||
header.format.format_len = 0x10;
|
||||
header.format.fixed = 1;
|
||||
header.format.channel_number = (short)channels;
|
||||
header.format.sample_rate = (int)sampleRate;
|
||||
header.format.bits_per_sample = (short)bits;
|
||||
header.format.byte_per_sample = (short)(bits * channels / 8);
|
||||
header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
|
||||
header.format.sample_rate = (int)sampleRate;
|
||||
|
||||
// fill in the 'fact' part...
|
||||
memcpy(&(header.fact.fact_field), factStr, 4);
|
||||
header.fact.fact_len = 4;
|
||||
header.fact.fact_sample_len = 0;
|
||||
|
||||
// fill in the 'data' part..
|
||||
|
||||
// copy string 'data' to data_field
|
||||
memcpy(&(header.data.data_field), dataStr, 4);
|
||||
// data_len unknown so far
|
||||
header.data.data_len = 0;
|
||||
}
|
||||
|
||||
|
||||
void WavOutFile::finishHeader()
|
||||
{
|
||||
// supplement the file length into the header structure
|
||||
header.riff.package_len = bytesWritten + sizeof(WavHeader) - sizeof(WavRiff) + 4;
|
||||
header.data.data_len = bytesWritten;
|
||||
header.fact.fact_sample_len = bytesWritten / header.format.byte_per_sample;
|
||||
|
||||
writeHeader();
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::writeHeader()
|
||||
{
|
||||
WavHeader hdrTemp;
|
||||
int res;
|
||||
|
||||
// swap byte order if necessary
|
||||
hdrTemp = header;
|
||||
_swap32((int &)hdrTemp.riff.package_len);
|
||||
_swap32((int &)hdrTemp.format.format_len);
|
||||
_swap16((short &)hdrTemp.format.fixed);
|
||||
_swap16((short &)hdrTemp.format.channel_number);
|
||||
_swap32((int &)hdrTemp.format.sample_rate);
|
||||
_swap32((int &)hdrTemp.format.byte_rate);
|
||||
_swap16((short &)hdrTemp.format.byte_per_sample);
|
||||
_swap16((short &)hdrTemp.format.bits_per_sample);
|
||||
_swap32((int &)hdrTemp.data.data_len);
|
||||
_swap32((int &)hdrTemp.fact.fact_len);
|
||||
_swap32((int &)hdrTemp.fact.fact_sample_len);
|
||||
|
||||
// write the supplemented header in the beginning of the file
|
||||
fseek(fptr, 0, SEEK_SET);
|
||||
res = (int)fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
|
||||
if (res != 1)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error while writing to a wav file.");
|
||||
}
|
||||
|
||||
// jump back to the end of the file
|
||||
fseek(fptr, 0, SEEK_END);
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::write(const unsigned char *buffer, int numElems)
|
||||
{
|
||||
int res;
|
||||
|
||||
if (header.format.bits_per_sample != 8)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
|
||||
}
|
||||
assert(sizeof(char) == 1);
|
||||
|
||||
res = (int)fwrite(buffer, 1, numElems, fptr);
|
||||
if (res != numElems)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error while writing to a wav file.");
|
||||
}
|
||||
|
||||
bytesWritten += numElems;
|
||||
}
|
||||
|
||||
|
||||
|
||||
void WavOutFile::write(const short *buffer, int numElems)
|
||||
{
|
||||
int res;
|
||||
|
||||
// 16 bit samples
|
||||
if (numElems < 1) return; // nothing to do
|
||||
|
||||
switch (header.format.bits_per_sample)
|
||||
{
|
||||
case 8:
|
||||
{
|
||||
int i;
|
||||
unsigned char *temp = (unsigned char *)getConvBuffer(numElems);
|
||||
// convert from 16bit format to 8bit format
|
||||
for (i = 0; i < numElems; i ++)
|
||||
{
|
||||
temp[i] = (unsigned char)(buffer[i] / 256 + 128);
|
||||
}
|
||||
// write in 8bit format
|
||||
write(temp, numElems);
|
||||
break;
|
||||
}
|
||||
|
||||
case 16:
|
||||
{
|
||||
// 16bit format
|
||||
|
||||
// use temp buffer to swap byte order if necessary
|
||||
short *pTemp = (short *)getConvBuffer(numElems * sizeof(short));
|
||||
memcpy(pTemp, buffer, numElems * 2);
|
||||
_swap16Buffer(pTemp, numElems);
|
||||
|
||||
res = (int)fwrite(pTemp, 2, numElems, fptr);
|
||||
|
||||
if (res != numElems)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error while writing to a wav file.");
|
||||
}
|
||||
bytesWritten += 2 * numElems;
|
||||
break;
|
||||
}
|
||||
|
||||
default:
|
||||
{
|
||||
stringstream ss;
|
||||
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
|
||||
ss << (int)header.format.bits_per_sample;
|
||||
ss << " bit sample format. ";
|
||||
ST_THROW_RT_ERROR(ss.str().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
/// Convert from float to integer and saturate
|
||||
inline int saturate(float fvalue, float minval, float maxval)
|
||||
{
|
||||
if (fvalue > maxval)
|
||||
{
|
||||
fvalue = maxval;
|
||||
}
|
||||
else if (fvalue < minval)
|
||||
{
|
||||
fvalue = minval;
|
||||
}
|
||||
return (int)fvalue;
|
||||
}
|
||||
|
||||
|
||||
void WavOutFile::write(const float *buffer, int numElems)
|
||||
{
|
||||
int numBytes;
|
||||
int bytesPerSample;
|
||||
|
||||
if (numElems == 0) return;
|
||||
|
||||
bytesPerSample = header.format.bits_per_sample / 8;
|
||||
numBytes = numElems * bytesPerSample;
|
||||
short *temp = (short*)getConvBuffer(numBytes);
|
||||
|
||||
switch (bytesPerSample)
|
||||
{
|
||||
case 1:
|
||||
{
|
||||
unsigned char *temp2 = (unsigned char *)temp;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 2:
|
||||
{
|
||||
short *temp2 = (short *)temp;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
|
||||
temp2[i] = _swap16(value);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 3:
|
||||
{
|
||||
char *temp2 = (char *)temp;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
|
||||
*((int*)temp2) = _swap32(value);
|
||||
temp2 += 3;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
case 4:
|
||||
{
|
||||
int *temp2 = (int *)temp;
|
||||
for (int i = 0; i < numElems; i ++)
|
||||
{
|
||||
int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
|
||||
temp2[i] = _swap32(value);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
default:
|
||||
assert(false);
|
||||
}
|
||||
|
||||
int res = (int)fwrite(temp, 1, numBytes, fptr);
|
||||
|
||||
if (res != numBytes)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error while writing to a wav file.");
|
||||
}
|
||||
bytesWritten += numBytes;
|
||||
}
|
||||
|
|
@ -4,10 +4,10 @@
|
|||
///
|
||||
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
|
||||
/// parse the WAV files with such processors.
|
||||
///
|
||||
/// Admittingly, more complete WAV reader routines may exist in public domain, but
|
||||
///
|
||||
/// Admittingly, more complete WAV reader routines may exist in public domain, but
|
||||
/// the reason for 'yet another' one is that those generic WAV reader libraries are
|
||||
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
|
||||
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
|
||||
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
|
|
@ -16,10 +16,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
|
||||
// Last changed : $Date: 2014-10-05 16:20:24 +0000 (Sun, 05 Oct 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: WavFile.h 63 2009-02-21 16:00:14Z oparviai $
|
||||
// $Id: WavFile.h 200 2014-10-05 16:20:24Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -51,11 +51,11 @@
|
|||
|
||||
#ifndef uint
|
||||
typedef unsigned int uint;
|
||||
#endif
|
||||
#endif
|
||||
|
||||
|
||||
/// WAV audio file 'riff' section header
|
||||
typedef struct
|
||||
typedef struct
|
||||
{
|
||||
char riff_char[4];
|
||||
int package_len;
|
||||
|
|
@ -63,7 +63,7 @@ typedef struct
|
|||
} WavRiff;
|
||||
|
||||
/// WAV audio file 'format' section header
|
||||
typedef struct
|
||||
typedef struct
|
||||
{
|
||||
char fmt[4];
|
||||
int format_len;
|
||||
|
|
@ -75,8 +75,16 @@ typedef struct
|
|||
short bits_per_sample;
|
||||
} WavFormat;
|
||||
|
||||
/// WAV audio file 'fact' section header
|
||||
typedef struct
|
||||
{
|
||||
char fact_field[4];
|
||||
int fact_len;
|
||||
uint fact_sample_len;
|
||||
} WavFact;
|
||||
|
||||
/// WAV audio file 'data' section header
|
||||
typedef struct
|
||||
typedef struct
|
||||
{
|
||||
char data_field[4];
|
||||
uint data_len;
|
||||
|
|
@ -84,23 +92,44 @@ typedef struct
|
|||
|
||||
|
||||
/// WAV audio file header
|
||||
typedef struct
|
||||
typedef struct
|
||||
{
|
||||
WavRiff riff;
|
||||
WavFormat format;
|
||||
WavFact fact;
|
||||
WavData data;
|
||||
} WavHeader;
|
||||
|
||||
|
||||
/// Base class for processing WAV audio files.
|
||||
class WavFileBase
|
||||
{
|
||||
private:
|
||||
/// Conversion working buffer;
|
||||
char *convBuff;
|
||||
int convBuffSize;
|
||||
|
||||
protected:
|
||||
WavFileBase();
|
||||
virtual ~WavFileBase();
|
||||
|
||||
/// Get pointer to conversion buffer of at min. given size
|
||||
void *getConvBuffer(int sizeByte);
|
||||
};
|
||||
|
||||
|
||||
/// Class for reading WAV audio files.
|
||||
class WavInFile
|
||||
class WavInFile : protected WavFileBase
|
||||
{
|
||||
private:
|
||||
/// File pointer.
|
||||
FILE *fptr;
|
||||
|
||||
/// Position within the audio stream
|
||||
long position;
|
||||
|
||||
/// Counter of how many bytes of sample data have been read from the file.
|
||||
uint dataRead;
|
||||
long dataRead;
|
||||
|
||||
/// WAV header information
|
||||
WavHeader header;
|
||||
|
|
@ -142,7 +171,7 @@ public:
|
|||
/// Get number of bits per sample, i.e. 8 or 16.
|
||||
uint getNumBits() const;
|
||||
|
||||
/// Get sample data size in bytes. Ahem, this should return same information as
|
||||
/// Get sample data size in bytes. Ahem, this should return same information as
|
||||
/// 'getBytesPerSample'...
|
||||
uint getDataSizeInBytes() const;
|
||||
|
||||
|
|
@ -151,22 +180,27 @@ public:
|
|||
|
||||
/// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
|
||||
uint getBytesPerSample() const;
|
||||
|
||||
|
||||
/// Get number of audio channels in the file (1=mono, 2=stereo)
|
||||
uint getNumChannels() const;
|
||||
|
||||
/// Get the audio file length in milliseconds
|
||||
uint getLengthMS() const;
|
||||
|
||||
/// Returns how many milliseconds of audio have so far been read from the file
|
||||
///
|
||||
/// \return elapsed duration in milliseconds
|
||||
uint getElapsedMS() const;
|
||||
|
||||
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
|
||||
/// Reads given number of elements from the file or if end-of-file reached, as many
|
||||
/// Reads given number of elements from the file or if end-of-file reached, as many
|
||||
/// elements as are left in the file.
|
||||
///
|
||||
/// \return Number of 8-bit integers read from the file.
|
||||
int read(char *buffer, int maxElems);
|
||||
int read(unsigned char *buffer, int maxElems);
|
||||
|
||||
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
|
||||
/// of elements from the file or if end-of-file reached, as many elements as are
|
||||
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
|
||||
/// of elements from the file or if end-of-file reached, as many elements as are
|
||||
/// left in the file.
|
||||
///
|
||||
/// \return Number of 16-bit integers read from the file.
|
||||
|
|
@ -174,9 +208,10 @@ public:
|
|||
int maxElems ///< Size of 'buffer' array (number of array elements).
|
||||
);
|
||||
|
||||
/// Reads audio samples from the WAV file to floating point format, converting
|
||||
/// Reads audio samples from the WAV file to floating point format, converting
|
||||
/// sample values to range [-1,1[. Reads given number of elements from the file
|
||||
/// or if end-of-file reached, as many elements as are left in the file.
|
||||
/// Notice that reading in float format supports 8/16/24/32bit sample formats.
|
||||
///
|
||||
/// \return Number of elements read from the file.
|
||||
int read(float *buffer, ///< Pointer to buffer where to read data.
|
||||
|
|
@ -192,7 +227,7 @@ public:
|
|||
|
||||
|
||||
/// Class for writing WAV audio files.
|
||||
class WavOutFile
|
||||
class WavOutFile : protected WavFileBase
|
||||
{
|
||||
private:
|
||||
/// Pointer to the WAV file
|
||||
|
|
@ -215,7 +250,7 @@ private:
|
|||
void writeHeader();
|
||||
|
||||
public:
|
||||
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
|
||||
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
|
||||
/// if file creation fails.
|
||||
WavOutFile(const char *fileName, ///< Filename
|
||||
int sampleRate, ///< Sample rate (e.g. 44100 etc)
|
||||
|
|
@ -228,10 +263,10 @@ public:
|
|||
/// Destructor: Finalizes & closes the WAV file.
|
||||
~WavOutFile();
|
||||
|
||||
/// Write data to WAV file. This function works only with 8bit samples.
|
||||
/// Write data to WAV file. This function works only with 8bit samples.
|
||||
/// Throws a 'runtime_error' exception if writing to file fails.
|
||||
void write(const char *buffer, ///< Pointer to sample data buffer.
|
||||
int numElems ///< How many array items are to be written to file.
|
||||
void write(const unsigned char *buffer, ///< Pointer to sample data buffer.
|
||||
int numElems ///< How many array items are to be written to file.
|
||||
);
|
||||
|
||||
/// Write data to WAV file. Throws a 'runtime_error' exception if writing to
|
||||
|
|
@ -12,10 +12,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2009-01-11 09:34:24 -0200 (dom, 11 jan 2009) $
|
||||
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
|
||||
// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -52,6 +52,30 @@ using namespace soundtouch;
|
|||
#define PI 3.141592655357989
|
||||
#define TWOPI (2 * PI)
|
||||
|
||||
// define this to save AA filter coefficients to a file
|
||||
// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1
|
||||
|
||||
#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
|
||||
#include <stdio.h>
|
||||
|
||||
static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
|
||||
{
|
||||
FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
|
||||
if (fptr == NULL) return;
|
||||
|
||||
for (int i = 0; i < len; i ++)
|
||||
{
|
||||
double temp = coeffs[i];
|
||||
fprintf(fptr, "%lf\n", temp);
|
||||
}
|
||||
fclose(fptr);
|
||||
}
|
||||
|
||||
#else
|
||||
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
|
||||
#endif
|
||||
|
||||
|
||||
/*****************************************************************************
|
||||
*
|
||||
* Implementation of the class 'AAFilter'
|
||||
|
|
@ -99,7 +123,7 @@ void AAFilter::calculateCoeffs()
|
|||
{
|
||||
uint i;
|
||||
double cntTemp, temp, tempCoeff,h, w;
|
||||
double fc2, wc;
|
||||
double wc;
|
||||
double scaleCoeff, sum;
|
||||
double *work;
|
||||
SAMPLETYPE *coeffs;
|
||||
|
|
@ -112,8 +136,7 @@ void AAFilter::calculateCoeffs()
|
|||
work = new double[length];
|
||||
coeffs = new SAMPLETYPE[length];
|
||||
|
||||
fc2 = 2.0 * cutoffFreq;
|
||||
wc = PI * fc2;
|
||||
wc = 2.0 * PI * cutoffFreq;
|
||||
tempCoeff = TWOPI / (double)length;
|
||||
|
||||
sum = 0;
|
||||
|
|
@ -124,7 +147,7 @@ void AAFilter::calculateCoeffs()
|
|||
temp = cntTemp * wc;
|
||||
if (temp != 0)
|
||||
{
|
||||
h = fc2 * sin(temp) / temp; // sinc function
|
||||
h = sin(temp) / temp; // sinc function
|
||||
}
|
||||
else
|
||||
{
|
||||
|
|
@ -153,17 +176,21 @@ void AAFilter::calculateCoeffs()
|
|||
|
||||
for (i = 0; i < length; i ++)
|
||||
{
|
||||
// scale & round to nearest integer
|
||||
temp = work[i] * scaleCoeff;
|
||||
//#if SOUNDTOUCH_INTEGER_SAMPLES
|
||||
// scale & round to nearest integer
|
||||
temp += (temp >= 0) ? 0.5 : -0.5;
|
||||
// ensure no overfloods
|
||||
assert(temp >= -32768 && temp <= 32767);
|
||||
//#endif
|
||||
coeffs[i] = (SAMPLETYPE)temp;
|
||||
}
|
||||
|
||||
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
|
||||
pFIR->setCoefficients(coeffs, length, 14);
|
||||
|
||||
_DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
|
||||
|
||||
delete[] work;
|
||||
delete[] coeffs;
|
||||
}
|
||||
|
|
@ -178,6 +205,31 @@ uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples
|
|||
}
|
||||
|
||||
|
||||
/// Applies the filter to the given src & dest pipes, so that processed amount of
|
||||
/// samples get removed from src, and produced amount added to dest
|
||||
/// Note : The amount of outputted samples is by value of 'filter length'
|
||||
/// smaller than the amount of input samples.
|
||||
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
|
||||
{
|
||||
SAMPLETYPE *pdest;
|
||||
const SAMPLETYPE *psrc;
|
||||
uint numSrcSamples;
|
||||
uint result;
|
||||
int numChannels = src.getChannels();
|
||||
|
||||
assert(numChannels == dest.getChannels());
|
||||
|
||||
numSrcSamples = src.numSamples();
|
||||
psrc = src.ptrBegin();
|
||||
pdest = dest.ptrEnd(numSrcSamples);
|
||||
result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
|
||||
src.receiveSamples(result);
|
||||
dest.putSamples(result);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
|
||||
uint AAFilter::getLength() const
|
||||
{
|
||||
return pFIR->getLength();
|
||||
|
|
@ -13,10 +13,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2008-02-10 14:26:55 -0200 (dom, 10 fev 2008) $
|
||||
// Last changed : $Date: 2014-01-07 19:41:23 +0000 (Tue, 07 Jan 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
|
||||
// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -45,6 +45,7 @@
|
|||
#define AAFilter_H
|
||||
|
||||
#include "STTypes.h"
|
||||
#include "FIFOSampleBuffer.h"
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
|
|
@ -84,6 +85,14 @@ public:
|
|||
const SAMPLETYPE *src,
|
||||
uint numSamples,
|
||||
uint numChannels) const;
|
||||
|
||||
/// Applies the filter to the given src & dest pipes, so that processed amount of
|
||||
/// samples get removed from src, and produced amount added to dest
|
||||
/// Note : The amount of outputted samples is by value of 'filter length'
|
||||
/// smaller than the amount of input samples.
|
||||
uint evaluate(FIFOSampleBuffer &dest,
|
||||
FIFOSampleBuffer &src) const;
|
||||
|
||||
};
|
||||
|
||||
}
|
||||
|
|
@ -26,10 +26,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-08-30 16:45:25 -0300 (qui, 30 ago 2012) $
|
||||
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z oparviai $
|
||||
// $Id: BPMDetect.cpp 202 2015-02-21 21:24:29Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -226,6 +226,7 @@ void BPMDetect::updateXCorr(int process_samples)
|
|||
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
|
||||
|
||||
pBuffer = buffer->ptrBegin();
|
||||
#pragma omp parallel for
|
||||
for (offs = windowStart; offs < windowLen; offs ++)
|
||||
{
|
||||
LONG_SAMPLETYPE sum;
|
||||
|
|
@ -15,7 +15,7 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
|
||||
// Last changed : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
|
|
@ -11,10 +11,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
|
||||
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $
|
||||
// $Id: FIRFilter.cpp 202 2015-02-21 21:24:29Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -72,8 +72,7 @@ FIRFilter::~FIRFilter()
|
|||
// Usual C-version of the filter routine for stereo sound
|
||||
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
|
||||
{
|
||||
uint i, j, end;
|
||||
LONG_SAMPLETYPE suml, sumr;
|
||||
int j, end;
|
||||
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
// when using floating point samples, use a scaler instead of a divider
|
||||
// because division is much slower operation than multiplying.
|
||||
|
|
@ -87,9 +86,12 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
|
|||
|
||||
end = 2 * (numSamples - length);
|
||||
|
||||
#pragma omp parallel for
|
||||
for (j = 0; j < end; j += 2)
|
||||
{
|
||||
const SAMPLETYPE *ptr;
|
||||
LONG_SAMPLETYPE suml, sumr;
|
||||
uint i;
|
||||
|
||||
suml = sumr = 0;
|
||||
ptr = src + j;
|
||||
|
|
@ -130,28 +132,31 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
|
|||
// Usual C-version of the filter routine for mono sound
|
||||
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
|
||||
{
|
||||
uint i, j, end;
|
||||
LONG_SAMPLETYPE sum;
|
||||
int j, end;
|
||||
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
// when using floating point samples, use a scaler instead of a divider
|
||||
// because division is much slower operation than multiplying.
|
||||
double dScaler = 1.0 / (double)resultDivider;
|
||||
#endif
|
||||
|
||||
|
||||
assert(length != 0);
|
||||
|
||||
end = numSamples - length;
|
||||
#pragma omp parallel for
|
||||
for (j = 0; j < end; j ++)
|
||||
{
|
||||
const SAMPLETYPE *pSrc = src + j;
|
||||
LONG_SAMPLETYPE sum;
|
||||
uint i;
|
||||
|
||||
sum = 0;
|
||||
for (i = 0; i < length; i += 4)
|
||||
{
|
||||
// loop is unrolled by factor of 4 here for efficiency
|
||||
sum += src[i + 0] * filterCoeffs[i + 0] +
|
||||
src[i + 1] * filterCoeffs[i + 1] +
|
||||
src[i + 2] * filterCoeffs[i + 2] +
|
||||
src[i + 3] * filterCoeffs[i + 3];
|
||||
sum += pSrc[i + 0] * filterCoeffs[i + 0] +
|
||||
pSrc[i + 1] * filterCoeffs[i + 1] +
|
||||
pSrc[i + 2] * filterCoeffs[i + 2] +
|
||||
pSrc[i + 3] * filterCoeffs[i + 3];
|
||||
}
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
sum >>= resultDivFactor;
|
||||
|
|
@ -161,12 +166,67 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
|
|||
sum *= dScaler;
|
||||
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
||||
dest[j] = (SAMPLETYPE)sum;
|
||||
src ++;
|
||||
}
|
||||
return end;
|
||||
}
|
||||
|
||||
|
||||
uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
|
||||
{
|
||||
int j, end;
|
||||
|
||||
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
// when using floating point samples, use a scaler instead of a divider
|
||||
// because division is much slower operation than multiplying.
|
||||
double dScaler = 1.0 / (double)resultDivider;
|
||||
#endif
|
||||
|
||||
assert(length != 0);
|
||||
assert(src != NULL);
|
||||
assert(dest != NULL);
|
||||
assert(filterCoeffs != NULL);
|
||||
assert(numChannels < 16);
|
||||
|
||||
end = numChannels * (numSamples - length);
|
||||
|
||||
#pragma omp parallel for
|
||||
for (j = 0; j < end; j += numChannels)
|
||||
{
|
||||
const SAMPLETYPE *ptr;
|
||||
LONG_SAMPLETYPE sums[16];
|
||||
uint c, i;
|
||||
|
||||
for (c = 0; c < numChannels; c ++)
|
||||
{
|
||||
sums[c] = 0;
|
||||
}
|
||||
|
||||
ptr = src + j;
|
||||
|
||||
for (i = 0; i < length; i ++)
|
||||
{
|
||||
SAMPLETYPE coef=filterCoeffs[i];
|
||||
for (c = 0; c < numChannels; c ++)
|
||||
{
|
||||
sums[c] += ptr[0] * coef;
|
||||
ptr ++;
|
||||
}
|
||||
}
|
||||
|
||||
for (c = 0; c < numChannels; c ++)
|
||||
{
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
sums[c] >>= resultDivFactor;
|
||||
#else
|
||||
sums[c] *= dScaler;
|
||||
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
||||
dest[j+c] = (SAMPLETYPE)sums[c];
|
||||
}
|
||||
}
|
||||
return numSamples - length;
|
||||
}
|
||||
|
||||
|
||||
// Set filter coeffiecients and length.
|
||||
//
|
||||
// Throws an exception if filter length isn't divisible by 8
|
||||
|
|
@ -199,18 +259,27 @@ uint FIRFilter::getLength() const
|
|||
//
|
||||
// Note : The amount of outputted samples is by value of 'filter_length'
|
||||
// smaller than the amount of input samples.
|
||||
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
|
||||
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
|
||||
{
|
||||
assert(numChannels == 1 || numChannels == 2);
|
||||
|
||||
assert(length > 0);
|
||||
assert(lengthDiv8 * 8 == length);
|
||||
|
||||
if (numSamples < length) return 0;
|
||||
if (numChannels == 2)
|
||||
|
||||
#ifndef USE_MULTICH_ALWAYS
|
||||
if (numChannels == 1)
|
||||
{
|
||||
return evaluateFilterMono(dest, src, numSamples);
|
||||
}
|
||||
else if (numChannels == 2)
|
||||
{
|
||||
return evaluateFilterStereo(dest, src, numSamples);
|
||||
} else {
|
||||
return evaluateFilterMono(dest, src, numSamples);
|
||||
}
|
||||
else
|
||||
#endif // USE_MULTICH_ALWAYS
|
||||
{
|
||||
assert(numChannels > 0);
|
||||
return evaluateFilterMulti(dest, src, numSamples, numChannels);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -11,10 +11,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-02-13 17:13:57 -0200 (dom, 13 fev 2011) $
|
||||
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $
|
||||
// $Id: FIRFilter.h 202 2015-02-21 21:24:29Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -71,6 +71,7 @@ protected:
|
|||
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) const;
|
||||
virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
|
||||
|
||||
public:
|
||||
FIRFilter();
|
||||
|
|
@ -90,7 +91,7 @@ public:
|
|||
uint evaluate(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples,
|
||||
uint numChannels) const;
|
||||
uint numChannels);
|
||||
|
||||
uint getLength() const;
|
||||
|
||||
|
|
@ -0,0 +1,200 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Cubic interpolation routine.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <stddef.h>
|
||||
#include <math.h>
|
||||
#include "InterpolateCubic.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
// cubic interpolation coefficients
|
||||
static const float _coeffs[]=
|
||||
{ -0.5f, 1.0f, -0.5f, 0.0f,
|
||||
1.5f, -2.5f, 0.0f, 1.0f,
|
||||
-1.5f, 2.0f, 0.5f, 0.0f,
|
||||
0.5f, -0.5f, 0.0f, 0.0f};
|
||||
|
||||
|
||||
InterpolateCubic::InterpolateCubic()
|
||||
{
|
||||
fract = 0;
|
||||
}
|
||||
|
||||
|
||||
void InterpolateCubic::resetRegisters()
|
||||
{
|
||||
fract = 0;
|
||||
}
|
||||
|
||||
|
||||
/// Transpose mono audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 4;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
float out;
|
||||
const float x3 = 1.0f;
|
||||
const float x2 = (float)fract; // x
|
||||
const float x1 = x2*x2; // x^2
|
||||
const float x0 = x1*x2; // x^3
|
||||
float y0, y1, y2, y3;
|
||||
|
||||
assert(fract < 1.0);
|
||||
|
||||
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
|
||||
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
|
||||
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
|
||||
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
|
||||
|
||||
out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
|
||||
|
||||
pdest[i] = (SAMPLETYPE)out;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
psrc += whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
/// Transpose stereo audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 4;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
const float x3 = 1.0f;
|
||||
const float x2 = (float)fract; // x
|
||||
const float x1 = x2*x2; // x^2
|
||||
const float x0 = x1*x2; // x^3
|
||||
float y0, y1, y2, y3;
|
||||
float out0, out1;
|
||||
|
||||
assert(fract < 1.0);
|
||||
|
||||
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
|
||||
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
|
||||
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
|
||||
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
|
||||
|
||||
out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
|
||||
out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
|
||||
|
||||
pdest[2*i] = (SAMPLETYPE)out0;
|
||||
pdest[2*i+1] = (SAMPLETYPE)out1;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
psrc += 2*whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
/// Transpose multi-channel audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 4;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
const float x3 = 1.0f;
|
||||
const float x2 = (float)fract; // x
|
||||
const float x1 = x2*x2; // x^2
|
||||
const float x0 = x1*x2; // x^3
|
||||
float y0, y1, y2, y3;
|
||||
|
||||
assert(fract < 1.0);
|
||||
|
||||
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
|
||||
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
|
||||
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
|
||||
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
|
||||
|
||||
for (int c = 0; c < numChannels; c ++)
|
||||
{
|
||||
float out;
|
||||
out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
|
||||
pdest[0] = (SAMPLETYPE)out;
|
||||
pdest ++;
|
||||
}
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
psrc += numChannels*whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
|
@ -0,0 +1,67 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Cubic interpolation routine.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateCubic.h 179 2014-01-06 18:41:42Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#ifndef _InterpolateCubic_H_
|
||||
#define _InterpolateCubic_H_
|
||||
|
||||
#include "RateTransposer.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
|
||||
class InterpolateCubic : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
virtual void resetRegisters();
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeMulti(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
|
||||
float fract;
|
||||
|
||||
public:
|
||||
InterpolateCubic();
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif
|
||||
|
|
@ -0,0 +1,299 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Linear interpolation algorithm.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateLinear.cpp 180 2014-01-06 19:16:02Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <assert.h>
|
||||
#include <stdlib.h>
|
||||
#include "InterpolateLinear.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// InterpolateLinearInteger - integer arithmetic implementation
|
||||
//
|
||||
|
||||
/// fixed-point interpolation routine precision
|
||||
#define SCALE 65536
|
||||
|
||||
|
||||
// Constructor
|
||||
InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
resetRegisters();
|
||||
setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
void InterpolateLinearInteger::resetRegisters()
|
||||
{
|
||||
iFract = 0;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
LONG_SAMPLETYPE temp;
|
||||
|
||||
assert(iFract < SCALE);
|
||||
|
||||
temp = (SCALE - iFract) * src[0] + iFract * src[1];
|
||||
dest[i] = (SAMPLETYPE)(temp / SCALE);
|
||||
i++;
|
||||
|
||||
iFract += iRate;
|
||||
|
||||
int iWhole = iFract / SCALE;
|
||||
iFract -= iWhole * SCALE;
|
||||
srcCount += iWhole;
|
||||
src += iWhole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Stereo' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
LONG_SAMPLETYPE temp0;
|
||||
LONG_SAMPLETYPE temp1;
|
||||
|
||||
assert(iFract < SCALE);
|
||||
|
||||
temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
|
||||
temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
|
||||
dest[0] = (SAMPLETYPE)(temp0 / SCALE);
|
||||
dest[1] = (SAMPLETYPE)(temp1 / SCALE);
|
||||
dest += 2;
|
||||
i++;
|
||||
|
||||
iFract += iRate;
|
||||
|
||||
int iWhole = iFract / SCALE;
|
||||
iFract -= iWhole * SCALE;
|
||||
srcCount += iWhole;
|
||||
src += 2*iWhole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
|
||||
assert(iFract < SCALE);
|
||||
vol1 = (SCALE - iFract);
|
||||
for (int c = 0; c < numChannels; c ++)
|
||||
{
|
||||
temp = vol1 * src[c] + iFract * src[c + numChannels];
|
||||
dest[0] = (SAMPLETYPE)(temp / SCALE);
|
||||
dest ++;
|
||||
}
|
||||
i++;
|
||||
|
||||
iFract += iRate;
|
||||
|
||||
int iWhole = iFract / SCALE;
|
||||
iFract -= iWhole * SCALE;
|
||||
srcCount += iWhole;
|
||||
src += iWhole * numChannels;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void InterpolateLinearInteger::setRate(float newRate)
|
||||
{
|
||||
iRate = (int)(newRate * SCALE + 0.5f);
|
||||
TransposerBase::setRate(newRate);
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// InterpolateLinearFloat - floating point arithmetic implementation
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
// Constructor
|
||||
InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
resetRegisters();
|
||||
setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
void InterpolateLinearFloat::resetRegisters()
|
||||
{
|
||||
fract = 0;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
double out;
|
||||
assert(fract < 1.0);
|
||||
|
||||
out = (1.0 - fract) * src[0] + fract * src[1];
|
||||
dest[i] = (SAMPLETYPE)out;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
src += whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// 'Mono' version of the routine. Returns the number of samples returned in
|
||||
// the "dest" buffer
|
||||
int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
double out0, out1;
|
||||
assert(fract < 1.0);
|
||||
|
||||
out0 = (1.0 - fract) * src[0] + fract * src[2];
|
||||
out1 = (1.0 - fract) * src[1] + fract * src[3];
|
||||
dest[2*i] = (SAMPLETYPE)out0;
|
||||
dest[2*i+1] = (SAMPLETYPE)out1;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
src += 2*whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 1;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
float temp, vol1;
|
||||
|
||||
vol1 = (1.0f- fract);
|
||||
for (int c = 0; c < numChannels; c ++)
|
||||
{
|
||||
temp = vol1 * src[c] + fract * src[c + numChannels];
|
||||
*dest = (SAMPLETYPE)temp;
|
||||
dest ++;
|
||||
}
|
||||
i++;
|
||||
|
||||
fract += rate;
|
||||
|
||||
int iWhole = (int)fract;
|
||||
fract -= iWhole;
|
||||
srcCount += iWhole;
|
||||
src += iWhole * numChannels;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
|
||||
return i;
|
||||
}
|
||||
|
|
@ -0,0 +1,92 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Linear interpolation routine.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateLinear.h 179 2014-01-06 18:41:42Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#ifndef _InterpolateLinear_H_
|
||||
#define _InterpolateLinear_H_
|
||||
|
||||
#include "RateTransposer.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
|
||||
/// Linear transposer class that uses integer arithmetics
|
||||
class InterpolateLinearInteger : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
int iFract;
|
||||
int iRate;
|
||||
|
||||
virtual void resetRegisters();
|
||||
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
|
||||
public:
|
||||
InterpolateLinearInteger();
|
||||
|
||||
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
|
||||
/// rate, larger faster rates.
|
||||
virtual void setRate(float newRate);
|
||||
};
|
||||
|
||||
|
||||
/// Linear transposer class that uses floating point arithmetics
|
||||
class InterpolateLinearFloat : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
float fract;
|
||||
|
||||
virtual void resetRegisters();
|
||||
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
|
||||
|
||||
public:
|
||||
InterpolateLinearFloat();
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif
|
||||
|
|
@ -0,0 +1,185 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
|
||||
/// with kaiser window.
|
||||
///
|
||||
/// Notice. This algorithm is remarkably much heavier than linear or cubic
|
||||
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
|
||||
/// for experimental purposes
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <math.h>
|
||||
#include "InterpolateShannon.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
|
||||
/// Kaiser window with beta = 2.0
|
||||
/// Values scaled down by 5% to avoid overflows
|
||||
static const double _kaiser8[8] =
|
||||
{
|
||||
0.41778693317814,
|
||||
0.64888025049173,
|
||||
0.83508562409944,
|
||||
0.93887857733412,
|
||||
0.93887857733412,
|
||||
0.83508562409944,
|
||||
0.64888025049173,
|
||||
0.41778693317814
|
||||
};
|
||||
|
||||
|
||||
InterpolateShannon::InterpolateShannon()
|
||||
{
|
||||
fract = 0;
|
||||
}
|
||||
|
||||
|
||||
void InterpolateShannon::resetRegisters()
|
||||
{
|
||||
fract = 0;
|
||||
}
|
||||
|
||||
|
||||
#define PI 3.1415926536
|
||||
#define sinc(x) (sin(PI * (x)) / (PI * (x)))
|
||||
|
||||
/// Transpose mono audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateShannon::transposeMono(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 8;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
double out;
|
||||
assert(fract < 1.0);
|
||||
|
||||
out = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
|
||||
out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
|
||||
out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
|
||||
if (fract < 1e-6)
|
||||
{
|
||||
out += psrc[3] * _kaiser8[3]; // sinc(0) = 1
|
||||
}
|
||||
else
|
||||
{
|
||||
out += psrc[3] * sinc(- fract) * _kaiser8[3];
|
||||
}
|
||||
out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
|
||||
out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
|
||||
out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
|
||||
out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
|
||||
|
||||
pdest[i] = (SAMPLETYPE)out;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
psrc += whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
/// Transpose stereo audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
int i;
|
||||
int srcSampleEnd = srcSamples - 8;
|
||||
int srcCount = 0;
|
||||
|
||||
i = 0;
|
||||
while (srcCount < srcSampleEnd)
|
||||
{
|
||||
double out0, out1, w;
|
||||
assert(fract < 1.0);
|
||||
|
||||
w = sinc(-3.0 - fract) * _kaiser8[0];
|
||||
out0 = psrc[0] * w; out1 = psrc[1] * w;
|
||||
w = sinc(-2.0 - fract) * _kaiser8[1];
|
||||
out0 += psrc[2] * w; out1 += psrc[3] * w;
|
||||
w = sinc(-1.0 - fract) * _kaiser8[2];
|
||||
out0 += psrc[4] * w; out1 += psrc[5] * w;
|
||||
w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract)); // sinc(0) = 1
|
||||
out0 += psrc[6] * w; out1 += psrc[7] * w;
|
||||
w = sinc( 1.0 - fract) * _kaiser8[4];
|
||||
out0 += psrc[8] * w; out1 += psrc[9] * w;
|
||||
w = sinc( 2.0 - fract) * _kaiser8[5];
|
||||
out0 += psrc[10] * w; out1 += psrc[11] * w;
|
||||
w = sinc( 3.0 - fract) * _kaiser8[6];
|
||||
out0 += psrc[12] * w; out1 += psrc[13] * w;
|
||||
w = sinc( 4.0 - fract) * _kaiser8[7];
|
||||
out0 += psrc[14] * w; out1 += psrc[15] * w;
|
||||
|
||||
pdest[2*i] = (SAMPLETYPE)out0;
|
||||
pdest[2*i+1] = (SAMPLETYPE)out1;
|
||||
i ++;
|
||||
|
||||
// update position fraction
|
||||
fract += rate;
|
||||
// update whole positions
|
||||
int whole = (int)fract;
|
||||
fract -= whole;
|
||||
psrc += 2*whole;
|
||||
srcCount += whole;
|
||||
}
|
||||
srcSamples = srcCount;
|
||||
return i;
|
||||
}
|
||||
|
||||
|
||||
/// Transpose stereo audio. Returns number of produced output samples, and
|
||||
/// updates "srcSamples" to amount of consumed source samples
|
||||
int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest,
|
||||
const SAMPLETYPE *psrc,
|
||||
int &srcSamples)
|
||||
{
|
||||
// not implemented
|
||||
assert(false);
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -0,0 +1,72 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
|
||||
/// with kaiser window.
|
||||
///
|
||||
/// Notice. This algorithm is remarkably much heavier than linear or cubic
|
||||
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
|
||||
/// for experimental purposes
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// $Id: InterpolateShannon.h 179 2014-01-06 18:41:42Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#ifndef _InterpolateShannon_H_
|
||||
#define _InterpolateShannon_H_
|
||||
|
||||
#include "RateTransposer.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
|
||||
class InterpolateShannon : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
void resetRegisters();
|
||||
int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
int transposeMulti(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples);
|
||||
|
||||
float fract;
|
||||
|
||||
public:
|
||||
InterpolateShannon();
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif
|
||||
|
|
@ -11,10 +11,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-12-28 17:52:47 -0200 (sex, 28 dez 2012) $
|
||||
// Last changed : $Date: 2015-05-18 15:22:02 +0000 (Mon, 18 May 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $
|
||||
// $Id: PeakFinder.cpp 213 2015-05-18 15:22:02Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -192,11 +192,21 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
|
|||
gp1 = findGround(data, peakpos, -1);
|
||||
gp2 = findGround(data, peakpos, 1);
|
||||
|
||||
groundLevel = 0.5f * (data[gp1] + data[gp2]);
|
||||
peakLevel = data[peakpos];
|
||||
|
||||
// calculate 70%-level of the peak
|
||||
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
|
||||
if (gp1 == gp2)
|
||||
{
|
||||
// avoid rounding errors when all are equal
|
||||
assert(gp1 == peakpos);
|
||||
cutLevel = groundLevel = peakLevel;
|
||||
} else {
|
||||
// get average of the ground levels
|
||||
groundLevel = 0.5f * (data[gp1] + data[gp2]);
|
||||
|
||||
// calculate 70%-level of the peak
|
||||
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
|
||||
}
|
||||
|
||||
// find mid-level crossings
|
||||
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
|
||||
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
|
||||
|
|
@ -9,7 +9,7 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2011-12-30 18:33:46 -0200 (sex, 30 dez 2011) $
|
||||
// Last changed : $Date: 2011-12-30 20:33:46 +0000 (Fri, 30 Dec 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $
|
||||
|
|
@ -0,0 +1,302 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Sample rate transposer. Changes sample rate by using linear interpolation
|
||||
/// together with anti-alias filtering (first order interpolation with anti-
|
||||
/// alias filtering should be quite adequate for this application)
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <memory.h>
|
||||
#include <assert.h>
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include "RateTransposer.h"
|
||||
#include "InterpolateLinear.h"
|
||||
#include "InterpolateCubic.h"
|
||||
#include "InterpolateShannon.h"
|
||||
#include "AAFilter.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
// Define default interpolation algorithm here
|
||||
TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
|
||||
|
||||
|
||||
// Constructor
|
||||
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
|
||||
{
|
||||
bUseAAFilter = true;
|
||||
|
||||
// Instantiates the anti-alias filter
|
||||
pAAFilter = new AAFilter(64);
|
||||
pTransposer = TransposerBase::newInstance();
|
||||
}
|
||||
|
||||
|
||||
|
||||
RateTransposer::~RateTransposer()
|
||||
{
|
||||
delete pAAFilter;
|
||||
delete pTransposer;
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
|
||||
void RateTransposer::enableAAFilter(bool newMode)
|
||||
{
|
||||
bUseAAFilter = newMode;
|
||||
}
|
||||
|
||||
|
||||
/// Returns nonzero if anti-alias filter is enabled.
|
||||
bool RateTransposer::isAAFilterEnabled() const
|
||||
{
|
||||
return bUseAAFilter;
|
||||
}
|
||||
|
||||
|
||||
AAFilter *RateTransposer::getAAFilter()
|
||||
{
|
||||
return pAAFilter;
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void RateTransposer::setRate(float newRate)
|
||||
{
|
||||
double fCutoff;
|
||||
|
||||
pTransposer->setRate(newRate);
|
||||
|
||||
// design a new anti-alias filter
|
||||
if (newRate > 1.0f)
|
||||
{
|
||||
fCutoff = 0.5f / newRate;
|
||||
}
|
||||
else
|
||||
{
|
||||
fCutoff = 0.5f * newRate;
|
||||
}
|
||||
pAAFilter->setCutoffFreq(fCutoff);
|
||||
}
|
||||
|
||||
|
||||
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
|
||||
// the input of the object.
|
||||
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
||||
{
|
||||
processSamples(samples, nSamples);
|
||||
}
|
||||
|
||||
|
||||
// Transposes sample rate by applying anti-alias filter to prevent folding.
|
||||
// Returns amount of samples returned in the "dest" buffer.
|
||||
// The maximum amount of samples that can be returned at a time is set by
|
||||
// the 'set_returnBuffer_size' function.
|
||||
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count;
|
||||
|
||||
if (nSamples == 0) return;
|
||||
|
||||
// Store samples to input buffer
|
||||
inputBuffer.putSamples(src, nSamples);
|
||||
|
||||
// If anti-alias filter is turned off, simply transpose without applying
|
||||
// the filter
|
||||
if (bUseAAFilter == false)
|
||||
{
|
||||
count = pTransposer->transpose(outputBuffer, inputBuffer);
|
||||
return;
|
||||
}
|
||||
|
||||
assert(pAAFilter);
|
||||
|
||||
// Transpose with anti-alias filter
|
||||
if (pTransposer->rate < 1.0f)
|
||||
{
|
||||
// If the parameter 'Rate' value is smaller than 1, first transpose
|
||||
// the samples and then apply the anti-alias filter to remove aliasing.
|
||||
|
||||
// Transpose the samples, store the result to end of "midBuffer"
|
||||
pTransposer->transpose(midBuffer, inputBuffer);
|
||||
|
||||
// Apply the anti-alias filter for transposed samples in midBuffer
|
||||
pAAFilter->evaluate(outputBuffer, midBuffer);
|
||||
}
|
||||
else
|
||||
{
|
||||
// If the parameter 'Rate' value is larger than 1, first apply the
|
||||
// anti-alias filter to remove high frequencies (prevent them from folding
|
||||
// over the lover frequencies), then transpose.
|
||||
|
||||
// Apply the anti-alias filter for samples in inputBuffer
|
||||
pAAFilter->evaluate(midBuffer, inputBuffer);
|
||||
|
||||
// Transpose the AA-filtered samples in "midBuffer"
|
||||
pTransposer->transpose(outputBuffer, midBuffer);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Sets the number of channels, 1 = mono, 2 = stereo
|
||||
void RateTransposer::setChannels(int nChannels)
|
||||
{
|
||||
assert(nChannels > 0);
|
||||
|
||||
if (pTransposer->numChannels == nChannels) return;
|
||||
pTransposer->setChannels(nChannels);
|
||||
|
||||
inputBuffer.setChannels(nChannels);
|
||||
midBuffer.setChannels(nChannels);
|
||||
outputBuffer.setChannels(nChannels);
|
||||
}
|
||||
|
||||
|
||||
// Clears all the samples in the object
|
||||
void RateTransposer::clear()
|
||||
{
|
||||
outputBuffer.clear();
|
||||
midBuffer.clear();
|
||||
inputBuffer.clear();
|
||||
}
|
||||
|
||||
|
||||
// Returns nonzero if there aren't any samples available for outputting.
|
||||
int RateTransposer::isEmpty() const
|
||||
{
|
||||
int res;
|
||||
|
||||
res = FIFOProcessor::isEmpty();
|
||||
if (res == 0) return 0;
|
||||
return inputBuffer.isEmpty();
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// TransposerBase - Base class for interpolation
|
||||
//
|
||||
|
||||
// static function to set interpolation algorithm
|
||||
void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
|
||||
{
|
||||
TransposerBase::algorithm = a;
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// Returns the number of samples returned in the "dest" buffer
|
||||
int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
|
||||
{
|
||||
int numSrcSamples = src.numSamples();
|
||||
int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
|
||||
int numOutput;
|
||||
SAMPLETYPE *psrc = src.ptrBegin();
|
||||
SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
|
||||
|
||||
#ifndef USE_MULTICH_ALWAYS
|
||||
if (numChannels == 1)
|
||||
{
|
||||
numOutput = transposeMono(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
else if (numChannels == 2)
|
||||
{
|
||||
numOutput = transposeStereo(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
else
|
||||
#endif // USE_MULTICH_ALWAYS
|
||||
{
|
||||
assert(numChannels > 0);
|
||||
numOutput = transposeMulti(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
dest.putSamples(numOutput);
|
||||
src.receiveSamples(numSrcSamples);
|
||||
return numOutput;
|
||||
}
|
||||
|
||||
|
||||
TransposerBase::TransposerBase()
|
||||
{
|
||||
numChannels = 0;
|
||||
rate = 1.0f;
|
||||
}
|
||||
|
||||
|
||||
TransposerBase::~TransposerBase()
|
||||
{
|
||||
}
|
||||
|
||||
|
||||
void TransposerBase::setChannels(int channels)
|
||||
{
|
||||
numChannels = channels;
|
||||
resetRegisters();
|
||||
}
|
||||
|
||||
|
||||
void TransposerBase::setRate(float newRate)
|
||||
{
|
||||
rate = newRate;
|
||||
}
|
||||
|
||||
|
||||
// static factory function
|
||||
TransposerBase *TransposerBase::newInstance()
|
||||
{
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
// Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
|
||||
return ::new InterpolateLinearInteger;
|
||||
#else
|
||||
switch (algorithm)
|
||||
{
|
||||
case LINEAR:
|
||||
return new InterpolateLinearFloat;
|
||||
|
||||
case CUBIC:
|
||||
return new InterpolateCubic;
|
||||
|
||||
case SHANNON:
|
||||
return new InterpolateShannon;
|
||||
|
||||
default:
|
||||
assert(false);
|
||||
return NULL;
|
||||
}
|
||||
#endif
|
||||
}
|
||||
|
|
@ -14,10 +14,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2009-02-21 13:00:14 -0300 (sáb, 21 fev 2009) $
|
||||
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
|
||||
// $Id: RateTransposer.h 195 2014-04-06 15:57:21Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -55,50 +55,71 @@
|
|||
namespace soundtouch
|
||||
{
|
||||
|
||||
/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
|
||||
class TransposerBase
|
||||
{
|
||||
public:
|
||||
enum ALGORITHM {
|
||||
LINEAR = 0,
|
||||
CUBIC,
|
||||
SHANNON
|
||||
};
|
||||
|
||||
protected:
|
||||
virtual void resetRegisters() = 0;
|
||||
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples) = 0;
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples) = 0;
|
||||
virtual int transposeMulti(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples) = 0;
|
||||
|
||||
static ALGORITHM algorithm;
|
||||
|
||||
public:
|
||||
float rate;
|
||||
int numChannels;
|
||||
|
||||
TransposerBase();
|
||||
virtual ~TransposerBase();
|
||||
|
||||
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
|
||||
virtual void setRate(float newRate);
|
||||
virtual void setChannels(int channels);
|
||||
|
||||
// static factory function
|
||||
static TransposerBase *newInstance();
|
||||
|
||||
// static function to set interpolation algorithm
|
||||
static void setAlgorithm(ALGORITHM a);
|
||||
};
|
||||
|
||||
|
||||
/// A common linear samplerate transposer class.
|
||||
///
|
||||
/// Note: Use function "RateTransposer::newInstance()" to create a new class
|
||||
/// instance instead of the "new" operator; that function automatically
|
||||
/// chooses a correct implementation depending on if integer or floating
|
||||
/// arithmetics are to be used.
|
||||
class RateTransposer : public FIFOProcessor
|
||||
{
|
||||
protected:
|
||||
/// Anti-alias filter object
|
||||
AAFilter *pAAFilter;
|
||||
|
||||
float fRate;
|
||||
|
||||
int numChannels;
|
||||
TransposerBase *pTransposer;
|
||||
|
||||
/// Buffer for collecting samples to feed the anti-alias filter between
|
||||
/// two batches
|
||||
FIFOSampleBuffer storeBuffer;
|
||||
FIFOSampleBuffer inputBuffer;
|
||||
|
||||
/// Buffer for keeping samples between transposing & anti-alias filter
|
||||
FIFOSampleBuffer tempBuffer;
|
||||
FIFOSampleBuffer midBuffer;
|
||||
|
||||
/// Output sample buffer
|
||||
FIFOSampleBuffer outputBuffer;
|
||||
|
||||
BOOL bUseAAFilter;
|
||||
bool bUseAAFilter;
|
||||
|
||||
virtual void resetRegisters() = 0;
|
||||
|
||||
virtual uint transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
virtual uint transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
inline uint transpose(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
void downsample(const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
void upsample(const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
/// Transposes sample rate by applying anti-alias filter to prevent folding.
|
||||
/// Returns amount of samples returned in the "dest" buffer.
|
||||
|
|
@ -107,34 +128,33 @@ protected:
|
|||
void processSamples(const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
|
||||
public:
|
||||
RateTransposer();
|
||||
virtual ~RateTransposer();
|
||||
|
||||
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
|
||||
/// depending on if we're to use integer or floating point arithmetics.
|
||||
static void *operator new(size_t s);
|
||||
// static void *operator new(size_t s);
|
||||
|
||||
/// Use this function instead of "new" operator to create a new instance of this class.
|
||||
/// This function automatically chooses a correct implementation, depending on if
|
||||
/// integer ot floating point arithmetics are to be used.
|
||||
static RateTransposer *newInstance();
|
||||
// static RateTransposer *newInstance();
|
||||
|
||||
/// Returns the output buffer object
|
||||
FIFOSamplePipe *getOutput() { return &outputBuffer; };
|
||||
|
||||
/// Returns the store buffer object
|
||||
FIFOSamplePipe *getStore() { return &storeBuffer; };
|
||||
// FIFOSamplePipe *getStore() { return &storeBuffer; };
|
||||
|
||||
/// Return anti-alias filter object
|
||||
AAFilter *getAAFilter();
|
||||
|
||||
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
|
||||
void enableAAFilter(BOOL newMode);
|
||||
void enableAAFilter(bool newMode);
|
||||
|
||||
/// Returns nonzero if anti-alias filter is enabled.
|
||||
BOOL isAAFilterEnabled() const;
|
||||
bool isAAFilterEnabled() const;
|
||||
|
||||
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
|
||||
/// rate, larger faster rates.
|
||||
|
|
@ -41,10 +41,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
|
||||
// Last changed : $Date: 2014-10-08 15:26:57 +0000 (Wed, 08 Oct 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: SoundTouch.cpp 143 2012-06-13 19:29:53Z oparviai $
|
||||
// $Id: SoundTouch.cpp 201 2014-10-08 15:26:57Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -97,7 +97,7 @@ SoundTouch::SoundTouch()
|
|||
{
|
||||
// Initialize rate transposer and tempo changer instances
|
||||
|
||||
pRateTransposer = RateTransposer::newInstance();
|
||||
pRateTransposer = new RateTransposer();
|
||||
pTDStretch = TDStretch::newInstance();
|
||||
|
||||
setOutPipe(pTDStretch);
|
||||
|
|
@ -111,7 +111,7 @@ SoundTouch::SoundTouch()
|
|||
calcEffectiveRateAndTempo();
|
||||
|
||||
channels = 0;
|
||||
bSrateSet = FALSE;
|
||||
bSrateSet = false;
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -143,10 +143,11 @@ uint SoundTouch::getVersionId()
|
|||
// Sets the number of channels, 1 = mono, 2 = stereo
|
||||
void SoundTouch::setChannels(uint numChannels)
|
||||
{
|
||||
if (numChannels != 1 && numChannels != 2)
|
||||
/*if (numChannels != 1 && numChannels != 2)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Illegal number of channels");
|
||||
}
|
||||
//ST_THROW_RT_ERROR("Illegal number of channels");
|
||||
return;
|
||||
}*/
|
||||
channels = numChannels;
|
||||
pRateTransposer->setChannels((int)numChannels);
|
||||
pTDStretch->setChannels((int)numChannels);
|
||||
|
|
@ -254,7 +255,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
|
|||
tempoOut = pTDStretch->getOutput();
|
||||
tempoOut->moveSamples(*output);
|
||||
// move samples in pitch transposer's store buffer to tempo changer's input
|
||||
pTDStretch->moveSamples(*pRateTransposer->getStore());
|
||||
// deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
|
||||
|
||||
output = pTDStretch;
|
||||
}
|
||||
|
|
@ -282,7 +283,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
|
|||
// Sets sample rate.
|
||||
void SoundTouch::setSampleRate(uint srate)
|
||||
{
|
||||
bSrateSet = TRUE;
|
||||
bSrateSet = true;
|
||||
// set sample rate, leave other tempo changer parameters as they are.
|
||||
pTDStretch->setParameters((int)srate);
|
||||
}
|
||||
|
|
@ -292,7 +293,7 @@ void SoundTouch::setSampleRate(uint srate)
|
|||
// the input of the object.
|
||||
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
||||
{
|
||||
if (bSrateSet == FALSE)
|
||||
if (bSrateSet == false)
|
||||
{
|
||||
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
|
||||
}
|
||||
|
|
@ -347,8 +348,8 @@ void SoundTouch::flush()
|
|||
int i;
|
||||
int nUnprocessed;
|
||||
int nOut;
|
||||
SAMPLETYPE buff[64*2]; // note: allocate 2*64 to cater 64 sample frames of stereo sound
|
||||
|
||||
SAMPLETYPE *buff = new SAMPLETYPE[64 * channels];
|
||||
|
||||
// check how many samples still await processing, and scale
|
||||
// that by tempo & rate to get expected output sample count
|
||||
nUnprocessed = numUnprocessedSamples();
|
||||
|
|
@ -377,6 +378,8 @@ void SoundTouch::flush()
|
|||
}
|
||||
}
|
||||
|
||||
delete[] buff;
|
||||
|
||||
// Clear working buffers
|
||||
pRateTransposer->clear();
|
||||
pTDStretch->clearInput();
|
||||
|
|
@ -387,7 +390,7 @@ void SoundTouch::flush()
|
|||
|
||||
// Changes a setting controlling the processing system behaviour. See the
|
||||
// 'SETTING_...' defines for available setting ID's.
|
||||
BOOL SoundTouch::setSetting(int settingId, int value)
|
||||
bool SoundTouch::setSetting(int settingId, int value)
|
||||
{
|
||||
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
|
||||
|
||||
|
|
@ -398,36 +401,36 @@ BOOL SoundTouch::setSetting(int settingId, int value)
|
|||
{
|
||||
case SETTING_USE_AA_FILTER :
|
||||
// enables / disabless anti-alias filter
|
||||
pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
|
||||
return TRUE;
|
||||
pRateTransposer->enableAAFilter((value != 0) ? true : false);
|
||||
return true;
|
||||
|
||||
case SETTING_AA_FILTER_LENGTH :
|
||||
// sets anti-alias filter length
|
||||
pRateTransposer->getAAFilter()->setLength(value);
|
||||
return TRUE;
|
||||
return true;
|
||||
|
||||
case SETTING_USE_QUICKSEEK :
|
||||
// enables / disables tempo routine quick seeking algorithm
|
||||
pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
|
||||
return TRUE;
|
||||
pTDStretch->enableQuickSeek((value != 0) ? true : false);
|
||||
return true;
|
||||
|
||||
case SETTING_SEQUENCE_MS:
|
||||
// change time-stretch sequence duration parameter
|
||||
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
|
||||
return TRUE;
|
||||
return true;
|
||||
|
||||
case SETTING_SEEKWINDOW_MS:
|
||||
// change time-stretch seek window length parameter
|
||||
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
|
||||
return TRUE;
|
||||
return true;
|
||||
|
||||
case SETTING_OVERLAP_MS:
|
||||
// change time-stretch overlap length parameter
|
||||
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
|
||||
return TRUE;
|
||||
return true;
|
||||
|
||||
default :
|
||||
return FALSE;
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -13,10 +13,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
|
||||
// Last changed : $Date: 2015-02-22 15:07:12 +0000 (Sun, 22 Feb 2015) $
|
||||
// File revision : $Revision: 1.12 $
|
||||
//
|
||||
// $Id: TDStretch.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
// $Id: TDStretch.cpp 205 2015-02-22 15:07:12Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -51,8 +51,6 @@
|
|||
#include "cpu_detect.h"
|
||||
#include "TDStretch.h"
|
||||
|
||||
#include <stdio.h>
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
#define max(x, y) (((x) > (y)) ? (x) : (y))
|
||||
|
|
@ -86,15 +84,15 @@ static const short _scanOffsets[5][24]={
|
|||
|
||||
TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
|
||||
{
|
||||
bQuickSeek = FALSE;
|
||||
bQuickSeek = false;
|
||||
channels = 2;
|
||||
|
||||
pMidBuffer = NULL;
|
||||
pMidBufferUnaligned = NULL;
|
||||
overlapLength = 0;
|
||||
|
||||
bAutoSeqSetting = TRUE;
|
||||
bAutoSeekSetting = TRUE;
|
||||
bAutoSeqSetting = true;
|
||||
bAutoSeekSetting = true;
|
||||
|
||||
// outDebt = 0;
|
||||
skipFract = 0;
|
||||
|
|
@ -134,23 +132,23 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
|
|||
if (aSequenceMS > 0)
|
||||
{
|
||||
this->sequenceMs = aSequenceMS;
|
||||
bAutoSeqSetting = FALSE;
|
||||
bAutoSeqSetting = false;
|
||||
}
|
||||
else if (aSequenceMS == 0)
|
||||
{
|
||||
// if zero, use automatic setting
|
||||
bAutoSeqSetting = TRUE;
|
||||
bAutoSeqSetting = true;
|
||||
}
|
||||
|
||||
if (aSeekWindowMS > 0)
|
||||
{
|
||||
this->seekWindowMs = aSeekWindowMS;
|
||||
bAutoSeekSetting = FALSE;
|
||||
bAutoSeekSetting = false;
|
||||
}
|
||||
else if (aSeekWindowMS == 0)
|
||||
{
|
||||
// if zero, use automatic setting
|
||||
bAutoSeekSetting = TRUE;
|
||||
bAutoSeekSetting = true;
|
||||
}
|
||||
|
||||
calcSeqParameters();
|
||||
|
|
@ -159,7 +157,6 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
|
|||
|
||||
// set tempo to recalculate 'sampleReq'
|
||||
setTempo(tempo);
|
||||
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -212,7 +209,7 @@ void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
|
|||
|
||||
void TDStretch::clearMidBuffer()
|
||||
{
|
||||
memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
|
||||
memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength);
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -234,14 +231,14 @@ void TDStretch::clear()
|
|||
|
||||
// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
|
||||
// to enable
|
||||
void TDStretch::enableQuickSeek(BOOL enable)
|
||||
void TDStretch::enableQuickSeek(bool enable)
|
||||
{
|
||||
bQuickSeek = enable;
|
||||
}
|
||||
|
||||
|
||||
// Returns nonzero if the quick seeking algorithm is enabled.
|
||||
BOOL TDStretch::isQuickSeekEnabled() const
|
||||
bool TDStretch::isQuickSeekEnabled() const
|
||||
{
|
||||
return bQuickSeek;
|
||||
}
|
||||
|
|
@ -265,13 +262,22 @@ int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
|
|||
// of 'ovlPos'.
|
||||
inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
|
||||
{
|
||||
if (channels == 2)
|
||||
#ifndef USE_MULTICH_ALWAYS
|
||||
if (channels == 1)
|
||||
{
|
||||
// mono sound.
|
||||
overlapMono(pOutput, pInput + ovlPos);
|
||||
}
|
||||
else if (channels == 2)
|
||||
{
|
||||
// stereo sound
|
||||
overlapStereo(pOutput, pInput + 2 * ovlPos);
|
||||
} else {
|
||||
// mono sound.
|
||||
overlapMono(pOutput, pInput + ovlPos);
|
||||
}
|
||||
else
|
||||
#endif // USE_MULTICH_ALWAYS
|
||||
{
|
||||
assert(channels > 0);
|
||||
overlapMulti(pOutput, pInput + channels * ovlPos);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -286,19 +292,32 @@ inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, ui
|
|||
int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
|
||||
{
|
||||
int bestOffs;
|
||||
double bestCorr, corr;
|
||||
double bestCorr;
|
||||
int i;
|
||||
double norm;
|
||||
|
||||
bestCorr = FLT_MIN;
|
||||
bestOffs = 0;
|
||||
|
||||
// Scans for the best correlation value by testing each possible position
|
||||
// over the permitted range.
|
||||
for (i = 0; i < seekLength; i ++)
|
||||
bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
|
||||
|
||||
#pragma omp parallel for
|
||||
for (i = 1; i < seekLength; i ++)
|
||||
{
|
||||
// Calculates correlation value for the mixing position corresponding
|
||||
// to 'i'
|
||||
corr = calcCrossCorr(refPos + channels * i, pMidBuffer);
|
||||
double corr;
|
||||
// Calculates correlation value for the mixing position corresponding to 'i'
|
||||
#ifdef _OPENMP
|
||||
// in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
|
||||
// iterate the loop in sequential order
|
||||
corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
|
||||
#else
|
||||
// In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
|
||||
// as "calcCrossCorr", but saves time by reusing & updating previously stored
|
||||
// "norm" value
|
||||
corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm);
|
||||
#endif
|
||||
// heuristic rule to slightly favour values close to mid of the range
|
||||
double tmp = (double)(2 * i - seekLength) / (double)seekLength;
|
||||
corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
|
||||
|
|
@ -306,8 +325,15 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
|
|||
// Checks for the highest correlation value
|
||||
if (corr > bestCorr)
|
||||
{
|
||||
bestCorr = corr;
|
||||
bestOffs = i;
|
||||
// For optimal performance, enter critical section only in case that best value found.
|
||||
// in such case repeat 'if' condition as it's possible that parallel execution may have
|
||||
// updated the bestCorr value in the mean time
|
||||
#pragma omp critical
|
||||
if (corr > bestCorr)
|
||||
{
|
||||
bestCorr = corr;
|
||||
bestOffs = i;
|
||||
}
|
||||
}
|
||||
}
|
||||
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
|
||||
|
|
@ -346,12 +372,13 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
|
|||
j = 0;
|
||||
while (_scanOffsets[scanCount][j])
|
||||
{
|
||||
double norm;
|
||||
tempOffset = corrOffset + _scanOffsets[scanCount][j];
|
||||
if (tempOffset >= seekLength) break;
|
||||
|
||||
// Calculates correlation value for the mixing position corresponding
|
||||
// to 'tempOffset'
|
||||
corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer);
|
||||
corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer, norm);
|
||||
// heuristic rule to slightly favour values close to mid of the range
|
||||
double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
|
||||
corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
|
||||
|
|
@ -458,11 +485,15 @@ void TDStretch::setChannels(int numChannels)
|
|||
{
|
||||
assert(numChannels > 0);
|
||||
if (channels == numChannels) return;
|
||||
assert(numChannels == 1 || numChannels == 2);
|
||||
// assert(numChannels == 1 || numChannels == 2);
|
||||
|
||||
channels = numChannels;
|
||||
inputBuffer.setChannels(channels);
|
||||
outputBuffer.setChannels(channels);
|
||||
|
||||
// re-init overlap/buffer
|
||||
overlapLength=0;
|
||||
setParameters(sampleRate);
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -498,7 +529,6 @@ void TDStretch::processNominalTempo()
|
|||
}
|
||||
*/
|
||||
|
||||
#include <stdio.h>
|
||||
|
||||
// Processes as many processing frames of the samples 'inputBuffer', store
|
||||
// the result into 'outputBuffer'
|
||||
|
|
@ -588,7 +618,7 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
|
|||
{
|
||||
delete[] pMidBufferUnaligned;
|
||||
|
||||
pMidBufferUnaligned = new SAMPLETYPE[overlapLength * 2 + 16 / sizeof(SAMPLETYPE)];
|
||||
pMidBufferUnaligned = new SAMPLETYPE[overlapLength * channels + 16 / sizeof(SAMPLETYPE)];
|
||||
// ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency
|
||||
pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned);
|
||||
|
||||
|
|
@ -666,6 +696,27 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
|
|||
}
|
||||
}
|
||||
|
||||
|
||||
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
|
||||
// version of the routine.
|
||||
void TDStretch::overlapMulti(SAMPLETYPE *poutput, const SAMPLETYPE *input) const
|
||||
{
|
||||
SAMPLETYPE m1=(SAMPLETYPE)0;
|
||||
SAMPLETYPE m2;
|
||||
int i=0;
|
||||
|
||||
for (m2 = (SAMPLETYPE)overlapLength; m2; m2 --)
|
||||
{
|
||||
for (int c = 0; c < channels; c ++)
|
||||
{
|
||||
poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength;
|
||||
i++;
|
||||
}
|
||||
|
||||
m1++;
|
||||
}
|
||||
}
|
||||
|
||||
// Calculates the x having the closest 2^x value for the given value
|
||||
static int _getClosest2Power(double value)
|
||||
{
|
||||
|
|
@ -699,32 +750,72 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
|
|||
}
|
||||
|
||||
|
||||
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare) const
|
||||
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const
|
||||
{
|
||||
long corr;
|
||||
long norm;
|
||||
long lnorm;
|
||||
int i;
|
||||
|
||||
corr = norm = 0;
|
||||
corr = lnorm = 0;
|
||||
// Same routine for stereo and mono. For stereo, unroll loop for better
|
||||
// efficiency and gives slightly better resolution against rounding.
|
||||
// For mono it same routine, just unrolls loop by factor of 4
|
||||
for (i = 0; i < channels * overlapLength; i += 4)
|
||||
{
|
||||
corr += (mixingPos[i] * compare[i] +
|
||||
mixingPos[i + 1] * compare[i + 1] +
|
||||
mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
|
||||
corr += (mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
|
||||
norm += (mixingPos[i] * mixingPos[i] +
|
||||
mixingPos[i + 1] * mixingPos[i + 1] +
|
||||
mixingPos[i + 2] * mixingPos[i + 2] +
|
||||
mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
|
||||
lnorm += (mixingPos[i] * mixingPos[i] +
|
||||
mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
|
||||
lnorm += (mixingPos[i + 2] * mixingPos[i + 2] +
|
||||
mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
|
||||
}
|
||||
|
||||
// Normalize result by dividing by sqrt(norm) - this step is easiest
|
||||
// done using floating point operation
|
||||
if (norm == 0) norm = 1; // to avoid div by zero
|
||||
return (double)corr / sqrt((double)norm);
|
||||
norm = (double)lnorm;
|
||||
return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
|
||||
}
|
||||
|
||||
|
||||
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
|
||||
double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const
|
||||
{
|
||||
long corr;
|
||||
long lnorm;
|
||||
int i;
|
||||
|
||||
// cancel first normalizer tap from previous round
|
||||
lnorm = 0;
|
||||
for (i = 1; i <= channels; i ++)
|
||||
{
|
||||
lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBits;
|
||||
}
|
||||
|
||||
corr = 0;
|
||||
// Same routine for stereo and mono. For stereo, unroll loop for better
|
||||
// efficiency and gives slightly better resolution against rounding.
|
||||
// For mono it same routine, just unrolls loop by factor of 4
|
||||
for (i = 0; i < channels * overlapLength; i += 4)
|
||||
{
|
||||
corr += (mixingPos[i] * compare[i] +
|
||||
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
|
||||
corr += (mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
|
||||
}
|
||||
|
||||
// update normalizer with last samples of this round
|
||||
for (int j = 0; j < channels; j ++)
|
||||
{
|
||||
i --;
|
||||
lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;
|
||||
}
|
||||
norm += (double)lnorm;
|
||||
|
||||
// Normalize result by dividing by sqrt(norm) - this step is easiest
|
||||
// done using floating point operation
|
||||
return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
|
||||
}
|
||||
|
||||
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
||||
|
|
@ -760,6 +851,34 @@ void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
|
|||
}
|
||||
|
||||
|
||||
// Overlaps samples in 'midBuffer' with the samples in 'input'.
|
||||
void TDStretch::overlapMulti(float *pOutput, const float *pInput) const
|
||||
{
|
||||
int i;
|
||||
float fScale;
|
||||
float f1;
|
||||
float f2;
|
||||
|
||||
fScale = 1.0f / (float)overlapLength;
|
||||
|
||||
f1 = 0;
|
||||
f2 = 1.0f;
|
||||
|
||||
i=0;
|
||||
for (int i2 = 0; i2 < overlapLength; i2 ++)
|
||||
{
|
||||
// note: Could optimize this slightly by taking into account that always channels > 2
|
||||
for (int c = 0; c < channels; c ++)
|
||||
{
|
||||
pOutput[i] = pInput[i] * f1 + pMidBuffer[i] * f2;
|
||||
i++;
|
||||
}
|
||||
f1 += fScale;
|
||||
f2 -= fScale;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
/// Calculates overlapInMsec period length in samples.
|
||||
void TDStretch::calculateOverlapLength(int overlapInMsec)
|
||||
{
|
||||
|
|
@ -776,7 +895,8 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
|
|||
}
|
||||
|
||||
|
||||
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) const
|
||||
/// Calculate cross-correlation
|
||||
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm) const
|
||||
{
|
||||
double corr;
|
||||
double norm;
|
||||
|
|
@ -801,8 +921,44 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) co
|
|||
mixingPos[i + 3] * mixingPos[i + 3];
|
||||
}
|
||||
|
||||
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
|
||||
return corr / sqrt(norm);
|
||||
anorm = norm;
|
||||
return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
|
||||
}
|
||||
|
||||
|
||||
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
|
||||
double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const
|
||||
{
|
||||
double corr;
|
||||
int i;
|
||||
|
||||
corr = 0;
|
||||
|
||||
// cancel first normalizer tap from previous round
|
||||
for (i = 1; i <= channels; i ++)
|
||||
{
|
||||
norm -= mixingPos[-i] * mixingPos[-i];
|
||||
}
|
||||
|
||||
// Same routine for stereo and mono. For Stereo, unroll by factor of 2.
|
||||
// For mono it's same routine yet unrollsd by factor of 4.
|
||||
for (i = 0; i < channels * overlapLength; i += 4)
|
||||
{
|
||||
corr += mixingPos[i] * compare[i] +
|
||||
mixingPos[i + 1] * compare[i + 1] +
|
||||
mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 3] * compare[i + 3];
|
||||
}
|
||||
|
||||
// update normalizer with last samples of this round
|
||||
for (int j = 0; j < channels; j ++)
|
||||
{
|
||||
i --;
|
||||
norm += mixingPos[i] * mixingPos[i];
|
||||
}
|
||||
|
||||
return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
|
||||
}
|
||||
|
||||
|
||||
#endif // SOUNDTOUCH_FLOAT_SAMPLES
|
||||
|
|
@ -13,10 +13,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-04-01 16:49:30 -0300 (dom, 01 abr 2012) $
|
||||
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: TDStretch.h 137 2012-04-01 19:49:30Z oparviai $
|
||||
// $Id: TDStretch.h 195 2014-04-06 15:57:21Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -125,21 +125,22 @@ protected:
|
|||
float skipFract;
|
||||
FIFOSampleBuffer outputBuffer;
|
||||
FIFOSampleBuffer inputBuffer;
|
||||
BOOL bQuickSeek;
|
||||
bool bQuickSeek;
|
||||
|
||||
int sampleRate;
|
||||
int sequenceMs;
|
||||
int seekWindowMs;
|
||||
int overlapMs;
|
||||
BOOL bAutoSeqSetting;
|
||||
BOOL bAutoSeekSetting;
|
||||
bool bAutoSeqSetting;
|
||||
bool bAutoSeekSetting;
|
||||
|
||||
void acceptNewOverlapLength(int newOverlapLength);
|
||||
|
||||
virtual void clearCrossCorrState();
|
||||
void calculateOverlapLength(int overlapMs);
|
||||
|
||||
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
|
||||
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
|
||||
virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
|
||||
|
||||
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
|
||||
virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
|
||||
|
|
@ -147,6 +148,7 @@ protected:
|
|||
|
||||
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
|
||||
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
|
||||
virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
|
||||
|
||||
void clearMidBuffer();
|
||||
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
|
||||
|
|
@ -193,10 +195,10 @@ public:
|
|||
|
||||
/// Enables/disables the quick position seeking algorithm. Zero to disable,
|
||||
/// nonzero to enable
|
||||
void enableQuickSeek(BOOL enable);
|
||||
void enableQuickSeek(bool enable);
|
||||
|
||||
/// Returns nonzero if the quick seeking algorithm is enabled.
|
||||
BOOL isQuickSeekEnabled() const;
|
||||
bool isQuickSeekEnabled() const;
|
||||
|
||||
/// Sets routine control parameters. These control are certain time constants
|
||||
/// defining how the sound is stretched to the desired duration.
|
||||
|
|
@ -247,7 +249,8 @@ public:
|
|||
class TDStretchMMX : public TDStretch
|
||||
{
|
||||
protected:
|
||||
double calcCrossCorr(const short *mixingPos, const short *compare) const;
|
||||
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const;
|
||||
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const;
|
||||
virtual void overlapStereo(short *output, const short *input) const;
|
||||
virtual void clearCrossCorrState();
|
||||
};
|
||||
|
|
@ -259,7 +262,8 @@ public:
|
|||
class TDStretchSSE : public TDStretch
|
||||
{
|
||||
protected:
|
||||
double calcCrossCorr(const float *mixingPos, const float *compare) const;
|
||||
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) const;
|
||||
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const;
|
||||
};
|
||||
|
||||
#endif /// SOUNDTOUCH_ALLOW_SSE
|
||||
|
|
@ -12,7 +12,7 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2008-02-10 14:26:55 -0200 (dom, 10 fev 2008) $
|
||||
// Last changed : $Date: 2008-02-10 16:26:55 +0000 (Sun, 10 Feb 2008) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
|
||||
|
|
@ -11,10 +11,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 16:44:37 -0200 (qui, 08 nov 2012) $
|
||||
// Last changed : $Date: 2014-01-07 18:24:28 +0000 (Tue, 07 Jan 2014) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect_x86.cpp 159 2012-11-08 18:44:37Z oparviai $
|
||||
// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -42,19 +42,20 @@
|
|||
#include "cpu_detect.h"
|
||||
#include "STTypes.h"
|
||||
|
||||
|
||||
#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
|
||||
|
||||
#if defined(__GNUC__) && defined(__i386__)
|
||||
// gcc
|
||||
#include "cpuid.h"
|
||||
#elif defined(_M_IX86)
|
||||
// windows non-gcc
|
||||
#include <intrin.h>
|
||||
#define bit_MMX (1 << 23)
|
||||
#define bit_SSE (1 << 25)
|
||||
#define bit_SSE2 (1 << 26)
|
||||
#endif
|
||||
#if defined(__GNUC__) && defined(__i386__)
|
||||
// gcc
|
||||
#include "cpuid.h"
|
||||
#elif defined(_M_IX86)
|
||||
// windows non-gcc
|
||||
#include <intrin.h>
|
||||
#endif
|
||||
|
||||
#define bit_MMX (1 << 23)
|
||||
#define bit_SSE (1 << 25)
|
||||
#define bit_SSE2 (1 << 26)
|
||||
#endif
|
||||
|
||||
|
||||
|
|
@ -20,10 +20,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
|
||||
// Last changed : $Date: 2015-02-22 15:10:38 +0000 (Sun, 22 Feb 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: mmx_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
// $Id: mmx_optimized.cpp 206 2015-02-22 15:10:38Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -68,7 +68,7 @@ using namespace soundtouch;
|
|||
|
||||
|
||||
// Calculates cross correlation of two buffers
|
||||
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
|
||||
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm) const
|
||||
{
|
||||
const __m64 *pVec1, *pVec2;
|
||||
__m64 shifter;
|
||||
|
|
@ -93,19 +93,19 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
|
|||
// _mm_add_pi32 : 2*32bit add
|
||||
// _m_psrad : 32bit right-shift
|
||||
|
||||
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
|
||||
_mm_madd_pi16(pVec1[1], pVec2[1]));
|
||||
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
|
||||
_mm_madd_pi16(pVec1[1], pVec1[1]));
|
||||
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
|
||||
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
|
||||
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
|
||||
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
|
||||
accu = _mm_add_pi32(accu, temp);
|
||||
normaccu = _mm_add_pi32(normaccu, temp2);
|
||||
|
||||
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
|
||||
_mm_madd_pi16(pVec1[3], pVec2[3]));
|
||||
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
|
||||
_mm_madd_pi16(pVec1[3], pVec1[3]));
|
||||
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
|
||||
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
|
||||
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
|
||||
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
|
||||
accu = _mm_add_pi32(accu, temp);
|
||||
normaccu = _mm_add_pi32(normaccu, temp2);
|
||||
|
||||
pVec1 += 4;
|
||||
pVec2 += 4;
|
||||
|
|
@ -125,14 +125,81 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
|
|||
|
||||
// Normalize result by dividing by sqrt(norm) - this step is easiest
|
||||
// done using floating point operation
|
||||
if (norm == 0) norm = 1; // to avoid div by zero
|
||||
dnorm = (double)norm;
|
||||
|
||||
return (double)corr / sqrt((double)norm);
|
||||
return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
|
||||
// Note: Warning about the missing EMMS instruction is harmless
|
||||
// as it'll be called elsewhere.
|
||||
}
|
||||
|
||||
|
||||
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
|
||||
double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm) const
|
||||
{
|
||||
const __m64 *pVec1, *pVec2;
|
||||
__m64 shifter;
|
||||
__m64 accu;
|
||||
long corr, lnorm;
|
||||
int i;
|
||||
|
||||
// cancel first normalizer tap from previous round
|
||||
lnorm = 0;
|
||||
for (i = 1; i <= channels; i ++)
|
||||
{
|
||||
lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBits;
|
||||
}
|
||||
|
||||
pVec1 = (__m64*)pV1;
|
||||
pVec2 = (__m64*)pV2;
|
||||
|
||||
shifter = _m_from_int(overlapDividerBits);
|
||||
accu = _mm_setzero_si64();
|
||||
|
||||
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
|
||||
// during each round for improved CPU-level parallellization.
|
||||
for (i = 0; i < channels * overlapLength / 16; i ++)
|
||||
{
|
||||
__m64 temp;
|
||||
|
||||
// dictionary of instructions:
|
||||
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
|
||||
// _mm_add_pi32 : 2*32bit add
|
||||
// _m_psrad : 32bit right-shift
|
||||
|
||||
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
|
||||
accu = _mm_add_pi32(accu, temp);
|
||||
|
||||
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
|
||||
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
|
||||
accu = _mm_add_pi32(accu, temp);
|
||||
|
||||
pVec1 += 4;
|
||||
pVec2 += 4;
|
||||
}
|
||||
|
||||
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
|
||||
// and finally store the result into the variable "corr"
|
||||
|
||||
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
|
||||
corr = _m_to_int(accu);
|
||||
|
||||
// Clear MMS state
|
||||
_m_empty();
|
||||
|
||||
// update normalizer with last samples of this round
|
||||
pV1 = (short *)pVec1;
|
||||
for (int j = 1; j <= channels; j ++)
|
||||
{
|
||||
lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBits;
|
||||
}
|
||||
dnorm += (double)lnorm;
|
||||
|
||||
// Normalize result by dividing by sqrt(norm) - this step is easiest
|
||||
// done using floating point operation
|
||||
return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
|
||||
}
|
||||
|
||||
|
||||
void TDStretchMMX::clearCrossCorrState()
|
||||
{
|
||||
|
|
@ -220,6 +287,7 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
|
|||
|
||||
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
|
||||
{
|
||||
filterCoeffsAlign = NULL;
|
||||
filterCoeffsUnalign = NULL;
|
||||
}
|
||||
|
||||
|
|
@ -23,10 +23,10 @@
|
|||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
|
||||
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: sse_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
// $Id: sse_optimized.cpp 202 2015-02-21 21:24:29Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
|
|
@ -71,7 +71,7 @@ using namespace soundtouch;
|
|||
#include <math.h>
|
||||
|
||||
// Calculates cross correlation of two buffers
|
||||
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
|
||||
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm) const
|
||||
{
|
||||
int i;
|
||||
const float *pVec1;
|
||||
|
|
@ -141,11 +141,11 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
|
|||
|
||||
// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
|
||||
float *pvNorm = (float*)&vNorm;
|
||||
double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
|
||||
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
|
||||
float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
|
||||
anorm = norm;
|
||||
|
||||
float *pvSum = (float*)&vSum;
|
||||
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
|
||||
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
|
||||
|
||||
/* This is approximately corresponding routine in C-language yet without normalization:
|
||||
double corr, norm;
|
||||
|
|
@ -182,6 +182,16 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
|
|||
}
|
||||
|
||||
|
||||
|
||||
double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm) const
|
||||
{
|
||||
// call usual calcCrossCorr function because SSE does not show big benefit of
|
||||
// accumulating "norm" value, and also the "norm" rolling algorithm would get
|
||||
// complicated due to SSE-specific alignment-vs-nonexact correlation rules.
|
||||
return calcCrossCorr(pV1, pV2, norm);
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// implementation of SSE optimized functions of class 'FIRFilter'
|
||||
|
|
@ -249,14 +259,17 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
|
|||
assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
|
||||
|
||||
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
|
||||
#pragma omp parallel for
|
||||
for (j = 0; j < count; j += 2)
|
||||
{
|
||||
const float *pSrc;
|
||||
float *pDest;
|
||||
const __m128 *pFil;
|
||||
__m128 sum1, sum2;
|
||||
uint i;
|
||||
|
||||
pSrc = (const float*)source; // source audio data
|
||||
pSrc = (const float*)source + j * 2; // source audio data
|
||||
pDest = dest + j * 2; // destination audio data
|
||||
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
|
||||
// are aligned to 16-byte boundary
|
||||
sum1 = sum2 = _mm_setzero_ps();
|
||||
|
|
@ -289,12 +302,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
|
|||
// to sum the two hi- and lo-floats of these registers together.
|
||||
|
||||
// post-shuffle & add the filtered values and store to dest.
|
||||
_mm_storeu_ps(dest, _mm_add_ps(
|
||||
_mm_storeu_ps(pDest, _mm_add_ps(
|
||||
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
|
||||
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
|
||||
));
|
||||
source += 4;
|
||||
dest += 4;
|
||||
}
|
||||
|
||||
// Ideas for further improvement:
|
||||
|
|
@ -63,6 +63,8 @@
|
|||
#define wxCOMPILER_PREFIX vc110
|
||||
#elif _MSC_VER == 1800
|
||||
#define wxCOMPILER_PREFIX vc120
|
||||
#elif _MSC_VER == 1900
|
||||
#define wxCOMPILER_PREFIX vc140
|
||||
#else
|
||||
#error "Unknown MSVC compiler version, please report to wx-dev."
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -53,7 +53,14 @@
|
|||
# define __VISUALC11__
|
||||
#elif __VISUALC__ < 1900
|
||||
# define __VISUALC12__
|
||||
#elif __VISUALC__ < 2000
|
||||
/* There is no __VISUALC13__! */
|
||||
# define __VISUALC14__
|
||||
#else
|
||||
/*
|
||||
Don't forget to update include/msvc/wx/setup.h as well when adding
|
||||
support for a newer MSVC version here.
|
||||
*/
|
||||
# pragma message("Please update wx/compiler.h to recognize this VC++ version")
|
||||
#endif
|
||||
|
||||
|
|
@ -102,7 +109,13 @@
|
|||
# define wxVISUALC_VERSION(major) 0
|
||||
# define wxCHECK_VISUALC_VERSION(major) 0
|
||||
#else
|
||||
# define wxVISUALC_VERSION(major) ( (6 + major) * 100 )
|
||||
/*
|
||||
Things used to be simple with the _MSC_VER value and the version number
|
||||
increasing in lock step, but _MSC_VER value of 1900 is VC14 and not the
|
||||
non existing (presumably for the superstitious reasons) VC13, so we now
|
||||
need to account for this with an extra offset.
|
||||
*/
|
||||
# define wxVISUALC_VERSION(major) ( (6 + (major >= 14 ? 1 : 0) + major) * 100 )
|
||||
# define wxCHECK_VISUALC_VERSION(major) ( __VISUALC__ >= wxVISUALC_VERSION(major) )
|
||||
#endif
|
||||
|
||||
|
|
|
|||
|
|
@ -1,6 +1,69 @@
|
|||
|
||||
ChangeLog file for zlib
|
||||
|
||||
Changes in 1.2.8 (28 Apr 2013)
|
||||
- Update contrib/minizip/iowin32.c for Windows RT [Vollant]
|
||||
- Do not force Z_CONST for C++
|
||||
- Clean up contrib/vstudio [Ro§]
|
||||
- Correct spelling error in zlib.h
|
||||
- Fix mixed line endings in contrib/vstudio
|
||||
|
||||
Changes in 1.2.7.3 (13 Apr 2013)
|
||||
- Fix version numbers and DLL names in contrib/vstudio/*/zlib.rc
|
||||
|
||||
Changes in 1.2.7.2 (13 Apr 2013)
|
||||
- Change check for a four-byte type back to hexadecimal
|
||||
- Fix typo in win32/Makefile.msc
|
||||
- Add casts in gzwrite.c for pointer differences
|
||||
|
||||
Changes in 1.2.7.1 (24 Mar 2013)
|
||||
- Replace use of unsafe string functions with snprintf if available
|
||||
- Avoid including stddef.h on Windows for Z_SOLO compile [Niessink]
|
||||
- Fix gzgetc undefine when Z_PREFIX set [Turk]
|
||||
- Eliminate use of mktemp in Makefile (not always available)
|
||||
- Fix bug in 'F' mode for gzopen()
|
||||
- Add inflateGetDictionary() function
|
||||
- Correct comment in deflate.h
|
||||
- Use _snprintf for snprintf in Microsoft C
|
||||
- On Darwin, only use /usr/bin/libtool if libtool is not Apple
|
||||
- Delete "--version" file if created by "ar --version" [Richard G.]
|
||||
- Fix configure check for veracity of compiler error return codes
|
||||
- Fix CMake compilation of static lib for MSVC2010 x64
|
||||
- Remove unused variable in infback9.c
|
||||
- Fix argument checks in gzlog_compress() and gzlog_write()
|
||||
- Clean up the usage of z_const and respect const usage within zlib
|
||||
- Clean up examples/gzlog.[ch] comparisons of different types
|
||||
- Avoid shift equal to bits in type (caused endless loop)
|
||||
- Fix unintialized value bug in gzputc() introduced by const patches
|
||||
- Fix memory allocation error in examples/zran.c [Nor]
|
||||
- Fix bug where gzopen(), gzclose() would write an empty file
|
||||
- Fix bug in gzclose() when gzwrite() runs out of memory
|
||||
- Check for input buffer malloc failure in examples/gzappend.c
|
||||
- Add note to contrib/blast to use binary mode in stdio
|
||||
- Fix comparisons of differently signed integers in contrib/blast
|
||||
- Check for invalid code length codes in contrib/puff
|
||||
- Fix serious but very rare decompression bug in inftrees.c
|
||||
- Update inflateBack() comments, since inflate() can be faster
|
||||
- Use underscored I/O function names for WINAPI_FAMILY
|
||||
- Add _tr_flush_bits to the external symbols prefixed by --zprefix
|
||||
- Add contrib/vstudio/vc10 pre-build step for static only
|
||||
- Quote --version-script argument in CMakeLists.txt
|
||||
- Don't specify --version-script on Apple platforms in CMakeLists.txt
|
||||
- Fix casting error in contrib/testzlib/testzlib.c
|
||||
- Fix types in contrib/minizip to match result of get_crc_table()
|
||||
- Simplify contrib/vstudio/vc10 with 'd' suffix
|
||||
- Add TOP support to win32/Makefile.msc
|
||||
- Suport i686 and amd64 assembler builds in CMakeLists.txt
|
||||
- Fix typos in the use of _LARGEFILE64_SOURCE in zconf.h
|
||||
- Add vc11 and vc12 build files to contrib/vstudio
|
||||
- Add gzvprintf() as an undocumented function in zlib
|
||||
- Fix configure for Sun shell
|
||||
- Remove runtime check in configure for four-byte integer type
|
||||
- Add casts and consts to ease user conversion to C++
|
||||
- Add man pages for minizip and miniunzip
|
||||
- In Makefile uninstall, don't rm if preceding cd fails
|
||||
- Do not return Z_BUF_ERROR if deflateParam() has nothing to write
|
||||
|
||||
Changes in 1.2.7 (2 May 2012)
|
||||
- Replace use of memmove() with a simple copy for portability
|
||||
- Test for existence of strerror
|
||||
|
|
|
|||
|
|
@ -1,6 +0,0 @@
|
|||
noinst_LIBRARIES = libpcsx2zlib.a
|
||||
|
||||
libpcsx2zlib_a_SOURCES = \
|
||||
adler32.c crc32.c deflate.h inffast.c inflate.c inftrees.h trees.h zlib.h \
|
||||
crc32.h gzio.c inffast.h inflate.h uncompr.c zutil.c \
|
||||
compress.c deflate.c infback.c inffixed.h inftrees.c trees.c zconf.h zutil.h
|
||||
|
|
@ -1,6 +1,6 @@
|
|||
ZLIB DATA COMPRESSION LIBRARY
|
||||
|
||||
zlib 1.2.7 is a general purpose data compression library. All the code is
|
||||
zlib 1.2.8 is a general purpose data compression library. All the code is
|
||||
thread safe. The data format used by the zlib library is described by RFCs
|
||||
(Request for Comments) 1950 to 1952 in the files
|
||||
http://tools.ietf.org/html/rfc1950 (zlib format), rfc1951 (deflate format) and
|
||||
|
|
@ -31,7 +31,7 @@ Mark Nelson <markn@ieee.org> wrote an article about zlib for the Jan. 1997
|
|||
issue of Dr. Dobb's Journal; a copy of the article is available at
|
||||
http://marknelson.us/1997/01/01/zlib-engine/ .
|
||||
|
||||
The changes made in version 1.2.7 are documented in the file ChangeLog.
|
||||
The changes made in version 1.2.8 are documented in the file ChangeLog.
|
||||
|
||||
Unsupported third party contributions are provided in directory contrib/ .
|
||||
|
||||
|
|
@ -84,7 +84,7 @@ Acknowledgments:
|
|||
|
||||
Copyright notice:
|
||||
|
||||
(C) 1995-2012 Jean-loup Gailly and Mark Adler
|
||||
(C) 1995-2013 Jean-loup Gailly and Mark Adler
|
||||
|
||||
This software is provided 'as-is', without any express or implied
|
||||
warranty. In no event will the authors be held liable for any damages
|
||||
|
|
|
|||
|
|
@ -29,7 +29,7 @@ int ZEXPORT compress2 (dest, destLen, source, sourceLen, level)
|
|||
z_stream stream;
|
||||
int err;
|
||||
|
||||
stream.next_in = (Bytef*)source;
|
||||
stream.next_in = (z_const Bytef *)source;
|
||||
stream.avail_in = (uInt)sourceLen;
|
||||
#ifdef MAXSEG_64K
|
||||
/* Check for source > 64K on 16-bit machine: */
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* deflate.c -- compress data using the deflation algorithm
|
||||
* Copyright (C) 1995-2012 Jean-loup Gailly and Mark Adler
|
||||
* Copyright (C) 1995-2013 Jean-loup Gailly and Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -52,7 +52,7 @@
|
|||
#include "deflate.h"
|
||||
|
||||
const char deflate_copyright[] =
|
||||
" deflate 1.2.7 Copyright 1995-2012 Jean-loup Gailly and Mark Adler ";
|
||||
" deflate 1.2.8 Copyright 1995-2013 Jean-loup Gailly and Mark Adler ";
|
||||
/*
|
||||
If you use the zlib library in a product, an acknowledgment is welcome
|
||||
in the documentation of your product. If for some reason you cannot
|
||||
|
|
@ -305,7 +305,7 @@ int ZEXPORT deflateInit2_(strm, level, method, windowBits, memLevel, strategy,
|
|||
if (s->window == Z_NULL || s->prev == Z_NULL || s->head == Z_NULL ||
|
||||
s->pending_buf == Z_NULL) {
|
||||
s->status = FINISH_STATE;
|
||||
strm->msg = (char*)ERR_MSG(Z_MEM_ERROR);
|
||||
strm->msg = ERR_MSG(Z_MEM_ERROR);
|
||||
deflateEnd (strm);
|
||||
return Z_MEM_ERROR;
|
||||
}
|
||||
|
|
@ -329,7 +329,7 @@ int ZEXPORT deflateSetDictionary (strm, dictionary, dictLength)
|
|||
uInt str, n;
|
||||
int wrap;
|
||||
unsigned avail;
|
||||
unsigned char *next;
|
||||
z_const unsigned char *next;
|
||||
|
||||
if (strm == Z_NULL || strm->state == Z_NULL || dictionary == Z_NULL)
|
||||
return Z_STREAM_ERROR;
|
||||
|
|
@ -359,7 +359,7 @@ int ZEXPORT deflateSetDictionary (strm, dictionary, dictLength)
|
|||
avail = strm->avail_in;
|
||||
next = strm->next_in;
|
||||
strm->avail_in = dictLength;
|
||||
strm->next_in = (Bytef *)dictionary;
|
||||
strm->next_in = (z_const Bytef *)dictionary;
|
||||
fill_window(s);
|
||||
while (s->lookahead >= MIN_MATCH) {
|
||||
str = s->strstart;
|
||||
|
|
@ -513,6 +513,8 @@ int ZEXPORT deflateParams(strm, level, strategy)
|
|||
strm->total_in != 0) {
|
||||
/* Flush the last buffer: */
|
||||
err = deflate(strm, Z_BLOCK);
|
||||
if (err == Z_BUF_ERROR && s->pending == 0)
|
||||
err = Z_OK;
|
||||
}
|
||||
if (s->level != level) {
|
||||
s->level = level;
|
||||
|
|
|
|||
|
|
@ -104,7 +104,7 @@ typedef struct internal_state {
|
|||
int wrap; /* bit 0 true for zlib, bit 1 true for gzip */
|
||||
gz_headerp gzhead; /* gzip header information to write */
|
||||
uInt gzindex; /* where in extra, name, or comment */
|
||||
Byte method; /* STORED (for zip only) or DEFLATED */
|
||||
Byte method; /* can only be DEFLATED */
|
||||
int last_flush; /* value of flush param for previous deflate call */
|
||||
|
||||
/* used by deflate.c: */
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* gzguts.h -- zlib internal header definitions for gz* operations
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012 Mark Adler
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012, 2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -35,6 +35,13 @@
|
|||
# include <io.h>
|
||||
#endif
|
||||
|
||||
#ifdef WINAPI_FAMILY
|
||||
# define open _open
|
||||
# define read _read
|
||||
# define write _write
|
||||
# define close _close
|
||||
#endif
|
||||
|
||||
#ifdef NO_DEFLATE /* for compatibility with old definition */
|
||||
# define NO_GZCOMPRESS
|
||||
#endif
|
||||
|
|
@ -60,7 +67,7 @@
|
|||
#ifndef HAVE_VSNPRINTF
|
||||
# ifdef MSDOS
|
||||
/* vsnprintf may exist on some MS-DOS compilers (DJGPP?),
|
||||
but for now we just assume it doesn't. */
|
||||
but for now we just assume it doesn't. */
|
||||
# define NO_vsnprintf
|
||||
# endif
|
||||
# ifdef __TURBOC__
|
||||
|
|
@ -88,6 +95,14 @@
|
|||
# endif
|
||||
#endif
|
||||
|
||||
/* unlike snprintf (which is required in C99, yet still not supported by
|
||||
Microsoft more than a decade later!), _snprintf does not guarantee null
|
||||
termination of the result -- however this is only used in gzlib.c where
|
||||
the result is assured to fit in the space provided */
|
||||
#ifdef _MSC_VER
|
||||
# define snprintf _snprintf
|
||||
#endif
|
||||
|
||||
#ifndef local
|
||||
# define local static
|
||||
#endif
|
||||
|
|
@ -127,7 +142,8 @@
|
|||
# define DEF_MEM_LEVEL MAX_MEM_LEVEL
|
||||
#endif
|
||||
|
||||
/* default i/o buffer size -- double this for output when reading */
|
||||
/* default i/o buffer size -- double this for output when reading (this and
|
||||
twice this must be able to fit in an unsigned type) */
|
||||
#define GZBUFSIZE 8192
|
||||
|
||||
/* gzip modes, also provide a little integrity check on the passed structure */
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* gzlib.c -- zlib functions common to reading and writing gzip files
|
||||
* Copyright (C) 2004, 2010, 2011, 2012 Mark Adler
|
||||
* Copyright (C) 2004, 2010, 2011, 2012, 2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -108,7 +108,7 @@ local gzFile gz_open(path, fd, mode)
|
|||
return NULL;
|
||||
|
||||
/* allocate gzFile structure to return */
|
||||
state = malloc(sizeof(gz_state));
|
||||
state = (gz_statep)malloc(sizeof(gz_state));
|
||||
if (state == NULL)
|
||||
return NULL;
|
||||
state->size = 0; /* no buffers allocated yet */
|
||||
|
|
@ -162,8 +162,10 @@ local gzFile gz_open(path, fd, mode)
|
|||
break;
|
||||
case 'F':
|
||||
state->strategy = Z_FIXED;
|
||||
break;
|
||||
case 'T':
|
||||
state->direct = 1;
|
||||
break;
|
||||
default: /* could consider as an error, but just ignore */
|
||||
;
|
||||
}
|
||||
|
|
@ -194,8 +196,8 @@ local gzFile gz_open(path, fd, mode)
|
|||
}
|
||||
else
|
||||
#endif
|
||||
len = strlen(path);
|
||||
state->path = malloc(len + 1);
|
||||
len = strlen((const char *)path);
|
||||
state->path = (char *)malloc(len + 1);
|
||||
if (state->path == NULL) {
|
||||
free(state);
|
||||
return NULL;
|
||||
|
|
@ -208,7 +210,11 @@ local gzFile gz_open(path, fd, mode)
|
|||
*(state->path) = 0;
|
||||
else
|
||||
#endif
|
||||
#if !defined(NO_snprintf) && !defined(NO_vsnprintf)
|
||||
snprintf(state->path, len + 1, "%s", (const char *)path);
|
||||
#else
|
||||
strcpy(state->path, path);
|
||||
#endif
|
||||
|
||||
/* compute the flags for open() */
|
||||
oflag =
|
||||
|
|
@ -236,7 +242,7 @@ local gzFile gz_open(path, fd, mode)
|
|||
#ifdef _WIN32
|
||||
fd == -2 ? _wopen(path, oflag, 0666) :
|
||||
#endif
|
||||
open(path, oflag, 0666));
|
||||
open((const char *)path, oflag, 0666));
|
||||
if (state->fd == -1) {
|
||||
free(state->path);
|
||||
free(state);
|
||||
|
|
@ -282,9 +288,13 @@ gzFile ZEXPORT gzdopen(fd, mode)
|
|||
char *path; /* identifier for error messages */
|
||||
gzFile gz;
|
||||
|
||||
if (fd == -1 || (path = malloc(7 + 3 * sizeof(int))) == NULL)
|
||||
if (fd == -1 || (path = (char *)malloc(7 + 3 * sizeof(int))) == NULL)
|
||||
return NULL;
|
||||
#if !defined(NO_snprintf) && !defined(NO_vsnprintf)
|
||||
snprintf(path, 7 + 3 * sizeof(int), "<fd:%d>", fd); /* for debugging */
|
||||
#else
|
||||
sprintf(path, "<fd:%d>", fd); /* for debugging */
|
||||
#endif
|
||||
gz = gz_open(path, fd, mode);
|
||||
free(path);
|
||||
return gz;
|
||||
|
|
@ -531,7 +541,8 @@ const char * ZEXPORT gzerror(file, errnum)
|
|||
/* return error information */
|
||||
if (errnum != NULL)
|
||||
*errnum = state->err;
|
||||
return state->msg == NULL ? "" : state->msg;
|
||||
return state->err == Z_MEM_ERROR ? "out of memory" :
|
||||
(state->msg == NULL ? "" : state->msg);
|
||||
}
|
||||
|
||||
/* -- see zlib.h -- */
|
||||
|
|
@ -582,21 +593,24 @@ void ZLIB_INTERNAL gz_error(state, err, msg)
|
|||
if (msg == NULL)
|
||||
return;
|
||||
|
||||
/* for an out of memory error, save as static string */
|
||||
if (err == Z_MEM_ERROR) {
|
||||
state->msg = (char *)msg;
|
||||
/* for an out of memory error, return literal string when requested */
|
||||
if (err == Z_MEM_ERROR)
|
||||
return;
|
||||
}
|
||||
|
||||
/* construct error message with path */
|
||||
if ((state->msg = malloc(strlen(state->path) + strlen(msg) + 3)) == NULL) {
|
||||
if ((state->msg = (char *)malloc(strlen(state->path) + strlen(msg) + 3)) ==
|
||||
NULL) {
|
||||
state->err = Z_MEM_ERROR;
|
||||
state->msg = (char *)"out of memory";
|
||||
return;
|
||||
}
|
||||
#if !defined(NO_snprintf) && !defined(NO_vsnprintf)
|
||||
snprintf(state->msg, strlen(state->path) + strlen(msg) + 3,
|
||||
"%s%s%s", state->path, ": ", msg);
|
||||
#else
|
||||
strcpy(state->msg, state->path);
|
||||
strcat(state->msg, ": ");
|
||||
strcat(state->msg, msg);
|
||||
#endif
|
||||
return;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* gzread.c -- zlib functions for reading gzip files
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012 Mark Adler
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012, 2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -58,7 +58,8 @@ local int gz_avail(state)
|
|||
return -1;
|
||||
if (state->eof == 0) {
|
||||
if (strm->avail_in) { /* copy what's there to the start */
|
||||
unsigned char *p = state->in, *q = strm->next_in;
|
||||
unsigned char *p = state->in;
|
||||
unsigned const char *q = strm->next_in;
|
||||
unsigned n = strm->avail_in;
|
||||
do {
|
||||
*p++ = *q++;
|
||||
|
|
@ -90,8 +91,8 @@ local int gz_look(state)
|
|||
/* allocate read buffers and inflate memory */
|
||||
if (state->size == 0) {
|
||||
/* allocate buffers */
|
||||
state->in = malloc(state->want);
|
||||
state->out = malloc(state->want << 1);
|
||||
state->in = (unsigned char *)malloc(state->want);
|
||||
state->out = (unsigned char *)malloc(state->want << 1);
|
||||
if (state->in == NULL || state->out == NULL) {
|
||||
if (state->out != NULL)
|
||||
free(state->out);
|
||||
|
|
@ -352,14 +353,14 @@ int ZEXPORT gzread(file, buf, len)
|
|||
|
||||
/* large len -- read directly into user buffer */
|
||||
else if (state->how == COPY) { /* read directly */
|
||||
if (gz_load(state, buf, len, &n) == -1)
|
||||
if (gz_load(state, (unsigned char *)buf, len, &n) == -1)
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* large len -- decompress directly into user buffer */
|
||||
else { /* state->how == GZIP */
|
||||
strm->avail_out = len;
|
||||
strm->next_out = buf;
|
||||
strm->next_out = (unsigned char *)buf;
|
||||
if (gz_decomp(state) == -1)
|
||||
return -1;
|
||||
n = state->x.have;
|
||||
|
|
@ -378,7 +379,11 @@ int ZEXPORT gzread(file, buf, len)
|
|||
}
|
||||
|
||||
/* -- see zlib.h -- */
|
||||
#undef gzgetc
|
||||
#ifdef Z_PREFIX_SET
|
||||
# undef z_gzgetc
|
||||
#else
|
||||
# undef gzgetc
|
||||
#endif
|
||||
int ZEXPORT gzgetc(file)
|
||||
gzFile file;
|
||||
{
|
||||
|
|
@ -518,7 +523,7 @@ char * ZEXPORT gzgets(file, buf, len)
|
|||
|
||||
/* look for end-of-line in current output buffer */
|
||||
n = state->x.have > left ? left : state->x.have;
|
||||
eol = memchr(state->x.next, '\n', n);
|
||||
eol = (unsigned char *)memchr(state->x.next, '\n', n);
|
||||
if (eol != NULL)
|
||||
n = (unsigned)(eol - state->x.next) + 1;
|
||||
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* gzwrite.c -- zlib functions for writing gzip files
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012 Mark Adler
|
||||
* Copyright (C) 2004, 2005, 2010, 2011, 2012, 2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -19,7 +19,7 @@ local int gz_init(state)
|
|||
z_streamp strm = &(state->strm);
|
||||
|
||||
/* allocate input buffer */
|
||||
state->in = malloc(state->want);
|
||||
state->in = (unsigned char *)malloc(state->want);
|
||||
if (state->in == NULL) {
|
||||
gz_error(state, Z_MEM_ERROR, "out of memory");
|
||||
return -1;
|
||||
|
|
@ -28,7 +28,7 @@ local int gz_init(state)
|
|||
/* only need output buffer and deflate state if compressing */
|
||||
if (!state->direct) {
|
||||
/* allocate output buffer */
|
||||
state->out = malloc(state->want);
|
||||
state->out = (unsigned char *)malloc(state->want);
|
||||
if (state->out == NULL) {
|
||||
free(state->in);
|
||||
gz_error(state, Z_MEM_ERROR, "out of memory");
|
||||
|
|
@ -168,7 +168,6 @@ int ZEXPORT gzwrite(file, buf, len)
|
|||
unsigned len;
|
||||
{
|
||||
unsigned put = len;
|
||||
unsigned n;
|
||||
gz_statep state;
|
||||
z_streamp strm;
|
||||
|
||||
|
|
@ -208,16 +207,19 @@ int ZEXPORT gzwrite(file, buf, len)
|
|||
if (len < state->size) {
|
||||
/* copy to input buffer, compress when full */
|
||||
do {
|
||||
unsigned have, copy;
|
||||
|
||||
if (strm->avail_in == 0)
|
||||
strm->next_in = state->in;
|
||||
n = state->size - strm->avail_in;
|
||||
if (n > len)
|
||||
n = len;
|
||||
memcpy(strm->next_in + strm->avail_in, buf, n);
|
||||
strm->avail_in += n;
|
||||
state->x.pos += n;
|
||||
buf = (char *)buf + n;
|
||||
len -= n;
|
||||
have = (unsigned)((strm->next_in + strm->avail_in) - state->in);
|
||||
copy = state->size - have;
|
||||
if (copy > len)
|
||||
copy = len;
|
||||
memcpy(state->in + have, buf, copy);
|
||||
strm->avail_in += copy;
|
||||
state->x.pos += copy;
|
||||
buf = (const char *)buf + copy;
|
||||
len -= copy;
|
||||
if (len && gz_comp(state, Z_NO_FLUSH) == -1)
|
||||
return 0;
|
||||
} while (len);
|
||||
|
|
@ -229,7 +231,7 @@ int ZEXPORT gzwrite(file, buf, len)
|
|||
|
||||
/* directly compress user buffer to file */
|
||||
strm->avail_in = len;
|
||||
strm->next_in = (voidp)buf;
|
||||
strm->next_in = (z_const Bytef *)buf;
|
||||
state->x.pos += len;
|
||||
if (gz_comp(state, Z_NO_FLUSH) == -1)
|
||||
return 0;
|
||||
|
|
@ -244,6 +246,7 @@ int ZEXPORT gzputc(file, c)
|
|||
gzFile file;
|
||||
int c;
|
||||
{
|
||||
unsigned have;
|
||||
unsigned char buf[1];
|
||||
gz_statep state;
|
||||
z_streamp strm;
|
||||
|
|
@ -267,12 +270,16 @@ int ZEXPORT gzputc(file, c)
|
|||
|
||||
/* try writing to input buffer for speed (state->size == 0 if buffer not
|
||||
initialized) */
|
||||
if (strm->avail_in < state->size) {
|
||||
if (state->size) {
|
||||
if (strm->avail_in == 0)
|
||||
strm->next_in = state->in;
|
||||
strm->next_in[strm->avail_in++] = c;
|
||||
state->x.pos++;
|
||||
return c & 0xff;
|
||||
have = (unsigned)((strm->next_in + strm->avail_in) - state->in);
|
||||
if (have < state->size) {
|
||||
state->in[have] = c;
|
||||
strm->avail_in++;
|
||||
state->x.pos++;
|
||||
return c & 0xff;
|
||||
}
|
||||
}
|
||||
|
||||
/* no room in buffer or not initialized, use gz_write() */
|
||||
|
|
@ -300,12 +307,11 @@ int ZEXPORT gzputs(file, str)
|
|||
#include <stdarg.h>
|
||||
|
||||
/* -- see zlib.h -- */
|
||||
int ZEXPORTVA gzprintf (gzFile file, const char *format, ...)
|
||||
int ZEXPORTVA gzvprintf(gzFile file, const char *format, va_list va)
|
||||
{
|
||||
int size, len;
|
||||
gz_statep state;
|
||||
z_streamp strm;
|
||||
va_list va;
|
||||
|
||||
/* get internal structure */
|
||||
if (file == NULL)
|
||||
|
|
@ -335,25 +341,20 @@ int ZEXPORTVA gzprintf (gzFile file, const char *format, ...)
|
|||
/* do the printf() into the input buffer, put length in len */
|
||||
size = (int)(state->size);
|
||||
state->in[size - 1] = 0;
|
||||
va_start(va, format);
|
||||
#ifdef NO_vsnprintf
|
||||
# ifdef HAS_vsprintf_void
|
||||
(void)vsprintf((char *)(state->in), format, va);
|
||||
va_end(va);
|
||||
for (len = 0; len < size; len++)
|
||||
if (state->in[len] == 0) break;
|
||||
# else
|
||||
len = vsprintf((char *)(state->in), format, va);
|
||||
va_end(va);
|
||||
# endif
|
||||
#else
|
||||
# ifdef HAS_vsnprintf_void
|
||||
(void)vsnprintf((char *)(state->in), size, format, va);
|
||||
va_end(va);
|
||||
len = strlen((char *)(state->in));
|
||||
# else
|
||||
len = vsnprintf((char *)(state->in), size, format, va);
|
||||
va_end(va);
|
||||
# endif
|
||||
#endif
|
||||
|
||||
|
|
@ -368,6 +369,17 @@ int ZEXPORTVA gzprintf (gzFile file, const char *format, ...)
|
|||
return len;
|
||||
}
|
||||
|
||||
int ZEXPORTVA gzprintf(gzFile file, const char *format, ...)
|
||||
{
|
||||
va_list va;
|
||||
int ret;
|
||||
|
||||
va_start(va, format);
|
||||
ret = gzvprintf(file, format, va);
|
||||
va_end(va);
|
||||
return ret;
|
||||
}
|
||||
|
||||
#else /* !STDC && !Z_HAVE_STDARG_H */
|
||||
|
||||
/* -- see zlib.h -- */
|
||||
|
|
@ -547,9 +559,9 @@ int ZEXPORT gzclose_w(file)
|
|||
}
|
||||
|
||||
/* flush, free memory, and close file */
|
||||
if (gz_comp(state, Z_FINISH) == -1)
|
||||
ret = state->err;
|
||||
if (state->size) {
|
||||
if (gz_comp(state, Z_FINISH) == -1)
|
||||
ret = state->err;
|
||||
if (!state->direct) {
|
||||
(void)deflateEnd(&(state->strm));
|
||||
free(state->out);
|
||||
|
|
|
|||
|
|
@ -255,7 +255,7 @@ out_func out;
|
|||
void FAR *out_desc;
|
||||
{
|
||||
struct inflate_state FAR *state;
|
||||
unsigned char FAR *next; /* next input */
|
||||
z_const unsigned char FAR *next; /* next input */
|
||||
unsigned char FAR *put; /* next output */
|
||||
unsigned have, left; /* available input and output */
|
||||
unsigned long hold; /* bit buffer */
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* inffast.c -- fast decoding
|
||||
* Copyright (C) 1995-2008, 2010 Mark Adler
|
||||
* Copyright (C) 1995-2008, 2010, 2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -69,8 +69,8 @@ z_streamp strm;
|
|||
unsigned start; /* inflate()'s starting value for strm->avail_out */
|
||||
{
|
||||
struct inflate_state FAR *state;
|
||||
unsigned char FAR *in; /* local strm->next_in */
|
||||
unsigned char FAR *last; /* while in < last, enough input available */
|
||||
z_const unsigned char FAR *in; /* local strm->next_in */
|
||||
z_const unsigned char FAR *last; /* have enough input while in < last */
|
||||
unsigned char FAR *out; /* local strm->next_out */
|
||||
unsigned char FAR *beg; /* inflate()'s initial strm->next_out */
|
||||
unsigned char FAR *end; /* while out < end, enough space available */
|
||||
|
|
|
|||
|
|
@ -93,11 +93,12 @@
|
|||
|
||||
/* function prototypes */
|
||||
local void fixedtables OF((struct inflate_state FAR *state));
|
||||
local int updatewindow OF((z_streamp strm, unsigned out));
|
||||
local int updatewindow OF((z_streamp strm, const unsigned char FAR *end,
|
||||
unsigned copy));
|
||||
#ifdef BUILDFIXED
|
||||
void makefixed OF((void));
|
||||
#endif
|
||||
local unsigned syncsearch OF((unsigned FAR *have, unsigned char FAR *buf,
|
||||
local unsigned syncsearch OF((unsigned FAR *have, const unsigned char FAR *buf,
|
||||
unsigned len));
|
||||
|
||||
int ZEXPORT inflateResetKeep(strm)
|
||||
|
|
@ -375,12 +376,13 @@ void makefixed()
|
|||
output will fall in the output data, making match copies simpler and faster.
|
||||
The advantage may be dependent on the size of the processor's data caches.
|
||||
*/
|
||||
local int updatewindow(strm, out)
|
||||
local int updatewindow(strm, end, copy)
|
||||
z_streamp strm;
|
||||
unsigned out;
|
||||
const Bytef *end;
|
||||
unsigned copy;
|
||||
{
|
||||
struct inflate_state FAR *state;
|
||||
unsigned copy, dist;
|
||||
unsigned dist;
|
||||
|
||||
state = (struct inflate_state FAR *)strm->state;
|
||||
|
||||
|
|
@ -400,19 +402,18 @@ unsigned out;
|
|||
}
|
||||
|
||||
/* copy state->wsize or less output bytes into the circular window */
|
||||
copy = out - strm->avail_out;
|
||||
if (copy >= state->wsize) {
|
||||
zmemcpy(state->window, strm->next_out - state->wsize, state->wsize);
|
||||
zmemcpy(state->window, end - state->wsize, state->wsize);
|
||||
state->wnext = 0;
|
||||
state->whave = state->wsize;
|
||||
}
|
||||
else {
|
||||
dist = state->wsize - state->wnext;
|
||||
if (dist > copy) dist = copy;
|
||||
zmemcpy(state->window + state->wnext, strm->next_out - copy, dist);
|
||||
zmemcpy(state->window + state->wnext, end - copy, dist);
|
||||
copy -= dist;
|
||||
if (copy) {
|
||||
zmemcpy(state->window, strm->next_out - copy, copy);
|
||||
zmemcpy(state->window, end - copy, copy);
|
||||
state->wnext = copy;
|
||||
state->whave = state->wsize;
|
||||
}
|
||||
|
|
@ -606,7 +607,7 @@ z_streamp strm;
|
|||
int flush;
|
||||
{
|
||||
struct inflate_state FAR *state;
|
||||
unsigned char FAR *next; /* next input */
|
||||
z_const unsigned char FAR *next; /* next input */
|
||||
unsigned char FAR *put; /* next output */
|
||||
unsigned have, left; /* available input and output */
|
||||
unsigned long hold; /* bit buffer */
|
||||
|
|
@ -920,7 +921,7 @@ int flush;
|
|||
while (state->have < 19)
|
||||
state->lens[order[state->have++]] = 0;
|
||||
state->next = state->codes;
|
||||
state->lencode = (code const FAR *)(state->next);
|
||||
state->lencode = (const code FAR *)(state->next);
|
||||
state->lenbits = 7;
|
||||
ret = inflate_table(CODES, state->lens, 19, &(state->next),
|
||||
&(state->lenbits), state->work);
|
||||
|
|
@ -994,7 +995,7 @@ int flush;
|
|||
values here (9 and 6) without reading the comments in inftrees.h
|
||||
concerning the ENOUGH constants, which depend on those values */
|
||||
state->next = state->codes;
|
||||
state->lencode = (code const FAR *)(state->next);
|
||||
state->lencode = (const code FAR *)(state->next);
|
||||
state->lenbits = 9;
|
||||
ret = inflate_table(LENS, state->lens, state->nlen, &(state->next),
|
||||
&(state->lenbits), state->work);
|
||||
|
|
@ -1003,7 +1004,7 @@ int flush;
|
|||
state->mode = BAD;
|
||||
break;
|
||||
}
|
||||
state->distcode = (code const FAR *)(state->next);
|
||||
state->distcode = (const code FAR *)(state->next);
|
||||
state->distbits = 6;
|
||||
ret = inflate_table(DISTS, state->lens + state->nlen, state->ndist,
|
||||
&(state->next), &(state->distbits), state->work);
|
||||
|
|
@ -1230,7 +1231,7 @@ int flush;
|
|||
RESTORE();
|
||||
if (state->wsize || (out != strm->avail_out && state->mode < BAD &&
|
||||
(state->mode < CHECK || flush != Z_FINISH)))
|
||||
if (updatewindow(strm, out)) {
|
||||
if (updatewindow(strm, strm->next_out, out - strm->avail_out)) {
|
||||
state->mode = MEM;
|
||||
return Z_MEM_ERROR;
|
||||
}
|
||||
|
|
@ -1264,6 +1265,29 @@ z_streamp strm;
|
|||
return Z_OK;
|
||||
}
|
||||
|
||||
int ZEXPORT inflateGetDictionary(strm, dictionary, dictLength)
|
||||
z_streamp strm;
|
||||
Bytef *dictionary;
|
||||
uInt *dictLength;
|
||||
{
|
||||
struct inflate_state FAR *state;
|
||||
|
||||
/* check state */
|
||||
if (strm == Z_NULL || strm->state == Z_NULL) return Z_STREAM_ERROR;
|
||||
state = (struct inflate_state FAR *)strm->state;
|
||||
|
||||
/* copy dictionary */
|
||||
if (state->whave && dictionary != Z_NULL) {
|
||||
zmemcpy(dictionary, state->window + state->wnext,
|
||||
state->whave - state->wnext);
|
||||
zmemcpy(dictionary + state->whave - state->wnext,
|
||||
state->window, state->wnext);
|
||||
}
|
||||
if (dictLength != Z_NULL)
|
||||
*dictLength = state->whave;
|
||||
return Z_OK;
|
||||
}
|
||||
|
||||
int ZEXPORT inflateSetDictionary(strm, dictionary, dictLength)
|
||||
z_streamp strm;
|
||||
const Bytef *dictionary;
|
||||
|
|
@ -1271,8 +1295,6 @@ uInt dictLength;
|
|||
{
|
||||
struct inflate_state FAR *state;
|
||||
unsigned long dictid;
|
||||
unsigned char *next;
|
||||
unsigned avail;
|
||||
int ret;
|
||||
|
||||
/* check state */
|
||||
|
|
@ -1291,13 +1313,7 @@ uInt dictLength;
|
|||
|
||||
/* copy dictionary to window using updatewindow(), which will amend the
|
||||
existing dictionary if appropriate */
|
||||
next = strm->next_out;
|
||||
avail = strm->avail_out;
|
||||
strm->next_out = (Bytef *)dictionary + dictLength;
|
||||
strm->avail_out = 0;
|
||||
ret = updatewindow(strm, dictLength);
|
||||
strm->avail_out = avail;
|
||||
strm->next_out = next;
|
||||
ret = updatewindow(strm, dictionary + dictLength, dictLength);
|
||||
if (ret) {
|
||||
state->mode = MEM;
|
||||
return Z_MEM_ERROR;
|
||||
|
|
@ -1337,7 +1353,7 @@ gz_headerp head;
|
|||
*/
|
||||
local unsigned syncsearch(have, buf, len)
|
||||
unsigned FAR *have;
|
||||
unsigned char FAR *buf;
|
||||
const unsigned char FAR *buf;
|
||||
unsigned len;
|
||||
{
|
||||
unsigned got;
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* inftrees.c -- generate Huffman trees for efficient decoding
|
||||
* Copyright (C) 1995-2012 Mark Adler
|
||||
* Copyright (C) 1995-2013 Mark Adler
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -9,7 +9,7 @@
|
|||
#define MAXBITS 15
|
||||
|
||||
const char inflate_copyright[] =
|
||||
" inflate 1.2.7 Copyright 1995-2012 Mark Adler ";
|
||||
" inflate 1.2.8 Copyright 1995-2013 Mark Adler ";
|
||||
/*
|
||||
If you use the zlib library in a product, an acknowledgment is welcome
|
||||
in the documentation of your product. If for some reason you cannot
|
||||
|
|
@ -62,7 +62,7 @@ unsigned short FAR *work;
|
|||
35, 43, 51, 59, 67, 83, 99, 115, 131, 163, 195, 227, 258, 0, 0};
|
||||
static const unsigned short lext[31] = { /* Length codes 257..285 extra */
|
||||
16, 16, 16, 16, 16, 16, 16, 16, 17, 17, 17, 17, 18, 18, 18, 18,
|
||||
19, 19, 19, 19, 20, 20, 20, 20, 21, 21, 21, 21, 16, 78, 68};
|
||||
19, 19, 19, 19, 20, 20, 20, 20, 21, 21, 21, 21, 16, 72, 78};
|
||||
static const unsigned short dbase[32] = { /* Distance codes 0..29 base */
|
||||
1, 2, 3, 4, 5, 7, 9, 13, 17, 25, 33, 49, 65, 97, 129, 193,
|
||||
257, 385, 513, 769, 1025, 1537, 2049, 3073, 4097, 6145,
|
||||
|
|
@ -208,8 +208,8 @@ unsigned short FAR *work;
|
|||
mask = used - 1; /* mask for comparing low */
|
||||
|
||||
/* check available table space */
|
||||
if ((type == LENS && used >= ENOUGH_LENS) ||
|
||||
(type == DISTS && used >= ENOUGH_DISTS))
|
||||
if ((type == LENS && used > ENOUGH_LENS) ||
|
||||
(type == DISTS && used > ENOUGH_DISTS))
|
||||
return 1;
|
||||
|
||||
/* process all codes and make table entries */
|
||||
|
|
@ -277,8 +277,8 @@ unsigned short FAR *work;
|
|||
|
||||
/* check for enough space */
|
||||
used += 1U << curr;
|
||||
if ((type == LENS && used >= ENOUGH_LENS) ||
|
||||
(type == DISTS && used >= ENOUGH_DISTS))
|
||||
if ((type == LENS && used > ENOUGH_LENS) ||
|
||||
(type == DISTS && used > ENOUGH_DISTS))
|
||||
return 1;
|
||||
|
||||
/* point entry in root table to sub-table */
|
||||
|
|
|
|||
|
|
@ -146,8 +146,8 @@ local void send_tree OF((deflate_state *s, ct_data *tree, int max_code));
|
|||
local int build_bl_tree OF((deflate_state *s));
|
||||
local void send_all_trees OF((deflate_state *s, int lcodes, int dcodes,
|
||||
int blcodes));
|
||||
local void compress_block OF((deflate_state *s, ct_data *ltree,
|
||||
ct_data *dtree));
|
||||
local void compress_block OF((deflate_state *s, const ct_data *ltree,
|
||||
const ct_data *dtree));
|
||||
local int detect_data_type OF((deflate_state *s));
|
||||
local unsigned bi_reverse OF((unsigned value, int length));
|
||||
local void bi_windup OF((deflate_state *s));
|
||||
|
|
@ -972,7 +972,8 @@ void ZLIB_INTERNAL _tr_flush_block(s, buf, stored_len, last)
|
|||
} else if (s->strategy == Z_FIXED || static_lenb == opt_lenb) {
|
||||
#endif
|
||||
send_bits(s, (STATIC_TREES<<1)+last, 3);
|
||||
compress_block(s, (ct_data *)static_ltree, (ct_data *)static_dtree);
|
||||
compress_block(s, (const ct_data *)static_ltree,
|
||||
(const ct_data *)static_dtree);
|
||||
#ifdef DEBUG
|
||||
s->compressed_len += 3 + s->static_len;
|
||||
#endif
|
||||
|
|
@ -980,7 +981,8 @@ void ZLIB_INTERNAL _tr_flush_block(s, buf, stored_len, last)
|
|||
send_bits(s, (DYN_TREES<<1)+last, 3);
|
||||
send_all_trees(s, s->l_desc.max_code+1, s->d_desc.max_code+1,
|
||||
max_blindex+1);
|
||||
compress_block(s, (ct_data *)s->dyn_ltree, (ct_data *)s->dyn_dtree);
|
||||
compress_block(s, (const ct_data *)s->dyn_ltree,
|
||||
(const ct_data *)s->dyn_dtree);
|
||||
#ifdef DEBUG
|
||||
s->compressed_len += 3 + s->opt_len;
|
||||
#endif
|
||||
|
|
@ -1057,8 +1059,8 @@ int ZLIB_INTERNAL _tr_tally (s, dist, lc)
|
|||
*/
|
||||
local void compress_block(s, ltree, dtree)
|
||||
deflate_state *s;
|
||||
ct_data *ltree; /* literal tree */
|
||||
ct_data *dtree; /* distance tree */
|
||||
const ct_data *ltree; /* literal tree */
|
||||
const ct_data *dtree; /* distance tree */
|
||||
{
|
||||
unsigned dist; /* distance of matched string */
|
||||
int lc; /* match length or unmatched char (if dist == 0) */
|
||||
|
|
|
|||
|
|
@ -30,7 +30,7 @@ int ZEXPORT uncompress (dest, destLen, source, sourceLen)
|
|||
z_stream stream;
|
||||
int err;
|
||||
|
||||
stream.next_in = (Bytef*)source;
|
||||
stream.next_in = (z_const Bytef *)source;
|
||||
stream.avail_in = (uInt)sourceLen;
|
||||
/* Check for source > 64K on 16-bit machine: */
|
||||
if ((uLong)stream.avail_in != sourceLen) return Z_BUF_ERROR;
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* zconf.h -- configuration of the zlib compression library
|
||||
* Copyright (C) 1995-2012 Jean-loup Gailly.
|
||||
* Copyright (C) 1995-2013 Jean-loup Gailly.
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -21,6 +21,7 @@
|
|||
# define _dist_code z__dist_code
|
||||
# define _length_code z__length_code
|
||||
# define _tr_align z__tr_align
|
||||
# define _tr_flush_bits z__tr_flush_bits
|
||||
# define _tr_flush_block z__tr_flush_block
|
||||
# define _tr_init z__tr_init
|
||||
# define _tr_stored_block z__tr_stored_block
|
||||
|
|
@ -77,6 +78,7 @@
|
|||
# define gzopen_w z_gzopen_w
|
||||
# endif
|
||||
# define gzprintf z_gzprintf
|
||||
# define gzvprintf z_gzvprintf
|
||||
# define gzputc z_gzputc
|
||||
# define gzputs z_gzputs
|
||||
# define gzread z_gzread
|
||||
|
|
@ -103,6 +105,7 @@
|
|||
# define inflateReset z_inflateReset
|
||||
# define inflateReset2 z_inflateReset2
|
||||
# define inflateSetDictionary z_inflateSetDictionary
|
||||
# define inflateGetDictionary z_inflateGetDictionary
|
||||
# define inflateSync z_inflateSync
|
||||
# define inflateSyncPoint z_inflateSyncPoint
|
||||
# define inflateUndermine z_inflateUndermine
|
||||
|
|
@ -388,20 +391,14 @@ typedef uLong FAR uLongf;
|
|||
typedef Byte *voidp;
|
||||
#endif
|
||||
|
||||
/* ./configure may #define Z_U4 here */
|
||||
|
||||
#if !defined(Z_U4) && !defined(Z_SOLO) && defined(STDC)
|
||||
# include <limits.h>
|
||||
# if (UINT_MAX == 0xffffffffUL)
|
||||
# define Z_U4 unsigned
|
||||
# else
|
||||
# if (ULONG_MAX == 0xffffffffUL)
|
||||
# define Z_U4 unsigned long
|
||||
# else
|
||||
# if (USHRT_MAX == 0xffffffffUL)
|
||||
# define Z_U4 unsigned short
|
||||
# endif
|
||||
# endif
|
||||
# elif (ULONG_MAX == 0xffffffffUL)
|
||||
# define Z_U4 unsigned long
|
||||
# elif (USHRT_MAX == 0xffffffffUL)
|
||||
# define Z_U4 unsigned short
|
||||
# endif
|
||||
#endif
|
||||
|
||||
|
|
@ -425,8 +422,16 @@ typedef uLong FAR uLongf;
|
|||
# endif
|
||||
#endif
|
||||
|
||||
#if defined(STDC) || defined(Z_HAVE_STDARG_H)
|
||||
# ifndef Z_SOLO
|
||||
# include <stdarg.h> /* for va_list */
|
||||
# endif
|
||||
#endif
|
||||
|
||||
#ifdef _WIN32
|
||||
# include <stddef.h> /* for wchar_t */
|
||||
# ifndef Z_SOLO
|
||||
# include <stddef.h> /* for wchar_t */
|
||||
# endif
|
||||
#endif
|
||||
|
||||
/* a little trick to accommodate both "#define _LARGEFILE64_SOURCE" and
|
||||
|
|
@ -435,7 +440,7 @@ typedef uLong FAR uLongf;
|
|||
* both "#undef _LARGEFILE64_SOURCE" and "#define _LARGEFILE64_SOURCE 0" as
|
||||
* equivalently requesting no 64-bit operations
|
||||
*/
|
||||
#if defined(LARGEFILE64_SOURCE) && -_LARGEFILE64_SOURCE - -1 == 1
|
||||
#if defined(_LARGEFILE64_SOURCE) && -_LARGEFILE64_SOURCE - -1 == 1
|
||||
# undef _LARGEFILE64_SOURCE
|
||||
#endif
|
||||
|
||||
|
|
@ -443,7 +448,7 @@ typedef uLong FAR uLongf;
|
|||
# define Z_HAVE_UNISTD_H
|
||||
#endif
|
||||
#ifndef Z_SOLO
|
||||
# if defined(Z_HAVE_UNISTD_H) || defined(LARGEFILE64_SOURCE)
|
||||
# if defined(Z_HAVE_UNISTD_H) || defined(_LARGEFILE64_SOURCE)
|
||||
# include <unistd.h> /* for SEEK_*, off_t, and _LFS64_LARGEFILE */
|
||||
# ifdef VMS
|
||||
# include <unixio.h> /* for off_t */
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/* zlib.h -- interface of the 'zlib' general purpose compression library
|
||||
version 1.2.7, May 2nd, 2012
|
||||
version 1.2.8, April 28th, 2013
|
||||
|
||||
Copyright (C) 1995-2012 Jean-loup Gailly and Mark Adler
|
||||
Copyright (C) 1995-2013 Jean-loup Gailly and Mark Adler
|
||||
|
||||
This software is provided 'as-is', without any express or implied
|
||||
warranty. In no event will the authors be held liable for any damages
|
||||
|
|
@ -37,11 +37,11 @@
|
|||
extern "C" {
|
||||
#endif
|
||||
|
||||
#define ZLIB_VERSION "1.2.7"
|
||||
#define ZLIB_VERNUM 0x1270
|
||||
#define ZLIB_VERSION "1.2.8"
|
||||
#define ZLIB_VERNUM 0x1280
|
||||
#define ZLIB_VER_MAJOR 1
|
||||
#define ZLIB_VER_MINOR 2
|
||||
#define ZLIB_VER_REVISION 7
|
||||
#define ZLIB_VER_REVISION 8
|
||||
#define ZLIB_VER_SUBREVISION 0
|
||||
|
||||
/*
|
||||
|
|
@ -839,6 +839,21 @@ ZEXTERN int ZEXPORT inflateSetDictionary OF((z_streamp strm,
|
|||
inflate().
|
||||
*/
|
||||
|
||||
ZEXTERN int ZEXPORT inflateGetDictionary OF((z_streamp strm,
|
||||
Bytef *dictionary,
|
||||
uInt *dictLength));
|
||||
/*
|
||||
Returns the sliding dictionary being maintained by inflate. dictLength is
|
||||
set to the number of bytes in the dictionary, and that many bytes are copied
|
||||
to dictionary. dictionary must have enough space, where 32768 bytes is
|
||||
always enough. If inflateGetDictionary() is called with dictionary equal to
|
||||
Z_NULL, then only the dictionary length is returned, and nothing is copied.
|
||||
Similary, if dictLength is Z_NULL, then it is not set.
|
||||
|
||||
inflateGetDictionary returns Z_OK on success, or Z_STREAM_ERROR if the
|
||||
stream state is inconsistent.
|
||||
*/
|
||||
|
||||
ZEXTERN int ZEXPORT inflateSync OF((z_streamp strm));
|
||||
/*
|
||||
Skips invalid compressed data until a possible full flush point (see above
|
||||
|
|
@ -846,7 +861,7 @@ ZEXTERN int ZEXPORT inflateSync OF((z_streamp strm));
|
|||
available input is skipped. No output is provided.
|
||||
|
||||
inflateSync searches for a 00 00 FF FF pattern in the compressed data.
|
||||
All full flush points have this pattern, but not all occurences of this
|
||||
All full flush points have this pattern, but not all occurrences of this
|
||||
pattern are full flush points.
|
||||
|
||||
inflateSync returns Z_OK if a possible full flush point has been found,
|
||||
|
|
@ -1007,7 +1022,8 @@ ZEXTERN int ZEXPORT inflateBackInit OF((z_streamp strm, int windowBits,
|
|||
the version of the header file.
|
||||
*/
|
||||
|
||||
typedef unsigned (*in_func) OF((void FAR *, unsigned char FAR * FAR *));
|
||||
typedef unsigned (*in_func) OF((void FAR *,
|
||||
z_const unsigned char FAR * FAR *));
|
||||
typedef int (*out_func) OF((void FAR *, unsigned char FAR *, unsigned));
|
||||
|
||||
ZEXTERN int ZEXPORT inflateBack OF((z_streamp strm,
|
||||
|
|
@ -1015,11 +1031,12 @@ ZEXTERN int ZEXPORT inflateBack OF((z_streamp strm,
|
|||
out_func out, void FAR *out_desc));
|
||||
/*
|
||||
inflateBack() does a raw inflate with a single call using a call-back
|
||||
interface for input and output. This is more efficient than inflate() for
|
||||
file i/o applications in that it avoids copying between the output and the
|
||||
sliding window by simply making the window itself the output buffer. This
|
||||
function trusts the application to not change the output buffer passed by
|
||||
the output function, at least until inflateBack() returns.
|
||||
interface for input and output. This is potentially more efficient than
|
||||
inflate() for file i/o applications, in that it avoids copying between the
|
||||
output and the sliding window by simply making the window itself the output
|
||||
buffer. inflate() can be faster on modern CPUs when used with large
|
||||
buffers. inflateBack() trusts the application to not change the output
|
||||
buffer passed by the output function, at least until inflateBack() returns.
|
||||
|
||||
inflateBackInit() must be called first to allocate the internal state
|
||||
and to initialize the state with the user-provided window buffer.
|
||||
|
|
@ -1736,6 +1753,13 @@ ZEXTERN int ZEXPORT deflateResetKeep OF((z_streamp));
|
|||
ZEXTERN gzFile ZEXPORT gzopen_w OF((const wchar_t *path,
|
||||
const char *mode));
|
||||
#endif
|
||||
#if defined(STDC) || defined(Z_HAVE_STDARG_H)
|
||||
# ifndef Z_SOLO
|
||||
ZEXTERN int ZEXPORTVA gzvprintf Z_ARG((gzFile file,
|
||||
const char *format,
|
||||
va_list va));
|
||||
# endif
|
||||
#endif
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
|
|
|
|||
|
|
@ -14,7 +14,7 @@
|
|||
struct internal_state {int dummy;}; /* for buggy compilers */
|
||||
#endif
|
||||
|
||||
const char * const z_errmsg[10] = {
|
||||
z_const char * const z_errmsg[10] = {
|
||||
"need dictionary", /* Z_NEED_DICT 2 */
|
||||
"stream end", /* Z_STREAM_END 1 */
|
||||
"", /* Z_OK 0 */
|
||||
|
|
|
|||
|
|
@ -1,5 +1,5 @@
|
|||
/* zutil.h -- internal interface and configuration of the compression library
|
||||
* Copyright (C) 1995-2012 Jean-loup Gailly.
|
||||
* Copyright (C) 1995-2013 Jean-loup Gailly.
|
||||
* For conditions of distribution and use, see copyright notice in zlib.h
|
||||
*/
|
||||
|
||||
|
|
@ -44,13 +44,13 @@ typedef unsigned short ush;
|
|||
typedef ush FAR ushf;
|
||||
typedef unsigned long ulg;
|
||||
|
||||
extern const char * const z_errmsg[10]; /* indexed by 2-zlib_error */
|
||||
extern z_const char * const z_errmsg[10]; /* indexed by 2-zlib_error */
|
||||
/* (size given to avoid silly warnings with Visual C++) */
|
||||
|
||||
#define ERR_MSG(err) z_errmsg[Z_NEED_DICT-(err)]
|
||||
|
||||
#define ERR_RETURN(strm,err) \
|
||||
return (strm->msg = (char*)ERR_MSG(err), (err))
|
||||
return (strm->msg = ERR_MSG(err), (err))
|
||||
/* To be used only when the state is known to be valid */
|
||||
|
||||
/* common constants */
|
||||
|
|
@ -168,7 +168,8 @@ extern const char * const z_errmsg[10]; /* indexed by 2-zlib_error */
|
|||
#endif
|
||||
|
||||
/* provide prototypes for these when building zlib without LFS */
|
||||
#if !defined(_WIN32) && (!defined(_LARGEFILE64_SOURCE) || _LFS64_LARGEFILE-0 == 0)
|
||||
#if !defined(_WIN32) && \
|
||||
(!defined(_LARGEFILE64_SOURCE) || _LFS64_LARGEFILE-0 == 0)
|
||||
ZEXTERN uLong ZEXPORT adler32_combine64 OF((uLong, uLong, z_off_t));
|
||||
ZEXTERN uLong ZEXPORT crc32_combine64 OF((uLong, uLong, z_off_t));
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -185,25 +185,32 @@ Global
|
|||
{18E42F6F-3A62-41EE-B42F-79366C4F1E95}.Release|x64.Build.0 = Release SSE2|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.ActiveCfg = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.Build.0 = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.Build.0 = Debug|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.ActiveCfg = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.Build.0 = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.Build.0 = Devel|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.Build.0 = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.Build.0 = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.Build.0 = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.Build.0 = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.Build.0 = Release|x64
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.ActiveCfg = Debug|Win32
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.Build.0 = Debug|Win32
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|x64.ActiveCfg = Debug|Win32
|
||||
|
|
|
|||
|
|
@ -6,7 +6,7 @@
|
|||
</PropertyGroup>
|
||||
<ItemDefinitionGroup>
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>$(SvnRootDir)\3rdparty\;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>$(SvnRootDir)\3rdparty\;$(SvnRootDir)\3rdparty\soundtouch\;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<AdditionalLibraryDirectories>$(SvnRootDir)\deps\$(Platform)\$(Configuration);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
|
|
|
|||
|
|
@ -97,13 +97,16 @@ Global
|
|||
{5F78E90B-BD22-47B1-9CA5-7A80F4DF5EF3}.Release|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.ActiveCfg = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.Build.0 = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.Build.0 = Debug|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.ActiveCfg = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.Build.0 = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.Build.0 = Devel|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.Build.0 = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|Win32
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|x64
|
||||
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.Build.0 = Release|x64
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.ActiveCfg = Debug|Win32
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.Build.0 = Debug|Win32
|
||||
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|x64.ActiveCfg = Debug|Win32
|
||||
|
|
|
|||
|
|
@ -19,7 +19,7 @@
|
|||
#ifdef __linux__
|
||||
#include "WavFile.h"
|
||||
#else
|
||||
#include "soundtouch/WavFile.h"
|
||||
#include "soundtouch/source/SoundStretch/WavFile.h"
|
||||
#endif
|
||||
|
||||
static WavOutFile* _new_WavOutFile( const char* destfile )
|
||||
|
|
|
|||
|
|
@ -31,7 +31,7 @@
|
|||
#ifdef __linux__
|
||||
#include "WavFile.h"
|
||||
#else
|
||||
#include "soundtouch/WavFile.h"
|
||||
#include "soundtouch/source/SoundStretch/WavFile.h"
|
||||
#endif
|
||||
|
||||
char libraryName[256];
|
||||
|
|
|
|||
|
|
@ -22,7 +22,7 @@
|
|||
#ifdef __linux__
|
||||
#include "WavFile.h"
|
||||
#else
|
||||
#include "soundtouch/WavFile.h"
|
||||
#include "soundtouch/source/SoundStretch/WavFile.h"
|
||||
#endif
|
||||
|
||||
s32 g_logsound = 0;
|
||||
|
|
|
|||
Loading…
Reference in New Issue