k
/
FFmpegAndroid
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

add library of sox

Browse Source
dev
xufulong 3 years ago
parent 1e632fd324
commit c918001551
205 changed files with 54065 additions and 0 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 342
      app/src/main/cpp/sox/8svx.c
  2. 184
      app/src/main/cpp/sox/Makefile.am
  3. 367
      app/src/main/cpp/sox/adpcm.c
  4. 79
      app/src/main/cpp/sox/adpcm.h
  5. 322
      app/src/main/cpp/sox/adpcms.c
  6. 55
      app/src/main/cpp/sox/adpcms.h
  7. 36
      app/src/main/cpp/sox/aifc-fmt.c
  8. 33
      app/src/main/cpp/sox/aiff-fmt.c
  9. 1151
      app/src/main/cpp/sox/aiff.c
  10. 22
      app/src/main/cpp/sox/aiff.h
  11. 21
      app/src/main/cpp/sox/al-fmt.c
  12. 379
      app/src/main/cpp/sox/alsa.c
  13. 90
      app/src/main/cpp/sox/amr-nb.c
  14. 115
      app/src/main/cpp/sox/amr-wb.c
  15. 335
      app/src/main/cpp/sox/amr.h
  16. 134
      app/src/main/cpp/sox/ao.c
  17. 237
      app/src/main/cpp/sox/au.c
  18. 287
      app/src/main/cpp/sox/avr.c
  19. 47
      app/src/main/cpp/sox/band.h
  20. 325
      app/src/main/cpp/sox/bend.c
  21. 178
      app/src/main/cpp/sox/biquad.c
  22. 78
      app/src/main/cpp/sox/biquad.h
  23. 416
      app/src/main/cpp/sox/biquads.c
  24. 273
      app/src/main/cpp/sox/bit-rot/CoreAudio/CoreAudio.h
  25. 0
      app/src/main/cpp/sox/bit-rot/mmreg.h
  26. 89
      app/src/main/cpp/sox/bit-rot/mmsystem.h
  27. 160
      app/src/main/cpp/sox/bit-rot/sndio.h
  28. 289
      app/src/main/cpp/sox/bit-rot/sys/audioio.h
  29. 34
      app/src/main/cpp/sox/bit-rot/windows.h
  30. 35
      app/src/main/cpp/sox/caf.c
  31. 49
      app/src/main/cpp/sox/cdr.c
  32. 351
      app/src/main/cpp/sox/chorus.c
  33. 293
      app/src/main/cpp/sox/compand.c
  34. 229
      app/src/main/cpp/sox/compandt.c
  35. 52
      app/src/main/cpp/sox/compandt.h
  36. 49
      app/src/main/cpp/sox/contrast.c
  37. 418
      app/src/main/cpp/sox/coreaudio.c
  38. 114
      app/src/main/cpp/sox/cvsd-fmt.c
  39. 680
      app/src/main/cpp/sox/cvsd.c
  40. 68
      app/src/main/cpp/sox/cvsd.h
  41. 112
      app/src/main/cpp/sox/cvsdfilt.h
  42. 162
      app/src/main/cpp/sox/dat.c
  43. 165
      app/src/main/cpp/sox/dcshift.c
  44. 118
      app/src/main/cpp/sox/deemph.plt
  45. 163
      app/src/main/cpp/sox/delay.c
  46. 137
      app/src/main/cpp/sox/dft_filter.c
  47. 16
      app/src/main/cpp/sox/dft_filter.h
  48. 436
      app/src/main/cpp/sox/dither.c
  49. 63
      app/src/main/cpp/sox/dither.h
  50. 73
      app/src/main/cpp/sox/divide.c
  51. 85
      app/src/main/cpp/sox/downsample.c
  52. 32
      app/src/main/cpp/sox/dvms-fmt.c
  53. 97
      app/src/main/cpp/sox/earwax.c
  54. 266
      app/src/main/cpp/sox/echo.c
  55. 278
      app/src/main/cpp/sox/echos.c
  56. 669
      app/src/main/cpp/sox/effects.c
  57. 90
      app/src/main/cpp/sox/effects.h
  58. 483
      app/src/main/cpp/sox/effects_i.c
  59. 645
      app/src/main/cpp/sox/effects_i_dsp.c
  60. 97
      app/src/main/cpp/sox/example0.c
  61. 167
      app/src/main/cpp/sox/example1.c
  62. 118
      app/src/main/cpp/sox/example2.c
  63. 113
      app/src/main/cpp/sox/example3.c
  64. 98
      app/src/main/cpp/sox/example4.c
  65. 83
      app/src/main/cpp/sox/example5.c
  66. 128
      app/src/main/cpp/sox/example6.c
  67. 21
      app/src/main/cpp/sox/f4-fmt.c
  68. 21
      app/src/main/cpp/sox/f8-fmt.c
  69. 399
      app/src/main/cpp/sox/fade.c
  70. 31
      app/src/main/cpp/sox/fap.c
  71. 1362
      app/src/main/cpp/sox/fft4g.c
  72. 37
      app/src/main/cpp/sox/fft4g.h
  73. 119
      app/src/main/cpp/sox/fifo.h
  74. 105
      app/src/main/cpp/sox/fir.c
  75. 145
      app/src/main/cpp/sox/firfit.c
  76. 606
      app/src/main/cpp/sox/flac.c
  77. 275
      app/src/main/cpp/sox/flanger.c
  78. 1351
      app/src/main/cpp/sox/formats.c
  79. 130
      app/src/main/cpp/sox/formats.h
  80. 545
      app/src/main/cpp/sox/formats_i.c
  81. 2421
      app/src/main/cpp/sox/g711.c
  82. 21
      app/src/main/cpp/sox/g711.h
  83. 167
      app/src/main/cpp/sox/g721.c
  84. 151
      app/src/main/cpp/sox/g723_24.c
  85. 171
      app/src/main/cpp/sox/g723_40.c
  86. 575
      app/src/main/cpp/sox/g72x.c
  87. 157
      app/src/main/cpp/sox/g72x.h
  88. 276
      app/src/main/cpp/sox/gain.c
  89. 349
      app/src/main/cpp/sox/getopt.c
  90. 247
      app/src/main/cpp/sox/gsm.c
  91. 207
      app/src/main/cpp/sox/gsrt.c
  92. 502
      app/src/main/cpp/sox/hcom.c
  93. 102
      app/src/main/cpp/sox/hilbert.c
  94. 80
      app/src/main/cpp/sox/htk.c
  95. 227
      app/src/main/cpp/sox/id3.c
  96. 25
      app/src/main/cpp/sox/id3.h
  97. 33
      app/src/main/cpp/sox/ima-fmt.c
  98. 365
      app/src/main/cpp/sox/ima_rw.c
  99. 84
      app/src/main/cpp/sox/ima_rw.h
  100. 58
      app/src/main/cpp/sox/input.c
  101. Some files were not shown because too many files have changed in this diff Show More

342
app/src/main/cpp/sox/8svx.c
Unescape View File

@ -0,0 +1,342 @@
/* Amiga 8SVX format handler: W V Neisius, February 1992 */
#include "sox_i.h"
#include <errno.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#define BUFLEN 512
/* Private data used by writer */
typedef struct{
uint32_t nsamples;
uint32_t left;
off_t ch0_pos;
sox_uint8_t buf[4][BUFLEN];
FILE* tmp[4];
} priv_t;
static void svxwriteheader(sox_format_t *, size_t);
/*======================================================================*/
/* 8SVXSTARTREAD */
/*======================================================================*/
static int startread(sox_format_t * ft)
{
priv_t * p = (priv_t * ) ft->priv;
char buf[12];
char *chunk_buf;
uint32_t totalsize;
uint32_t chunksize;
uint32_t channels;
unsigned short rate;
if (! ft->seekable)
{
lsx_fail_errno(ft,SOX_EINVAL,"8svx input file must be a file, not a pipe");
return (SOX_EOF);
}
rate = 0;
channels = 1;
/* read FORM chunk */
if (lsx_reads(ft, buf, (size_t)4) == SOX_EOF || strncmp(buf, "FORM", (size_t)4) != 0)
{
lsx_fail_errno(ft, SOX_EHDR, "Header did not begin with magic word `FORM'");
return(SOX_EOF);
}
lsx_readdw(ft, &totalsize);
if (lsx_reads(ft, buf, (size_t)4) == SOX_EOF || strncmp(buf, "8SVX", (size_t)4) != 0)
{
lsx_fail_errno(ft, SOX_EHDR, "'FORM' chunk does not specify `8SVX' as type");
return(SOX_EOF);
}
/* read chunks until 'BODY' (or end) */
while (lsx_reads(ft, buf, (size_t)4) == SOX_SUCCESS && strncmp(buf,"BODY",(size_t)4) != 0) {
if (strncmp(buf,"VHDR",(size_t)4) == 0) {
lsx_readdw(ft, &chunksize);
if (chunksize != 20)
{
lsx_fail_errno(ft, SOX_EHDR, "VHDR chunk has bad size");
return(SOX_EOF);
}
lsx_seeki(ft,(off_t)12,SEEK_CUR);
lsx_readw(ft, &rate);
lsx_seeki(ft,(off_t)1,SEEK_CUR);
lsx_readbuf(ft, buf,(size_t)1);
if (buf[0] != 0)
{
lsx_fail_errno(ft, SOX_EFMT, "Unsupported data compression");
return(SOX_EOF);
}
lsx_seeki(ft,(off_t)4,SEEK_CUR);
continue;
}
if (strncmp(buf,"ANNO",(size_t)4) == 0) {
lsx_readdw(ft, &chunksize);
if (chunksize & 1)
chunksize++;
chunk_buf = lsx_malloc(chunksize + (size_t)2);
if (lsx_readbuf(ft, chunk_buf,(size_t)chunksize)
!= chunksize)
{
lsx_fail_errno(ft, SOX_EHDR, "Couldn't read all of header");
return(SOX_EOF);
}
chunk_buf[chunksize] = '\0';
lsx_debug("%s",chunk_buf);
free(chunk_buf);
continue;
}
if (strncmp(buf,"NAME",(size_t)4) == 0) {
lsx_readdw(ft, &chunksize);
if (chunksize & 1)
chunksize++;
chunk_buf = lsx_malloc(chunksize + (size_t)1);
if (lsx_readbuf(ft, chunk_buf,(size_t)chunksize)
!= chunksize)
{
lsx_fail_errno(ft, SOX_EHDR, "Couldn't read all of header");
return(SOX_EOF);
}
chunk_buf[chunksize] = '\0';
lsx_debug("%s",chunk_buf);
free(chunk_buf);
continue;
}
if (strncmp(buf,"CHAN",(size_t)4) == 0) {
lsx_readdw(ft, &chunksize);
if (chunksize != 4)
{
lsx_fail_errno(ft, SOX_EHDR, "Couldn't read all of header");
return(SOX_EOF);
}
lsx_readdw(ft, &channels);
channels = (channels & 0x01) +
((channels & 0x02) >> 1) +
((channels & 0x04) >> 2) +
((channels & 0x08) >> 3);
continue;
}
/* some other kind of chunk */
lsx_readdw(ft, &chunksize);
if (chunksize & 1)
chunksize++;
lsx_seeki(ft,(off_t)chunksize,SEEK_CUR);
continue;
}
if (rate == 0)
{
lsx_fail_errno(ft, SOX_EHDR, "Invalid sample rate");
return(SOX_EOF);
}
if (strncmp(buf,"BODY",(size_t)4) != 0)
{
lsx_fail_errno(ft, SOX_EHDR, "BODY chunk not found");
return(SOX_EOF);
}
lsx_readdw(ft, &(p->nsamples));
p->left = p->nsamples;
p->ch0_pos = lsx_tell(ft);
ft->signal.length = p->nsamples;
ft->signal.channels = channels;
ft->signal.rate = rate;
ft->encoding.encoding = SOX_ENCODING_SIGN2;
ft->encoding.bits_per_sample = 8;
return(SOX_SUCCESS);
}
/*======================================================================*/
/* 8SVXREAD */
/*======================================================================*/
static size_t read_samples(sox_format_t * ft, sox_sample_t *buf, size_t nsamp)
{
size_t done = 0;
priv_t * p = (priv_t * ) ft->priv;
size_t frames = nsamp / ft->signal.channels;
unsigned width = p->nsamples / ft->signal.channels;
if (p->left < frames)
frames = p->left;
while (done != frames) {
size_t chunk = frames - done;
size_t i;
unsigned ch;
if (chunk > BUFLEN)
chunk = BUFLEN;
for (ch = 0; ch != ft->signal.channels; ch++) {
if (lsx_seeki(ft, p->ch0_pos + ch * width, SEEK_SET) ||
chunk != lsx_readbuf(ft, p->buf[ch], chunk))
return done * ft->signal.channels;
}
for (i = 0; i != chunk; i++) {
for (ch = 0; ch != ft->signal.channels; ch++) {
/* scale signed up to long's range */
*buf++ = SOX_SIGNED_8BIT_TO_SAMPLE(p->buf[ch][i], dummy);
}
}
done += chunk;
p->left -= chunk * ft->signal.channels;
p->ch0_pos += chunk;
}
return done * ft->signal.channels;
}
/*======================================================================*/
/* 8SVXSTARTWRITE */
/*======================================================================*/
static int startwrite(sox_format_t * ft)
{
priv_t * p = (priv_t * ) ft->priv;
size_t i;
/* open channel output files */
for (i = 0; i < ft->signal.channels; i++) {
if ((p->tmp[i] = lsx_tmpfile()) == NULL)
{
lsx_fail_errno(ft,errno,"Can't open channel output file");
return(SOX_EOF);
}
}
p->nsamples = 0;
return(SOX_SUCCESS);
}
/*======================================================================*/
/* 8SVXWRITE */
/*======================================================================*/
static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t len)
{
priv_t * p = (priv_t * ) ft->priv;
SOX_SAMPLE_LOCALS;
unsigned char datum;
size_t done = 0, i;
p->nsamples += len;
while(done < len) {
for (i = 0; i < ft->signal.channels; i++) {
datum = SOX_SAMPLE_TO_SIGNED_8BIT(*buf++, ft->clips);
putc(datum, p->tmp[i]);
}
done += ft->signal.channels;
}
return (done);
}
/*======================================================================*/
/* 8SVXSTOPWRITE */
/*======================================================================*/
static int stopwrite(sox_format_t * ft)
{
priv_t * p = (priv_t * ) ft->priv;
size_t i, len;
char svxbuf[512];
svxwriteheader(ft, (size_t) p->nsamples);
/* append all channel pieces to channel 0 */
/* close temp files */
for (i = 0; i < ft->signal.channels; i++) {
if (fseeko(p->tmp[i], (off_t)0, 0))
{
lsx_fail_errno (ft,errno,"Can't rewind channel output file %lu",(unsigned long)i);
return(SOX_EOF);
}
while (!feof(p->tmp[i])) {
len = fread(svxbuf, (size_t) 1, (size_t) 512, p->tmp[i]);
if (lsx_writebuf(ft, svxbuf, len) != len) {
lsx_fail_errno (ft,errno,"Can't write channel output file %lu",(unsigned long)i);
return SOX_EOF;
}
}
fclose (p->tmp[i]);
}
/* add a pad byte if BODY size is odd */
if(p->nsamples % 2 != 0)
lsx_writeb(ft, '\0');
return(SOX_SUCCESS);
}
/*======================================================================*/
/* 8SVXWRITEHEADER */
/*======================================================================*/
#define SVXHEADERSIZE 100
static void svxwriteheader(sox_format_t * ft, size_t nsamples)
{
size_t formsize = nsamples + SVXHEADERSIZE - 8;
/* FORM size must be even */
if(formsize % 2 != 0) formsize++;
lsx_writes(ft, "FORM");
lsx_writedw(ft, (unsigned) formsize); /* size of file */
lsx_writes(ft, "8SVX"); /* File type */
lsx_writes(ft, "VHDR");
lsx_writedw(ft, 20); /* number of bytes to follow */
lsx_writedw(ft, (unsigned) nsamples/ft->signal.channels); /* samples, 1-shot */
lsx_writedw(ft, 0); /* samples, repeat */
lsx_writedw(ft, 0); /* samples per repeat cycle */
lsx_writew(ft, min(65535, (unsigned)(ft->signal.rate + .5)));
lsx_writeb(ft,1); /* number of octabes */
lsx_writeb(ft,0); /* data compression (none) */
lsx_writew(ft,1); lsx_writew(ft,0); /* volume */
lsx_writes(ft, "ANNO");
lsx_writedw(ft, 32); /* length of block */
lsx_writes(ft, "File created by Sound Exchange ");
lsx_writes(ft, "CHAN");
lsx_writedw(ft, 4);
lsx_writedw(ft, (ft->signal.channels == 2) ? 6u :
(ft->signal.channels == 4) ? 15u : 2u);
lsx_writes(ft, "BODY");
lsx_writedw(ft, (unsigned) nsamples); /* samples in file */
}
LSX_FORMAT_HANDLER(svx)
{
static char const * const names[] = {"8svx", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_SIGN2, 8, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Amiga audio format (a subformat of the Interchange File Format)",
names, SOX_FILE_BIG_END|SOX_FILE_MONO|SOX_FILE_STEREO|SOX_FILE_QUAD,
startread, read_samples, NULL,
startwrite, write_samples, stopwrite,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

184
app/src/main/cpp/sox/Makefile.am
Unescape View File

@ -0,0 +1,184 @@
## Process this file with automake to produce Makefile.in
RM = rm -f
AM_CPPFLAGS = -DLADSPA_PATH="\"@LADSPA_PATH@\""
if HAVE_LIBLTDL
# This is being used as a short cut to turn off versioning of ALL dynamic
# fmt libraries. If any fmt ever needs to add a specific LDFLAGS
# then it will need to also add -avoid-version because AM_LDFLAGS
# is ignored when you specify a more specific one.
# We want to version libsox and we are OK because they
# have a more specific LDFLAGS that includes -version-info.
AM_LDFLAGS = -avoid-version -module
AM_CPPFLAGS += -DPKGLIBDIR="\"$(pkglibdir)\""
endif
#########################
# SoX - the application #
#########################
bin_PROGRAMS = sox
EXTRA_PROGRAMS = example0 example1 example2 example3 example4 example5 example6 sox_sample_test
lib_LTLIBRARIES = libsox.la
include_HEADERS = sox.h
sox_SOURCES = sox.c
if HAVE_WIN32_GLOB
sox_SOURCES += win32/glob.c win32/glob.h
AM_CPPFLAGS += -I$(srcdir)/win32
endif
sox_LDADD = libsox.la
example0_SOURCES = example0.c
example1_SOURCES = example1.c
example2_SOURCES = example2.c
example3_SOURCES = example3.c
example4_SOURCES = example4.c
example5_SOURCES = example5.c
example6_SOURCES = example6.c
sox_sample_test_SOURCES = sox_sample_test.c
######################################################
# libsox - file format, effects, and utility library #
######################################################
# Format handlers and utils source
libsox_la_SOURCES = adpcms.c adpcms.h aiff.c aiff.h cvsd.c cvsd.h cvsdfilt.h \
g711.c g711.h g721.c g723_24.c g723_40.c g72x.c g72x.h vox.c vox.h \
raw.c raw.h formats.c formats.h formats_i.c sox_i.h skelform.c \
xmalloc.c xmalloc.h getopt.c \
util.c util.h libsox.c libsox_i.c sox-fmt.c soxomp.h
# Effects source
libsox_la_SOURCES += \
band.h bend.c biquad.c biquad.h biquads.c chorus.c compand.c \
compandt.c compandt.h contrast.c dcshift.c delay.c dft_filter.c \
dft_filter.h dither.c dither.h divide.c downsample.c earwax.c \
echo.c echos.c effects.c effects.h effects_i.c effects_i_dsp.c \
fade.c fft4g.c fft4g.h fifo.h fir.c firfit.c flanger.c gain.c \
hilbert.c input.c ladspa.h ladspa.c loudness.c mcompand.c \
mcompand_xover.h noiseprof.c noisered.c \
noisered.h output.c overdrive.c pad.c phaser.c rate.c \
rate_filters.h rate_half_fir.h rate_poly_fir0.h rate_poly_fir.h \
remix.c repeat.c reverb.c reverse.c silence.c sinc.c skeleff.c \
speed.c splice.c stat.c stats.c stretch.c swap.c \
synth.c tempo.c tremolo.c trim.c upsample.c vad.c vol.c
if HAVE_PNG
libsox_la_SOURCES += spectrogram.c
endif
libsox_la_LIBADD =
# Libraries required by libsox for file handlers, effects, or utils;
# regardless if libltdl is used or not.
if HAVE_PNG
libsox_la_LIBADD += @PNG_LIBS@
endif
if HAVE_MAGIC
libsox_la_LIBADD += @MAGIC_LIBS@
endif
if HAVE_LIBGSM
libsox_la_LIBADD += @LIBGSM_LIBS@
endif
libsox_la_CFLAGS =
libsox_la_LDFLAGS = -no-undefined -version-info @SHLIB_VERSION@ \
-export-symbols $(srcdir)/libsox.sym
EXTRA_libsox_la_DEPENDENCIES = $(srcdir)/libsox.sym
if HAVE_LIBLTDL
libsox_la_CFLAGS += $(LIBLTDL_CFLAGS)
libsox_la_LIBADD += $(LIBLTDL_LIBS)
endif
#########################
# libsox - File Formats #
#########################
# Uncomment for bit-rot detection on linux
#libsox_la_SOURCES += coreaudio.c sndio.c sunaudio.c
#libsox_la_CFLAGS += -Ibit-rot
libsox_la_SOURCES += raw-fmt.c s1-fmt.c s2-fmt.c s3-fmt.c \
s4-fmt.c u1-fmt.c u2-fmt.c u3-fmt.c u4-fmt.c al-fmt.c la-fmt.c ul-fmt.c \
lu-fmt.c 8svx.c aiff-fmt.c aifc-fmt.c au.c avr.c cdr.c cvsd-fmt.c \
dvms-fmt.c dat.c hcom.c htk.c maud.c prc.c sf.c smp.c \
sounder.c soundtool.c sphere.c tx16w.c voc.c vox-fmt.c ima-fmt.c adpcm.c adpcm.h \
ima_rw.c ima_rw.h wav.c wve.c xa.c nulfile.c f4-fmt.c f8-fmt.c gsrt.c \
id3.c id3.h
if HAVE_ID3TAG
libsox_la_LIBADD += @ID3TAG_LIBS@
endif
pkglib_LTLIBRARIES =
include optional-fmts.am
# example programs will need same link options as sox does.
example0_LDADD = ${sox_LDADD}
example1_LDADD = ${sox_LDADD}
example2_LDADD = ${sox_LDADD}
example3_LDADD = ${sox_LDADD}
example4_LDADD = ${sox_LDADD}
example5_LDADD = ${sox_LDADD}
example6_LDADD = ${sox_LDADD}
EXTRA_DIST = monkey.wav optional-fmts.am \
tests.sh testall.sh tests.bat testall.bat test-comments
all: sox$(EXEEXT)
examples: example0$(EXEEXT) example1$(EXEEXT) example2$(EXEEXT) example3$(EXEEXT) example4$(EXEEXT) example5$(EXEEXT) example6$(EXEEXT)
extras: examples sox_sample_test$(EXEEXT)
MAKELINKS = for n in $(SYMLINKS); do $(RM) $$n$(EXEEXT) && $(LN_S) sox$(EXEEXT) $$n$(EXEEXT); done
all-local: sox$(EXEEXT)
$(MAKELINKS)
install-exec-hook:
cd $(DESTDIR)$(bindir) && $(MAKELINKS)
uninstall-hook:
cd $(DESTDIR)$(bindir) && $(RM) play$(EXEEXT) rec$(EXEEXT) soxi$(EXEEXT)
clean-local:
$(RM) play$(EXEEXT) rec$(EXEEXT) soxi$(EXEEXT)
$(RM) sox_sample_test$(EXEEXT)
$(RM) example0$(EXEEXT) example1$(EXEEXT) example2$(EXEEXT) example3$(EXEEXT) example4$(EXEEXT) example5$(EXEEXT) example6$(EXEEXT)
distclean-local:
loc:
sloccount \
$(include_HEADERS) \
$(sox_SOURCES) \
$(example0_SOURCES) \
$(example1_SOURCES) \
$(example2_SOURCES) \
$(example3_SOURCES) \
$(example4_SOURCES) \
$(example5_SOURCES) \
$(example6_SOURCES) \
$(sox_sample_test_SOURCES) \
$(libsox_la_SOURCES)
# Ideally we would use the "check" target so that "make distcheck"
# would run the test suite, but an uninstalled libltdl build cannot
# currently load its formats and effects, so the checks would fail.
installcheck:
$(srcdir)/tests.sh --bindir=$(DESTDIR)${bindir} --builddir=${builddir} --srcdir=${srcdir}
$(srcdir)/testall.sh --bindir=$(DESTDIR)${bindir} --srcdir=${srcdir}

367
app/src/main/cpp/sox/adpcm.c
Unescape View File

@ -0,0 +1,367 @@
/* adpcm.c codex functions for MS_ADPCM data
* (hopefully) provides interoperability with
* Microsoft's ADPCM format, but, as usual,
* see LACK-OF-WARRANTY information below.
*
* Copyright (C) 1999 Stanley J. Brooks <stabro@megsinet.net>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/*
* November 22, 1999
* specs I've seen are unclear about ADPCM supporting more than 2 channels,
* but these routines support more channels in a manner which looks (IMHO)
* like the most natural extension.
*
* Remark: code still turbulent, encoding very new.
*
*/
#include "sox_i.h"
#include "adpcm.h"
#include <sys/types.h>
#include <stdio.h>
typedef struct {
sox_sample_t step; /* step size */
short coef[2];
} MsState_t;
#define lsbshortldi(x,p) { (x)=((short)((int)(p)[0] + ((int)(p)[1]<<8))); (p) += 2; }
/*
* Lookup tables for MS ADPCM format
*/
/* these are step-size adjust factors, where
* 1.0 is scaled to 0x100
*/
static const
sox_sample_t stepAdjustTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
/* TODO : The first 7 lsx_ms_adpcm_i_coef sets are always hardcoded and must
appear in the actual WAVE file. They should be read in
in case a sound program added extras to the list. */
const short lsx_ms_adpcm_i_coef[7][2] = {
{ 256, 0},
{ 512,-256},
{ 0, 0},
{ 192, 64},
{ 240, 0},
{ 460,-208},
{ 392,-232}
};
extern void *lsx_ms_adpcm_alloc(unsigned chans)
{
return lsx_malloc(chans * sizeof(MsState_t));
}
static inline sox_sample_t AdpcmDecode(sox_sample_t c, MsState_t *state,
sox_sample_t sample1, sox_sample_t sample2)
{
sox_sample_t vlin;
sox_sample_t sample;
sox_sample_t step;
/** Compute next step value **/
step = state->step;
{
sox_sample_t nstep;
nstep = (stepAdjustTable[c] * step) >> 8;
state->step = (nstep < 16)? 16:nstep;
}
/** make linear prediction for next sample **/
vlin =
((sample1 * state->coef[0]) +
(sample2 * state->coef[1])) >> 8;
/** then add the code*step adjustment **/
c -= (c & 0x08) << 1;
sample = (c * step) + vlin;
if (sample > 0x7fff) sample = 0x7fff;
else if (sample < -0x8000) sample = -0x8000;
return (sample);
}
/* lsx_ms_adpcm_block_expand_i() outputs interleaved samples into one output buffer */
const char *lsx_ms_adpcm_block_expand_i(
void *priv,
unsigned chans, /* total channels */
int nCoef,
const short *coef,
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* output samples, n*chans */
int n /* samples to decode PER channel */
)
{
const unsigned char *ip;
unsigned ch;
const char *errmsg = NULL;
MsState_t *state = priv; /* One decompressor state for each channel */
/* Read the four-byte header for each channel */
ip = ibuff;
for (ch = 0; ch < chans; ch++) {
unsigned char bpred = *ip++;
if (bpred >= nCoef) {
errmsg = "MSADPCM bpred >= nCoef, arbitrarily using 0\n";
bpred = 0;
}
state[ch].coef[0] = coef[(int)bpred*2+0];
state[ch].coef[1] = coef[(int)bpred*2+1];
}
for (ch = 0; ch < chans; ch++)
lsbshortldi(state[ch].step, ip);
/* sample1's directly into obuff */
for (ch = 0; ch < chans; ch++)
lsbshortldi(obuff[chans+ch], ip);
/* sample2's directly into obuff */
for (ch = 0; ch < chans; ch++)
lsbshortldi(obuff[ch], ip);
{
unsigned ch2;
unsigned char b;
short *op, *top, *tmp;
/* already have 1st 2 samples from block-header */
op = obuff + 2*chans;
top = obuff + n*chans;
ch2 = 0;
while (op < top) { /*** N.B. Without int casts, crashes on 64-bit arch ***/
b = *ip++;
tmp = op;
*op++ = AdpcmDecode(b >> 4, state+ch2, tmp[-(int)chans], tmp[-(int)(2*chans)]);
if (++ch2 == chans) ch2 = 0;
tmp = op;
*op++ = AdpcmDecode(b&0x0f, state+ch2, tmp[-(int)chans], tmp[-(int)(2*chans)]);
if (++ch2 == chans) ch2 = 0;
}
}
return errmsg;
}
static int AdpcmMashS(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
SAMPL v[2], /* values to use as starting 2 */
const short coef[2],/* lin predictor coeffs */
const SAMPL *ibuff, /* ibuff[] is interleaved input samples */
int n, /* samples to encode PER channel */
int *iostep, /* input/output step, REQUIRE 16 <= *st <= 0x7fff */
unsigned char *obuff /* output buffer[blockAlign], or NULL for no output */
)
{
const SAMPL *ip, *itop;
unsigned char *op;
int ox = 0; /* */
int d, v0, v1, step;
double d2; /* long long is okay also, speed abt the same */
ip = ibuff + ch; /* point ip to 1st input sample for this channel */
itop = ibuff + n*chans;
v0 = v[0];
v1 = v[1];
d = *ip - v1; ip += chans; /* 1st input sample for this channel */
d2 = d*d; /* d2 will be sum of squares of errors, given input v0 and *st */
d = *ip - v0; ip += chans; /* 2nd input sample for this channel */
d2 += d*d;
step = *iostep;
op = obuff; /* output pointer (or NULL) */
if (op) { /* NULL means don't output, just compute the rms error */
op += chans; /* skip bpred indices */
op += 2*ch; /* channel's stepsize */
op[0] = step; op[1] = step>>8;
op += 2*chans; /* skip to v0 */
op[0] = v0; op[1] = v0>>8;
op += 2*chans; /* skip to v1 */
op[0] = v1; op[1] = v1>>8;
op = obuff+7*chans; /* point to base of output nibbles */
ox = 4*ch;
}
for (; ip < itop; ip+=chans) {
int vlin,d3,dp,c;
/* make linear prediction for next sample */
vlin = (v0 * coef[0] + v1 * coef[1]) >> 8;
d3 = *ip - vlin; /* difference between linear prediction and current sample */
dp = d3 + (step<<3) + (step>>1);
c = 0;
if (dp>0) {
c = dp/step;
if (c>15) c = 15;
}
c -= 8;
dp = c * step; /* quantized estimate of samp - vlin */
c &= 0x0f; /* mask to 4 bits */
v1 = v0; /* shift history */
v0 = vlin + dp;
if (v0<-0x8000) v0 = -0x8000;
else if (v0>0x7fff) v0 = 0x7fff;
d3 = *ip - v0;
d2 += d3*d3; /* update square-error */
if (op) { /* if we want output, put it in proper place */
op[ox>>3] |= (ox&4)? c:(c<<4);
ox += 4*chans;
lsx_debug_more("%.1x",c);
}
/* Update the step for the next sample */
step = (stepAdjustTable[c] * step) >> 8;
if (step < 16) step = 16;
}
if (op) lsx_debug_more("\n");
d2 /= n; /* be sure it's non-negative */
lsx_debug_more("ch%d: st %d->%d, d %.1f\n", ch, *iostep, step, sqrt(d2));
*iostep = step;
return (int) sqrt(d2);
}
static inline void AdpcmMashChannel(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE */
int *st, /* input/output steps, 16<=st[i] */
unsigned char *obuff /* output buffer[blockAlign] */
)
{
SAMPL v[2];
int n0,s0,s1,ss,smin;
int dmin,k,kmin;
n0 = n/2; if (n0>32) n0=32;
if (*st<16) *st = 16;
v[1] = ip[ch];
v[0] = ip[ch+chans];
dmin = 0; kmin = 0; smin = 0;
/* for each of 7 standard coeff sets, we try compression
* beginning with last step-value, and with slightly
* forward-adjusted step-value, taking best of the 14
*/
for (k=0; k<7; k++) {
int d0,d1;
ss = s0 = *st;
d0=AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n, &ss, NULL); /* with step s0 */
s1 = s0;
AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n0, &s1, NULL);
lsx_debug_more(" s32 %d\n",s1);
ss = s1 = (3*s0+s1)/4;
d1=AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n, &ss, NULL); /* with step s1 */
if (!k || d0<dmin || d1<dmin) {
kmin = k;
if (d0<=d1) {
dmin = d0;
smin = s0;
}else{
dmin = d1;
smin = s1;
}
}
}
*st = smin;
lsx_debug_more("kmin %d, smin %5d, ",kmin,smin);
AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[kmin], ip, n, st, obuff);
obuff[ch] = kmin;
}
void lsx_ms_adpcm_block_mash_i(
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[n*chans] is interleaved input samples */
int n, /* samples to encode PER channel */
int *st, /* input/output steps, 16<=st[i] */
unsigned char *obuff, /* output buffer[blockAlign] */
int blockAlign /* >= 7*chans + chans*(n-2)/2.0 */
)
{
unsigned ch;
unsigned char *p;
lsx_debug_more("AdpcmMashI(chans %d, ip %p, n %d, st %p, obuff %p, bA %d)\n",
chans, (void *)ip, n, (void *)st, obuff, blockAlign);
for (p=obuff+7*chans; p<obuff+blockAlign; p++) *p=0;
for (ch=0; ch<chans; ch++)
AdpcmMashChannel(ch, chans, ip, n, st+ch, obuff);
}
/*
* lsx_ms_adpcm_samples_in(dataLen, chans, blockAlign, samplesPerBlock)
* returns the number of samples/channel which would be
* in the dataLen, given the other parameters ...
* if input samplesPerBlock is 0, then returns the max
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
size_t lsx_ms_adpcm_samples_in(
size_t dataLen,
size_t chans,
size_t blockAlign,
size_t samplesPerBlock
){
size_t m, n;
if (samplesPerBlock) {
n = (dataLen / blockAlign) * samplesPerBlock;
m = (dataLen % blockAlign);
} else {
n = 0;
m = blockAlign;
}
if (m >= (size_t)(7*chans)) {
m -= 7*chans; /* bytes beyond block-header */
m = (2*m)/chans + 2; /* nibbles/chans + 2 in header */
if (samplesPerBlock && m > samplesPerBlock) m = samplesPerBlock;
n += m;
}
return n;
}
size_t lsx_ms_adpcm_bytes_per_block(
size_t chans,
size_t samplesPerBlock
)
{
size_t n;
n = 7*chans; /* header */
if (samplesPerBlock > 2)
n += (((size_t)samplesPerBlock-2)*chans + 1)/2;
return n;
}

79
app/src/main/cpp/sox/adpcm.h
Unescape View File

@ -0,0 +1,79 @@
/* adpcm.h codex functions for MS_ADPCM data
* (hopefully) provides interoperability with
* Microsoft's ADPCM format, but, as usual,
* see LACK-OF-WARRANTY information below.
*
* Copyright (C) 1999 Stanley J. Brooks <stabro@megsinet.net>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#include "sox.h"
#ifndef SAMPL
#define SAMPL short
#endif
/* default coef sets */
extern const short lsx_ms_adpcm_i_coef[7][2];
extern void *lsx_ms_adpcm_alloc(unsigned chans);
/* lsx_ms_adpcm_block_expand_i() outputs interleaved samples into one output buffer */
extern const char *lsx_ms_adpcm_block_expand_i(
void *priv,
unsigned chans, /* total channels */
int nCoef,
const short *coef,
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* output samples, n*chans */
int n /* samples to decode PER channel, REQUIRE n % 8 == 1 */
);
extern void lsx_ms_adpcm_block_mash_i(
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[n*chans] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE */
int *st, /* input/output steps, 16<=st[i] */
unsigned char *obuff, /* output buffer[blockAlign] */
int blockAlign /* >= 7*chans + n/2 */
);
/* Some helper functions for computing samples/block and blockalign */
/*
* lsx_ms_adpcm_samples_in(dataLen, chans, blockAlign, samplesPerBlock)
* returns the number of samples/channel which would be
* in the dataLen, given the other parameters ...
* if input samplesPerBlock is 0, then returns the max
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
extern size_t lsx_ms_adpcm_samples_in(
size_t dataLen,
size_t chans,
size_t blockAlign,
size_t samplesPerBlock
);
/*
* size_t lsx_ms_adpcm_bytes_per_block(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
extern size_t lsx_ms_adpcm_bytes_per_block(
size_t chans,
size_t samplesPerBlock
);

322
app/src/main/cpp/sox/adpcms.c
Unescape View File

@ -0,0 +1,322 @@
/* libSoX ADPCM codecs: IMA, OKI, CL. (c) 2007-8 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "adpcms.h"
static int const ima_steps[89] = { /* ~16-bit precision; 4 bit code */
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230,
253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767};
static int const oki_steps[49] = { /* ~12-bit precision; 4 bit code */
256, 272, 304, 336, 368, 400, 448, 496, 544, 592, 656, 720, 800, 880, 960,
1056, 1168, 1280, 1408, 1552, 1712, 1888, 2080, 2288, 2512, 2768, 3040, 3344,
3680, 4048, 4464, 4912, 5392, 5936, 6528, 7184, 7904, 8704, 9568, 10528,
11584, 12736, 14016, 15408, 16960, 18656, 20512, 22576, 24832};
static int const step_changes[8] = {-1, -1, -1, -1, 2, 4, 6, 8};
/* Creative Labs ~8 bit precision; 4, 3, & 2 bit codes: */
static int const cl4_steps[4] = {0x100, 0x200, 0x400, 0x800};
static int const cl4_changes[8] = {-1, 0, 0, 0, 0, 1, 1, 1};
static int const cl3_steps[5] = {0x100, 0x200, 0x400, 0x800, 0xA00};
static int const cl3_changes[4] = {-1, 0, 0, 1};
static int const cl2_steps[6] = {0x100, 0x200, 0x400, 0x800, 0x1000, 0x2000};
static int const cl2_changes[2] = {-1, 1};
static adpcm_setup_t const setup_table[] = {
{88, 8, 2, ima_steps, step_changes, ~0},
{48, 8, 2, oki_steps, step_changes, ~15},
{ 3, 8, 0, cl4_steps, cl4_changes , ~255},
{ 4, 4, 0, cl3_steps, cl3_changes , ~255},
{ 5, 2, 0, cl2_steps, cl2_changes , ~255},
};
void lsx_adpcm_init(adpcm_t * p, int type, int first_sample)
{
p->setup = setup_table[type];
p->last_output = first_sample;
p->step_index = 0;
p->errors = 0;
}
#define min_sample -0x8000
#define max_sample 0x7fff
int lsx_adpcm_decode(int code, adpcm_t * p)
{
int s = ((code & (p->setup.sign - 1)) << 1) | 1;
s = ((p->setup.steps[p->step_index] * s) >> (p->setup.shift + 1)) & p->setup.mask;
if (code & p->setup.sign)
s = -s;
s += p->last_output;
if (s < min_sample || s > max_sample) {
int grace = (p->setup.steps[p->step_index] >> (p->setup.shift + 1)) & p->setup.mask;
if (s < min_sample - grace || s > max_sample + grace) {
lsx_debug_most("code=%i step=%i grace=%i s=%i",
code & (2 * p->setup.sign - 1), p->setup.steps[p->step_index], grace, s);
p->errors++;
}
s = s < min_sample? min_sample : max_sample;
}
p->step_index += p->setup.changes[code & (p->setup.sign - 1)];
p->step_index = range_limit(p->step_index, 0, p->setup.max_step_index);
return p->last_output = s;
}
int lsx_adpcm_encode(int sample, adpcm_t * p)
{
int delta = sample - p->last_output;
int sign = 0;
int code;
if (delta < 0) {
sign = p->setup.sign;
delta = -delta;
}
code = (delta << p->setup.shift) / p->setup.steps[p->step_index];
code = sign | min(code, p->setup.sign - 1);
lsx_adpcm_decode(code, p); /* Update encoder state */
return code;
}
/*
* Format methods
*
* Almost like the raw format functions, but cannot be used directly
* since they require an additional state parameter.
*/
/******************************************************************************
* Function : lsx_adpcm_reset
* Description: Resets the ADPCM codec state.
* Parameters : state - ADPCM state structure
* type - SOX_ENCODING_OKI_ADPCM or SOX_ENCODING_IMA_ADPCM
* Returns :
* Exceptions :
* Notes : 1. This function is used for framed ADPCM formats to reset
* the decoder between frames.
******************************************************************************/
void lsx_adpcm_reset(adpcm_io_t * state, sox_encoding_t type)
{
state->file.count = 0;
state->file.pos = 0;
state->store.byte = 0;
state->store.flag = 0;
lsx_adpcm_init(&state->encoder, (type == SOX_ENCODING_OKI_ADPCM) ? 1 : 0, 0);
}
/******************************************************************************
* Function : lsx_adpcm_start
* Description: Initialises the file parameters and ADPCM codec state.
* Parameters : ft - file info structure
* state - ADPCM state structure
* type - SOX_ENCODING_OKI_ADPCM or SOX_ENCODING_IMA_ADPCM
* Returns : int - SOX_SUCCESS
* SOX_EOF
* Exceptions :
* Notes : 1. This function can be used as a startread or
* startwrite method.
* 2. VOX file format is 4-bit OKI ADPCM that decodes to
* to 12 bit signed linear PCM.
* 3. Dialogic only supports 6kHz, 8kHz and 11 kHz sampling
* rates but the codecs allows any user specified rate.
******************************************************************************/
static int adpcm_start(sox_format_t * ft, adpcm_io_t * state, sox_encoding_t type)
{
/* setup file info */
state->file.buf = lsx_malloc(sox_globals.bufsiz);
state->file.size = sox_globals.bufsiz;
ft->signal.channels = 1;
lsx_adpcm_reset(state, type);
return lsx_rawstart(ft, sox_true, sox_false, sox_true, type, 4);
}
int lsx_adpcm_oki_start(sox_format_t * ft, adpcm_io_t * state)
{
return adpcm_start(ft, state, SOX_ENCODING_OKI_ADPCM);
}
int lsx_adpcm_ima_start(sox_format_t * ft, adpcm_io_t * state)
{
return adpcm_start(ft, state, SOX_ENCODING_IMA_ADPCM);
}
/******************************************************************************
* Function : lsx_adpcm_read
* Description: Converts the OKI ADPCM 4-bit samples to 16-bit signed PCM and
* then scales the samples to full sox_sample_t range.
* Parameters : ft - file info structure
* state - ADPCM state structure
* buffer - output buffer
* len - size of output buffer
* Returns : - number of samples returned in buffer
* Exceptions :
* Notes :
******************************************************************************/
size_t lsx_adpcm_read(sox_format_t * ft, adpcm_io_t * state, sox_sample_t * buffer, size_t len)
{
size_t n = 0;
uint8_t byte;
int16_t word;
if (len && state->store.flag) {
word = lsx_adpcm_decode(state->store.byte, &state->encoder);
*buffer++ = SOX_SIGNED_16BIT_TO_SAMPLE(word, ft->clips);
state->store.flag = 0;
++n;
}
while (n < len && lsx_read_b_buf(ft, &byte, (size_t) 1) == 1) {
word = lsx_adpcm_decode(byte >> 4, &state->encoder);
*buffer++ = SOX_SIGNED_16BIT_TO_SAMPLE(word, ft->clips);
if (++n < len) {
word = lsx_adpcm_decode(byte, &state->encoder);
*buffer++ = SOX_SIGNED_16BIT_TO_SAMPLE(word, ft->clips);
++n;
} else {
state->store.byte = byte;
state->store.flag = 1;
}
}
return n;
}
/******************************************************************************
* Function : stopread
* Description: Frees the internal buffer allocated in voxstart/imastart.
* Parameters : ft - file info structure
* state - ADPCM state structure
* Returns : int - SOX_SUCCESS
* Exceptions :
* Notes :
******************************************************************************/
int lsx_adpcm_stopread(sox_format_t * ft UNUSED, adpcm_io_t * state)
{
if (state->encoder.errors)
lsx_warn("%s: ADPCM state errors: %u", ft->filename, state->encoder.errors);
free(state->file.buf);
return (SOX_SUCCESS);
}
/******************************************************************************
* Function : write
* Description: Converts the supplied buffer to 12 bit linear PCM and encodes
* to OKI ADPCM 4-bit samples (packed a two nibbles per byte).
* Parameters : ft - file info structure
* state - ADPCM state structure
* buffer - output buffer
* length - size of output buffer
* Returns : int - SOX_SUCCESS
* SOX_EOF
* Exceptions :
* Notes :
******************************************************************************/
size_t lsx_adpcm_write(sox_format_t * ft, adpcm_io_t * state, const sox_sample_t * buffer, size_t length)
{
size_t count = 0;
uint8_t byte = state->store.byte;
uint8_t flag = state->store.flag;
short word;
while (count < length) {
SOX_SAMPLE_LOCALS;
word = SOX_SAMPLE_TO_SIGNED_16BIT(*buffer++, ft->clips);
byte <<= 4;
byte |= lsx_adpcm_encode(word, &state->encoder) & 0x0F;
flag = !flag;
if (flag == 0) {
state->file.buf[state->file.count++] = byte;
if (state->file.count >= state->file.size) {
lsx_writebuf(ft, state->file.buf, state->file.count);
state->file.count = 0;
}
}
count++;
}
/* keep last byte across calls */
state->store.byte = byte;
state->store.flag = flag;
return (count);
}
/******************************************************************************
* Function : lsx_adpcm_flush
* Description: Flushes any leftover samples.
* Parameters : ft - file info structure
* state - ADPCM state structure
* Returns :
* Exceptions :
* Notes : 1. Called directly for writing framed formats
******************************************************************************/
void lsx_adpcm_flush(sox_format_t * ft, adpcm_io_t * state)
{
uint8_t byte = state->store.byte;
uint8_t flag = state->store.flag;
/* flush remaining samples */
if (flag != 0) {
byte <<= 4;
state->file.buf[state->file.count++] = byte;
}
if (state->file.count > 0)
lsx_writebuf(ft, state->file.buf, state->file.count);
}
/******************************************************************************
* Function : lsx_adpcm_stopwrite
* Description: Flushes any leftover samples and frees the internal buffer
* allocated in voxstart/imastart.
* Parameters : ft - file info structure
* state - ADPCM state structure
* Returns : int - SOX_SUCCESS
* Exceptions :
* Notes :
******************************************************************************/
int lsx_adpcm_stopwrite(sox_format_t * ft, adpcm_io_t * state)
{
lsx_adpcm_flush(ft, state);
free(state->file.buf);
return (SOX_SUCCESS);
}

55
app/src/main/cpp/sox/adpcms.h
Unescape View File

@ -0,0 +1,55 @@
/* libSoX ADPCM codecs: IMA, OKI, CL. (c) 2007-8 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
typedef struct {
int max_step_index;
int sign;
int shift;
int const * steps;
int const * changes;
int mask;
} adpcm_setup_t;
typedef struct {
adpcm_setup_t setup;
int last_output;
int step_index;
int errors;
} adpcm_t;
void lsx_adpcm_init(adpcm_t * p, int type, int first_sample);
int lsx_adpcm_decode(int code, adpcm_t * p);
int lsx_adpcm_encode(int sample, adpcm_t * p);
typedef struct {
adpcm_t encoder;
struct {
uint8_t byte; /* write store */
uint8_t flag;
} store;
sox_fileinfo_t file;
} adpcm_io_t;
/* Format methods */
void lsx_adpcm_reset(adpcm_io_t * state, sox_encoding_t type);
int lsx_adpcm_oki_start(sox_format_t * ft, adpcm_io_t * state);
int lsx_adpcm_ima_start(sox_format_t * ft, adpcm_io_t * state);
size_t lsx_adpcm_read(sox_format_t * ft, adpcm_io_t * state, sox_sample_t *buffer, size_t len);
int lsx_adpcm_stopread(sox_format_t * ft, adpcm_io_t * state);
size_t lsx_adpcm_write(sox_format_t * ft, adpcm_io_t * state, const sox_sample_t *buffer, size_t length);
void lsx_adpcm_flush(sox_format_t * ft, adpcm_io_t * state);
int lsx_adpcm_stopwrite(sox_format_t * ft, adpcm_io_t * state);

36
app/src/main/cpp/sox/aifc-fmt.c
Unescape View File

@ -0,0 +1,36 @@
/* File format: AIFF-C (see aiff.c) (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "aiff.h"
LSX_FORMAT_HANDLER(aifc)
{
static char const * const names[] = {"aifc", "aiffc", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2, 32, 24, 16, 8, 0,
SOX_ENCODING_FLOAT, 32, 64, 0,
0};
static sox_format_handler_t const sox_aifc_format = {SOX_LIB_VERSION_CODE,
"AIFF-C (not compressed), defined in DAVIC 1.4 Part 9 Annex B",
names, SOX_FILE_BIG_END,
lsx_aiffstartread, lsx_rawread, lsx_aiffstopread,
lsx_aifcstartwrite, lsx_rawwrite, lsx_aifcstopwrite,
lsx_rawseek, write_encodings, NULL, 0
};
return &sox_aifc_format;
}

33
app/src/main/cpp/sox/aiff-fmt.c
Unescape View File

@ -0,0 +1,33 @@
/* File format: AIFF (see aiff.c) (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "aiff.h"
LSX_FORMAT_HANDLER(aiff)
{
static char const * const names[] = {"aiff", "aif", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2, 32, 24, 16, 8, 0, 0};
static sox_format_handler_t const sox_aiff_format = {SOX_LIB_VERSION_CODE,
"AIFF files used on Apple IIc/IIgs and SGI", names, SOX_FILE_BIG_END,
lsx_aiffstartread, lsx_rawread, lsx_aiffstopread,
lsx_aiffstartwrite, lsx_rawwrite, lsx_aiffstopwrite,
lsx_rawseek, write_encodings, NULL, 0
};
return &sox_aiff_format;
}

1151
app/src/main/cpp/sox/aiff.c
View File

File diff suppressed because it is too large Load Diff

22
app/src/main/cpp/sox/aiff.h
Unescape View File

@ -0,0 +1,22 @@
/* libSoX SGI/Amiga AIFF format.
* Copyright 1991-2007 Guido van Rossum And Sundry Contributors
*
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Guido van Rossum And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* Used by SGI on 4D/35 and Indigo.
* This is a subformat of the EA-IFF-85 format.
* This is related to the IFF format used by the Amiga.
* But, apparently, not the same.
* Also AIFF-C format output that is defined in DAVIC 1.4 Part 9 Annex B
* (usable for japanese-data-broadcasting, specified by ARIB STD-B24.)
*/
int lsx_aiffstartread(sox_format_t * ft);
int lsx_aiffstopread(sox_format_t * ft);
int lsx_aiffstartwrite(sox_format_t * ft);
int lsx_aiffstopwrite(sox_format_t * ft);
int lsx_aifcstartwrite(sox_format_t * ft);
int lsx_aifcstopwrite(sox_format_t * ft);

21
app/src/main/cpp/sox/al-fmt.c
Unescape View File

@ -0,0 +1,21 @@
/* File format: raw A-law (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "raw.h"
RAW_FORMAT(al, 8, 0, ALAW)

379
app/src/main/cpp/sox/alsa.c
Unescape View File

@ -0,0 +1,379 @@
/* libSoX device driver: ALSA (c) 2006-2012 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <alsa/asoundlib.h>
typedef struct {
snd_pcm_uframes_t buf_len, period;
snd_pcm_t * pcm;
char * buf;
unsigned int format;
} priv_t;
static const
struct {
unsigned int bits;
enum _snd_pcm_format alsa_fmt;
unsigned int bytes; /* occupied in the buffer per sample */
sox_encoding_t enc;
} formats[] = {
/* order by # of bits; within that, preferred first */
{ 8, SND_PCM_FORMAT_S8, 1, SOX_ENCODING_SIGN2 },
{ 8, SND_PCM_FORMAT_U8, 1, SOX_ENCODING_UNSIGNED },
{ 16, SND_PCM_FORMAT_S16, 2, SOX_ENCODING_SIGN2 },
{ 16, SND_PCM_FORMAT_U16, 2, SOX_ENCODING_UNSIGNED },
{ 24, SND_PCM_FORMAT_S24, 4, SOX_ENCODING_SIGN2 },
{ 24, SND_PCM_FORMAT_U24, 4, SOX_ENCODING_UNSIGNED },
{ 24, SND_PCM_FORMAT_S24_3LE, 3, SOX_ENCODING_SIGN2 },
{ 32, SND_PCM_FORMAT_S32, 4, SOX_ENCODING_SIGN2 },
{ 32, SND_PCM_FORMAT_U32, 4, SOX_ENCODING_UNSIGNED },
{ 0, 0, 0, SOX_ENCODING_UNKNOWN } /* end of list */
};
static int select_format(
sox_encoding_t * encoding_,
unsigned * nbits_,
snd_pcm_format_mask_t const * mask,
unsigned int * format)
{
unsigned int from = 0, to; /* NB: "to" actually points one after the last */
int cand = -1;
while (formats[from].bits < *nbits_ && formats[from].bits != 0)
from++; /* find the first entry with at least *nbits_ bits */
for (to = from; formats[to].bits != 0; to++) ; /* find end of list */
while (to > 0) {
unsigned int i, bits_next = 0;
for (i = from; i < to; i++) {
lsx_debug_most("select_format: trying #%u", i);
if (snd_pcm_format_mask_test(mask, formats[i].alsa_fmt)) {
if (formats[i].enc == *encoding_) {
cand = i;
break; /* found a match */
} else if (cand == -1) /* don't overwrite a candidate that
was earlier in the list */
cand = i; /* will work, but encoding differs */
}
}
if (cand != -1)
break;
/* no candidate found yet; now try formats with less bits: */
to = from;
if (from > 0)
bits_next = formats[from-1].bits;
while (from && formats[from-1].bits == bits_next)
from--; /* go back to the first entry with bits_next bits */
}
if (cand == -1) {
lsx_debug("select_format: no suitable ALSA format found");
return -1;
}
if (*nbits_ != formats[cand].bits || *encoding_ != formats[cand].enc) {
lsx_warn("can't encode %u-bit %s", *nbits_,
sox_encodings_info[*encoding_].desc);
*nbits_ = formats[cand].bits;
*encoding_ = formats[cand].enc;
}
lsx_debug("selecting format %d: %s (%s)", cand,
snd_pcm_format_name(formats[cand].alsa_fmt),
snd_pcm_format_description(formats[cand].alsa_fmt));
*format = cand;
return 0;
}
#define _(x,y) do {if ((err = x y) < 0) {lsx_fail_errno(ft, SOX_EPERM, #x " error: %s", snd_strerror(err)); goto error;} } while (0)
static int setup(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
snd_pcm_hw_params_t * params = NULL;
snd_pcm_format_mask_t * mask = NULL;
snd_pcm_uframes_t min, max;
unsigned n;
int err;
_(snd_pcm_open, (&p->pcm, ft->filename, ft->mode == 'r'? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 0));
_(snd_pcm_hw_params_malloc, (&params));
_(snd_pcm_hw_params_any, (p->pcm, params));
#if SND_LIB_VERSION >= 0x010009 /* Disable alsa-lib resampling: */
_(snd_pcm_hw_params_set_rate_resample, (p->pcm, params, 0));
#endif
_(snd_pcm_hw_params_set_access, (p->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED));
_(snd_pcm_format_mask_malloc, (&mask)); /* Set format: */
snd_pcm_hw_params_get_format_mask(params, mask);
_(select_format, (&ft->encoding.encoding, &ft->encoding.bits_per_sample, mask, &p->format));
_(snd_pcm_hw_params_set_format, (p->pcm, params, formats[p->format].alsa_fmt));
snd_pcm_format_mask_free(mask), mask = NULL;
n = ft->signal.rate; /* Set rate: */
_(snd_pcm_hw_params_set_rate_near, (p->pcm, params, &n, 0));
ft->signal.rate = n;
n = ft->signal.channels; /* Set channels: */
_(snd_pcm_hw_params_set_channels_near, (p->pcm, params, &n));
ft->signal.channels = n;
/* Get number of significant bits: */
if ((err = snd_pcm_hw_params_get_sbits(params)) > 0)
ft->signal.precision = min(err, SOX_SAMPLE_PRECISION);
else lsx_debug("snd_pcm_hw_params_get_sbits can't tell precision: %s",
snd_strerror(err));
/* Set buf_len > > sox_globals.bufsiz for no underrun: */
p->buf_len = sox_globals.bufsiz * 8 / formats[p->format].bytes /
ft->signal.channels;
_(snd_pcm_hw_params_get_buffer_size_min, (params, &min));
_(snd_pcm_hw_params_get_buffer_size_max, (params, &max));
p->period = range_limit(p->buf_len, min, max) / 8;
p->buf_len = p->period * 8;
_(snd_pcm_hw_params_set_period_size_near, (p->pcm, params, &p->period, 0));
_(snd_pcm_hw_params_set_buffer_size_near, (p->pcm, params, &p->buf_len));
if (p->period * 2 > p->buf_len) {
lsx_fail_errno(ft, SOX_EPERM, "buffer too small");
goto error;
}
_(snd_pcm_hw_params, (p->pcm, params)); /* Configure ALSA */
snd_pcm_hw_params_free(params), params = NULL;
_(snd_pcm_prepare, (p->pcm));
p->buf_len *= ft->signal.channels; /* No longer in `frames' */
p->buf = lsx_malloc(p->buf_len * formats[p->format].bytes);
return SOX_SUCCESS;
error:
if (mask) snd_pcm_format_mask_free(mask);
if (params) snd_pcm_hw_params_free(params);
return SOX_EOF;
}
static int recover(sox_format_t * ft, snd_pcm_t * pcm, int err)
{
if (err == -EPIPE)
lsx_warn("%s-run", ft->mode == 'r'? "over" : "under");
else if (err != -ESTRPIPE)
lsx_warn("%s", snd_strerror(err));
else while ((err = snd_pcm_resume(pcm)) == -EAGAIN) {
lsx_report("suspended");
sleep(1); /* Wait until the suspend flag is released */
}
if (err < 0 && (err = snd_pcm_prepare(pcm)) < 0)
lsx_fail_errno(ft, SOX_EPERM, "%s", snd_strerror(err));
return err;
}
static size_t read_(sox_format_t * ft, sox_sample_t * buf, size_t len)
{
priv_t * p = (priv_t *)ft->priv;
snd_pcm_sframes_t i, n;
size_t done;
len = min(len, p->buf_len);
for (done = 0; done < len; done += n) {
do {
n = snd_pcm_readi(p->pcm, p->buf, (len - done) / ft->signal.channels);
if (n < 0 && recover(ft, p->pcm, (int)n) < 0)
return 0;
} while (n <= 0);
i = n *= ft->signal.channels;
switch (formats[p->format].alsa_fmt) {
case SND_PCM_FORMAT_S8: {
int8_t * buf1 = (int8_t *)p->buf;
while (i--) *buf++ = SOX_SIGNED_8BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_U8: {
uint8_t * buf1 = (uint8_t *)p->buf;
while (i--) *buf++ = SOX_UNSIGNED_8BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_S16: {
int16_t * buf1 = (int16_t *)p->buf;
if (ft->encoding.reverse_bytes) while (i--)
*buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(lsx_swapw(*buf1++),);
else
while (i--) *buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_U16: {
uint16_t * buf1 = (uint16_t *)p->buf;
if (ft->encoding.reverse_bytes) while (i--)
*buf++ = SOX_UNSIGNED_16BIT_TO_SAMPLE(lsx_swapw(*buf1++),);
else
while (i--) *buf++ = SOX_UNSIGNED_16BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_S24: {
sox_int24_t * buf1 = (sox_int24_t *)p->buf;
while (i--) *buf++ = SOX_SIGNED_24BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_S24_3LE: {
unsigned char *buf1 = (unsigned char *)p->buf;
while (i--) {
uint32_t temp;
temp = *buf1++;
temp |= *buf1++ << 8;
temp |= *buf1++ << 16;
*buf++ = SOX_SIGNED_24BIT_TO_SAMPLE((sox_int24_t)temp,);
}
break;
}
case SND_PCM_FORMAT_U24: {
sox_uint24_t * buf1 = (sox_uint24_t *)p->buf;
while (i--) *buf++ = SOX_UNSIGNED_24BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_S32: {
int32_t * buf1 = (int32_t *)p->buf;
while (i--) *buf++ = SOX_SIGNED_32BIT_TO_SAMPLE(*buf1++,);
break;
}
case SND_PCM_FORMAT_U32: {
uint32_t * buf1 = (uint32_t *)p->buf;
while (i--) *buf++ = SOX_UNSIGNED_32BIT_TO_SAMPLE(*buf1++,);
break;
}
default: lsx_fail_errno(ft, SOX_EFMT, "invalid format");
return 0;
}
}
return len;
}
static size_t write_(sox_format_t * ft, sox_sample_t const * buf, size_t len)
{
priv_t * p = (priv_t *)ft->priv;
size_t done, i, n;
snd_pcm_sframes_t actual;
SOX_SAMPLE_LOCALS;
for (done = 0; done < len; done += n) {
i = n = min(len - done, p->buf_len);
switch (formats[p->format].alsa_fmt) {
case SND_PCM_FORMAT_S8: {
int8_t * buf1 = (int8_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_SIGNED_8BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_U8: {
uint8_t * buf1 = (uint8_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_UNSIGNED_8BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_S16: {
int16_t * buf1 = (int16_t *)p->buf;
if (ft->encoding.reverse_bytes) while (i--)
*buf1++ = lsx_swapw(SOX_SAMPLE_TO_SIGNED_16BIT(*buf++, ft->clips));
else
while (i--) *buf1++ = SOX_SAMPLE_TO_SIGNED_16BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_U16: {
uint16_t * buf1 = (uint16_t *)p->buf;
if (ft->encoding.reverse_bytes) while (i--)
*buf1++ = lsx_swapw(SOX_SAMPLE_TO_UNSIGNED_16BIT(*buf++, ft->clips));
else
while (i--) *buf1++ = SOX_SAMPLE_TO_UNSIGNED_16BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_S24: {
sox_int24_t * buf1 = (sox_int24_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_SIGNED_24BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_S24_3LE: {
unsigned char *buf1 = (unsigned char *)p->buf;
while (i--) {
uint32_t temp = (uint32_t)SOX_SAMPLE_TO_SIGNED_24BIT(*buf++, ft->clips);
*buf1++ = (temp & 0x000000FF);
*buf1++ = (temp & 0x0000FF00) >> 8;
*buf1++ = (temp & 0x00FF0000) >> 16;
}
break;
}
case SND_PCM_FORMAT_U24: {
sox_uint24_t * buf1 = (sox_uint24_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_UNSIGNED_24BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_S32: {
int32_t * buf1 = (int32_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_SIGNED_32BIT(*buf++, ft->clips);
break;
}
case SND_PCM_FORMAT_U32: {
uint32_t * buf1 = (uint32_t *)p->buf;
while (i--) *buf1++ = SOX_SAMPLE_TO_UNSIGNED_32BIT(*buf++, ft->clips);
break;
}
default: lsx_fail_errno(ft, SOX_EFMT, "invalid format");
return 0;
}
for (i = 0; i < n; i += actual * ft->signal.channels) do {
actual = snd_pcm_writei(p->pcm,
p->buf + i * formats[p->format].bytes,
(n - i) / ft->signal.channels);
if (errno == EAGAIN) /* Happens naturally; don't report it: */
errno = 0;
if (actual < 0 && recover(ft, p->pcm, (int)actual) < 0)
return 0;
} while (actual < 0);
}
return len;
}
static int stop(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
snd_pcm_close(p->pcm);
free(p->buf);
return SOX_SUCCESS;
}
static int stop_write(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
size_t n = ft->signal.channels * p->period, npad = n - (ft->olength % n);
sox_sample_t * buf = lsx_calloc(npad, sizeof(*buf)); /* silent samples */
if (npad != n) /* pad to hardware period: */
write_(ft, buf, npad);
free(buf);
snd_pcm_drain(p->pcm);
return stop(ft);
}
LSX_FORMAT_HANDLER(alsa)
{
static char const * const names[] = {"alsa", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2 , 32, 24, 16, 8, 0,
SOX_ENCODING_UNSIGNED, 32, 24, 16, 8, 0,
0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Advanced Linux Sound Architecture device driver",
names, SOX_FILE_DEVICE | SOX_FILE_NOSTDIO,
setup, read_, stop, setup, write_, stop_write,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

90
app/src/main/cpp/sox/amr-nb.c
Unescape View File

@ -0,0 +1,90 @@
/* File format: AMR-NB (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* In order to use the AMR format with SoX, you need to have an
* AMR library installed at SoX build time. The SoX build system
* recognizes the AMR implementations available from
* http://opencore-amr.sourceforge.net/
*/
#include "sox_i.h"
/* Common definitions: */
enum amrnb_mode { amrnb_mode_dummy };
static const unsigned amrnb_block_size[] = {13, 14, 16, 18, 20, 21, 27, 32, 6, 0, 0, 0, 0, 0, 0, 1};
static char const amrnb_magic[] = "#!AMR\n";
#define amr_block_size amrnb_block_size
#define amr_magic amrnb_magic
#define amr_priv_t amrnb_priv_t
#define amr_opencore_funcs amrnb_opencore_funcs
#define amr_gp3_funcs amrnb_gp3_funcs
#define AMR_CODED_MAX 32 /* max coded size */
#define AMR_ENCODING SOX_ENCODING_AMR_NB
#define AMR_FORMAT_FN lsx_amr_nb_format_fn
#define AMR_FRAME 160 /* 20ms @ 8kHz */
#define AMR_MODE_MAX 7
#define AMR_NAMES "amr-nb", "anb"
#define AMR_RATE 8000
#define AMR_DESC "3GPP Adaptive Multi Rate Narrow-Band (AMR-NB) lossy speech compressor"
#ifdef DL_OPENCORE_AMRNB
#define AMR_FUNC LSX_DLENTRY_DYNAMIC
#else
#define AMR_FUNC LSX_DLENTRY_STATIC
#endif /* DL_AMRNB */
/* OpenCore definitions: */
#define AMR_OPENCORE 1
#define AMR_OPENCORE_ENABLE_ENCODE 1
#define AMR_OPENCORE_FUNC_ENTRIES(f,x) \
AMR_FUNC(f,x, void*, Encoder_Interface_init, (int dtx)) \
AMR_FUNC(f,x, int, Encoder_Interface_Encode, (void* state, enum amrnb_mode mode, const short* in, unsigned char* out, int forceSpeech)) \
AMR_FUNC(f,x, void, Encoder_Interface_exit, (void* state)) \
AMR_FUNC(f,x, void*, Decoder_Interface_init, (void)) \
AMR_FUNC(f,x, void, Decoder_Interface_Decode, (void* state, const unsigned char* in, short* out, int bfi)) \
AMR_FUNC(f,x, void, Decoder_Interface_exit, (void* state)) \
#define AmrEncoderInit() \
Encoder_Interface_init(1)
#define AmrEncoderEncode(state, mode, in, out, forceSpeech) \
Encoder_Interface_Encode(state, mode, in, out, forceSpeech)
#define AmrEncoderExit(state) \
Encoder_Interface_exit(state)
#define AmrDecoderInit() \
Decoder_Interface_init()
#define AmrDecoderDecode(state, in, out, bfi) \
Decoder_Interface_Decode(state, in, out, bfi)
#define AmrDecoderExit(state) \
Decoder_Interface_exit(state)
#define AMR_OPENCORE_DESC "amr-nb OpenCore library"
static const char* const amr_opencore_library_names[] =
{
#ifdef DL_OPENCORE_AMRNB
"libopencore-amrnb",
"libopencore-amrnb-0",
#endif
NULL
};
#include "amr.h"

115
app/src/main/cpp/sox/amr-wb.c
Unescape View File

@ -0,0 +1,115 @@
/* File format: AMR-WB (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* In order to use the AMR format with SoX, you need to have an
* AMR library installed at SoX build time. The SoX build system
* recognizes the AMR implementations available from
* http://opencore-amr.sourceforge.net/
*/
#include "sox_i.h"
/* Common definitions: */
static const uint8_t amrwb_block_size[] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
static char const amrwb_magic[] = "#!AMR-WB\n";
#define amr_block_size amrwb_block_size
#define amr_magic amrwb_magic
#define amr_priv_t amrwb_priv_t
#define amr_opencore_funcs amrwb_opencore_funcs
#define amr_vo_funcs amrwb_vo_funcs
#define AMR_CODED_MAX 61 /* NB_SERIAL_MAX */
#define AMR_ENCODING SOX_ENCODING_AMR_WB
#define AMR_FORMAT_FN lsx_amr_wb_format_fn
#define AMR_FRAME 320 /* L_FRAME16k */
#define AMR_MODE_MAX 8
#define AMR_NAMES "amr-wb", "awb"
#define AMR_RATE 16000
#define AMR_DESC "3GPP Adaptive Multi Rate Wide-Band (AMR-WB) lossy speech compressor"
/* OpenCore definitions: */
#ifdef DL_OPENCORE_AMRWB
#define AMR_OC_FUNC LSX_DLENTRY_DYNAMIC
#else
#define AMR_OC_FUNC LSX_DLENTRY_STATIC
#endif
#if defined(HAVE_OPENCORE_AMRWB_DEC_IF_H) || defined(DL_OPENCORE_AMRWB)
#define AMR_OPENCORE 1
#define AMR_OPENCORE_ENABLE_ENCODE 0
#endif
#define AMR_OPENCORE_FUNC_ENTRIES(f,x) \
AMR_OC_FUNC(f,x, void*, D_IF_init, (void)) \
AMR_OC_FUNC(f,x, void, D_IF_decode, (void* state, const unsigned char* in, short* out, int bfi)) \
AMR_OC_FUNC(f,x, void, D_IF_exit, (void* state)) \
#define AmrDecoderInit() \
D_IF_init()
#define AmrDecoderDecode(state, in, out, bfi) \
D_IF_decode(state, in, out, bfi)
#define AmrDecoderExit(state) \
D_IF_exit(state)
#define AMR_OPENCORE_DESC "amr-wb OpenCore library"
static const char* const amr_opencore_library_names[] =
{
#ifdef DL_OPENCORE_AMRWB
"libopencore-amrwb",
"libopencore-amrwb-0",
#endif
NULL
};
/* VO definitions: */
#ifdef DL_VO_AMRWBENC
#define AMR_VO_FUNC LSX_DLENTRY_DYNAMIC
#else
#define AMR_VO_FUNC LSX_DLENTRY_STATIC
#endif
#if defined(HAVE_VO_AMRWBENC_ENC_IF_H) || defined(DL_VO_AMRWBENC)
#define AMR_VO 1
#endif
#define AMR_VO_FUNC_ENTRIES(f,x) \
AMR_VO_FUNC(f,x, void*, E_IF_init, (void)) \
AMR_VO_FUNC(f,x, int, E_IF_encode,(void* state, int16_t mode, int16_t* in, uint8_t* out, int16_t dtx)) \
AMR_VO_FUNC(f,x, void, E_IF_exit, (void* state)) \
#define AmrEncoderInit() \
E_IF_init()
#define AmrEncoderEncode(state, mode, in, out, forceSpeech) \
E_IF_encode(state, mode, in, out, forceSpeech)
#define AmrEncoderExit(state) \
E_IF_exit(state)
#define AMR_VO_DESC "amr-wb VisualOn library"
static const char* const amr_vo_library_names[] =
{
#ifdef DL_VO_AMRWBENC
"libvo-amrwbenc",
"libvo-amrwbenc-0",
#endif
NULL
};
#include "amr.h"

335
app/src/main/cpp/sox/amr.h
Unescape View File

@ -0,0 +1,335 @@
/* File format: AMR (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <string.h>
#include <math.h>
#ifdef AMR_OPENCORE
LSX_DLENTRIES_TO_FUNCTIONS(AMR_OPENCORE_FUNC_ENTRIES)
typedef struct amr_opencore_funcs {
LSX_DLENTRIES_TO_PTRS(AMR_OPENCORE_FUNC_ENTRIES, amr_dl);
} amr_opencore_funcs;
#endif /* AMR_OPENCORE */
#ifdef AMR_VO
LSX_DLENTRIES_TO_FUNCTIONS(AMR_VO_FUNC_ENTRIES)
typedef struct amr_vo_funcs {
LSX_DLENTRIES_TO_PTRS(AMR_VO_FUNC_ENTRIES, amr_dl);
} amr_vo_funcs;
#endif /* AMR_VO */
#define AMR_CALL(p, func, args) ((p)->opencore.func args)
#ifdef AMR_VO
#define AMR_CALL_ENCODER(p, func, args) ((p)->vo.func args)
#else
#define AMR_CALL_ENCODER(p, func, args) ((p)->opencore.func args)
#endif
typedef struct amr_priv_t {
void* state;
unsigned mode;
size_t pcm_index;
#ifdef AMR_OPENCORE
amr_opencore_funcs opencore;
#endif /* AMR_OPENCORE */
#ifdef AMR_VO
amr_vo_funcs vo;
#endif /* AMR_VO */
short pcm[AMR_FRAME];
} priv_t;
#ifdef AMR_OPENCORE
static size_t decode_1_frame(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
size_t n;
uint8_t coded[AMR_CODED_MAX];
if (lsx_readbuf(ft, &coded[0], (size_t)1) != 1)
return AMR_FRAME;
n = amr_block_size[(coded[0] >> 3) & 0x0F];
if (!n) {
lsx_fail("invalid block type");
return AMR_FRAME;
}
n--;
if (lsx_readbuf(ft, &coded[1], n) != n)
return AMR_FRAME;
AMR_CALL(p, AmrDecoderDecode, (p->state, coded, p->pcm, 0));
return 0;
}
#endif
static int openlibrary(priv_t* p, int encoding)
{
int open_library_result;
(void)encoding;
#ifdef AMR_OPENCORE
if (AMR_OPENCORE_ENABLE_ENCODE || !encoding)
{
LSX_DLLIBRARY_TRYOPEN(
0,
&p->opencore,
amr_dl,
AMR_OPENCORE_FUNC_ENTRIES,
AMR_OPENCORE_DESC,
amr_opencore_library_names,
open_library_result);
if (!open_library_result)
return SOX_SUCCESS;
lsx_fail("Unable to open " AMR_OPENCORE_DESC);
return SOX_EOF;
}
#endif /* AMR_OPENCORE */
#ifdef AMR_VO
if (encoding) {
LSX_DLLIBRARY_TRYOPEN(
0,
&p->vo,
amr_dl,
AMR_VO_FUNC_ENTRIES,
AMR_VO_DESC,
amr_vo_library_names,
open_library_result);
if (!open_library_result)
return SOX_SUCCESS;
lsx_fail("Unable to open " AMR_VO_DESC);
}
#endif /* AMR_VO */
return SOX_EOF;
}
static void closelibrary(priv_t* p)
{
#ifdef AMR_OPENCORE
LSX_DLLIBRARY_CLOSE(&p->opencore, amr_dl);
#endif
#ifdef AMR_VO
LSX_DLLIBRARY_CLOSE(&p->vo, amr_dl);
#endif
}
#ifdef AMR_OPENCORE
static size_t amr_duration_frames(sox_format_t * ft)
{
off_t frame_size, data_start_offset = lsx_tell(ft);
size_t frames;
uint8_t coded;
for (frames = 0; lsx_readbuf(ft, &coded, (size_t)1) == 1; ++frames) {
frame_size = amr_block_size[coded >> 3 & 15];
if (!frame_size) {
lsx_fail("invalid block type");
break;
}
if (lsx_seeki(ft, frame_size - 1, SEEK_CUR)) {
lsx_fail("seek");
break;
}
}
lsx_debug("frames=%lu", (unsigned long)frames);
lsx_seeki(ft, data_start_offset, SEEK_SET);
return frames;
}
#endif
static int startread(sox_format_t * ft)
{
#if !defined(AMR_OPENCORE)
lsx_fail_errno(ft, SOX_EOF, "SoX was compiled without AMR-WB decoding support.");
return SOX_EOF;
#else
priv_t * p = (priv_t *)ft->priv;
char buffer[sizeof(amr_magic) - 1];
int open_library_result;
if (lsx_readchars(ft, buffer, sizeof(buffer)))
return SOX_EOF;
if (memcmp(buffer, amr_magic, sizeof(buffer))) {
lsx_fail_errno(ft, SOX_EHDR, "invalid magic number");
return SOX_EOF;
}
open_library_result = openlibrary(p, 0);
if (open_library_result != SOX_SUCCESS)
return open_library_result;
p->pcm_index = AMR_FRAME;
p->state = AMR_CALL(p, AmrDecoderInit, ());
if (!p->state)
{
closelibrary(p);
lsx_fail("AMR decoder failed to initialize.");
return SOX_EOF;
}
ft->signal.rate = AMR_RATE;
ft->encoding.encoding = AMR_ENCODING;
ft->signal.channels = 1;
ft->signal.length = ft->signal.length != SOX_IGNORE_LENGTH && ft->seekable?
(size_t)(amr_duration_frames(ft) * .02 * ft->signal.rate +.5) : SOX_UNSPEC;
return SOX_SUCCESS;
#endif
}
#ifdef AMR_OPENCORE
static size_t read_samples(sox_format_t * ft, sox_sample_t * buf, size_t len)
{
priv_t * p = (priv_t *)ft->priv;
size_t done;
for (done = 0; done < len; done++) {
if (p->pcm_index >= AMR_FRAME)
p->pcm_index = decode_1_frame(ft);
if (p->pcm_index >= AMR_FRAME)
break;
*buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(p->pcm[p->pcm_index++], ft->clips);
}
return done;
}
static int stopread(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
AMR_CALL(p, AmrDecoderExit, (p->state));
closelibrary(p);
return SOX_SUCCESS;
}
#else
#define read_samples NULL
#define stopread NULL
#endif
static int startwrite(sox_format_t * ft)
{
#if !defined(AMR_VO) && !AMR_OPENCORE_ENABLE_ENCODE
lsx_fail_errno(ft, SOX_EOF, "SoX was compiled without AMR-WB encoding support.");
return SOX_EOF;
#else
priv_t * p = (priv_t *)ft->priv;
int open_library_result;
if (ft->encoding.compression != HUGE_VAL) {
p->mode = (unsigned)ft->encoding.compression;
if (p->mode != ft->encoding.compression || p->mode > AMR_MODE_MAX) {
lsx_fail_errno(ft, SOX_EINVAL, "compression level must be a whole number from 0 to %i", AMR_MODE_MAX);
return SOX_EOF;
}
}
else p->mode = 0;
open_library_result = openlibrary(p, 1);
if (open_library_result != SOX_SUCCESS)
return open_library_result;
p->state = AMR_CALL_ENCODER(p, AmrEncoderInit, ());
if (!p->state)
{
closelibrary(p);
lsx_fail("AMR encoder failed to initialize.");
return SOX_EOF;
}
lsx_writes(ft, amr_magic);
p->pcm_index = 0;
return SOX_SUCCESS;
#endif
}
#if defined(AMR_VO) || AMR_OPENCORE_ENABLE_ENCODE
static sox_bool encode_1_frame(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
uint8_t coded[AMR_CODED_MAX];
int n = AMR_CALL_ENCODER(p, AmrEncoderEncode, (p->state, p->mode, p->pcm, coded, 1));
sox_bool result = lsx_writebuf(ft, coded, (size_t) (size_t) (unsigned)n) == (unsigned)n;
if (!result)
lsx_fail_errno(ft, errno, "write error");
return result;
}
static size_t write_samples(sox_format_t * ft, const sox_sample_t * buf, size_t len)
{
priv_t * p = (priv_t *)ft->priv;
size_t done;
for (done = 0; done < len; ++done) {
SOX_SAMPLE_LOCALS;
p->pcm[p->pcm_index++] = SOX_SAMPLE_TO_SIGNED_16BIT(*buf++, ft->clips);
if (p->pcm_index == AMR_FRAME) {
p->pcm_index = 0;
if (!encode_1_frame(ft))
return 0;
}
}
return done;
}
static int stopwrite(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
int result = SOX_SUCCESS;
if (p->pcm_index) {
do {
p->pcm[p->pcm_index++] = 0;
} while (p->pcm_index < AMR_FRAME);
if (!encode_1_frame(ft))
result = SOX_EOF;
}
AMR_CALL_ENCODER(p, AmrEncoderExit, (p->state));
return result;
}
#else
#define write_samples NULL
#define stopwrite NULL
#endif /* defined(AMR_VO) || AMR_OPENCORE_ENABLE_ENCODE */
sox_format_handler_t const * AMR_FORMAT_FN(void);
sox_format_handler_t const * AMR_FORMAT_FN(void)
{
static char const * const names[] = {AMR_NAMES, NULL};
static sox_rate_t const write_rates[] = {AMR_RATE, 0};
static unsigned const write_encodings[] = {AMR_ENCODING, 0, 0};
static sox_format_handler_t handler = {
SOX_LIB_VERSION_CODE,
AMR_DESC,
names, SOX_FILE_MONO,
startread, read_samples, stopread,
startwrite, write_samples, stopwrite,
NULL, write_encodings, write_rates, sizeof(priv_t)
};
return &handler;
}

134
app/src/main/cpp/sox/ao.c
Unescape View File

@ -0,0 +1,134 @@
/* libao player support for sox
* (c) Reuben Thomas <rrt@sc3d.org> 2007
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <ao/ao.h>
typedef struct {
int driver_id;
ao_device *device;
ao_sample_format format;
char *buf;
size_t buf_size;
} priv_t;
static int startwrite(sox_format_t * ft)
{
priv_t * ao = (priv_t *)ft->priv;
ao->buf_size = sox_globals.bufsiz - (sox_globals.bufsiz % (ft->encoding.bits_per_sample >> 3));
ao->buf_size *= (ft->encoding.bits_per_sample >> 3);
ao->buf = lsx_malloc(ao->buf_size);
if (!ao->buf)
{
lsx_fail_errno(ft, SOX_ENOMEM, "Can not allocate memory for ao driver");
return SOX_EOF;
}
ao_initialize();
if (strcmp(ft->filename,"default") == 0)
{
if ((ao->driver_id = ao_default_driver_id()) < 0) {
lsx_fail("Could not find a default ao driver");
return SOX_EOF;
}
}
else
{
if ((ao->driver_id = ao_driver_id(ft->filename)) < 0) {
lsx_fail("Could not find a ao driver %s", ft->filename);
return SOX_EOF;
}
}
ao->format.bits = ft->encoding.bits_per_sample;
ao->format.rate = ft->signal.rate;
ao->format.channels = ft->signal.channels;
ao->format.byte_format = AO_FMT_NATIVE;
if ((ao->device = ao_open_live(ao->driver_id, &ao->format, NULL)) == NULL) {
lsx_fail("Could not open device: error %d", errno);
return SOX_EOF;
}
return SOX_SUCCESS;
}
static void sox_sw_write_buf(char *buf1, sox_sample_t const * buf2, size_t len, sox_bool swap, sox_uint64_t * clips)
{
while (len--)
{
SOX_SAMPLE_LOCALS;
uint16_t datum = SOX_SAMPLE_TO_SIGNED_16BIT(*buf2++, *clips);
if (swap)
datum = lsx_swapw(datum);
*(uint16_t *)buf1 = datum;
buf1++; buf1++;
}
}
static size_t write_samples(sox_format_t *ft, const sox_sample_t *buf, size_t len)
{
priv_t * ao = (priv_t *)ft->priv;
uint_32 aobuf_size;
if (len > ao->buf_size / (ft->encoding.bits_per_sample >> 3))
len = ao->buf_size / (ft->encoding.bits_per_sample >> 3);
aobuf_size = (ft->encoding.bits_per_sample >> 3) * len;
sox_sw_write_buf(ao->buf, buf, len, ft->encoding.reverse_bytes,
&(ft->clips));
if (ao_play(ao->device, (void *)ao->buf, aobuf_size) == 0)
return 0;
return len;
}
static int stopwrite(sox_format_t * ft)
{
priv_t * ao = (priv_t *)ft->priv;
free(ao->buf);
if (ao_close(ao->device) == 0) {
lsx_fail("Error closing libao output");
return SOX_EOF;
}
ao_shutdown();
return SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(ao)
{
static char const * const names[] = {"ao", NULL};
static unsigned const encodings[] = {SOX_ENCODING_SIGN2, 16, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Xiph's libao device driver", names, SOX_FILE_DEVICE | SOX_FILE_NOSTDIO,
NULL, NULL, NULL,
startwrite, write_samples, stopwrite,
NULL, encodings, NULL, sizeof(priv_t)
};
return &handler;
}

237
app/src/main/cpp/sox/au.c
Unescape View File

@ -0,0 +1,237 @@
/* libSoX Sun format with header (SunOS 4.1; see /usr/demo/SOUND).
* Copyright 1991, 1992, 1993 Guido van Rossum And Sundry Contributors.
*
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Guido van Rossum And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* October 7, 1998 - cbagwell@sprynet.com
* G.723 was using incorrect # of bits. Corrected to 3 and 5 bits.
*
* NeXT uses this format also, but has more format codes defined.
* DEC uses a slight variation and swaps bytes.
* We support only the common formats, plus
* CCITT G.721 (32 kbit/s) and G.723 (24/40 kbit/s),
* courtesy of Sun's public domain implementation.
*/
#include "sox_i.h"
#include "g72x.h"
#include <string.h>
/* Magic numbers used in Sun and NeXT audio files */
static struct {char str[4]; sox_bool reverse_bytes; char const * desc;} id[] = {
{"\x2e\x73\x6e\x64", MACHINE_IS_LITTLEENDIAN, "big-endian `.snd'"},
{"\x64\x6e\x73\x2e", MACHINE_IS_BIGENDIAN , "little-endian `.snd'"},
{"\x00\x64\x73\x2e", MACHINE_IS_BIGENDIAN , "little-endian `\\0ds.' (for DEC)"},
{"\x2e\x73\x64\x00", MACHINE_IS_LITTLEENDIAN, "big-endian `\\0ds.'"},
{" ", 0, NULL}
};
#define FIXED_HDR 24
#define SUN_UNSPEC ~0u /* Unspecified data size (this is legal) */
typedef enum {
Unspecified, Mulaw_8, Linear_8, Linear_16, Linear_24, Linear_32, Float,
Double, Indirect, Nested, Dsp_core, Dsp_data_8, Dsp_data_16, Dsp_data_24,
Dsp_data_32, Unknown, Display, Mulaw_squelch, Emphasized, Compressed,
Compressed_emphasized, Dsp_commands, Dsp_commands_samples, Adpcm_g721,
Adpcm_g722, Adpcm_g723_3, Adpcm_g723_5, Alaw_8, Unknown_other} ft_encoding_t;
static char const * const str[] = {
"Unspecified", "8-bit mu-law", "8-bit signed linear", "16-bit signed linear",
"24-bit signed linear", "32-bit signed linear", "Floating-point",
"Double precision float", "Fragmented sampled data", "Unknown", "DSP program",
"8-bit fixed-point", "16-bit fixed-point", "24-bit fixed-point",
"32-bit fixed-point", "Unknown", "Non-audio data", "Mu-law squelch",
"16-bit linear with emphasis", "16-bit linear with compression",
"16-bit linear with emphasis and compression", "Music Kit DSP commands",
"Music Kit DSP samples", "4-bit G.721 ADPCM", "G.722 ADPCM",
"3-bit G.723 ADPCM", "5-bit G.723 ADPCM", "8-bit a-law", "Unknown"};
static ft_encoding_t ft_enc(unsigned size, sox_encoding_t encoding)
{
if (encoding == SOX_ENCODING_ULAW && size == 8) return Mulaw_8;
if (encoding == SOX_ENCODING_ALAW && size == 8) return Alaw_8;
if (encoding == SOX_ENCODING_SIGN2 && size == 8) return Linear_8;
if (encoding == SOX_ENCODING_SIGN2 && size == 16) return Linear_16;
if (encoding == SOX_ENCODING_SIGN2 && size == 24) return Linear_24;
if (encoding == SOX_ENCODING_SIGN2 && size == 32) return Linear_32;
if (encoding == SOX_ENCODING_FLOAT && size == 32) return Float;
if (encoding == SOX_ENCODING_FLOAT && size == 64) return Double;
return Unspecified;
}
static sox_encoding_t sox_enc(uint32_t ft_encoding, unsigned * size)
{
switch (ft_encoding) {
case Mulaw_8 : *size = 8; return SOX_ENCODING_ULAW;
case Alaw_8 : *size = 8; return SOX_ENCODING_ALAW;
case Linear_8 : *size = 8; return SOX_ENCODING_SIGN2;
case Linear_16 : *size = 16; return SOX_ENCODING_SIGN2;
case Linear_24 : *size = 24; return SOX_ENCODING_SIGN2;
case Linear_32 : *size = 32; return SOX_ENCODING_SIGN2;
case Float : *size = 32; return SOX_ENCODING_FLOAT;
case Double : *size = 64; return SOX_ENCODING_FLOAT;
case Adpcm_g721 : *size = 4; return SOX_ENCODING_G721; /* read-only */
case Adpcm_g723_3: *size = 3; return SOX_ENCODING_G723; /* read-only */
case Adpcm_g723_5: *size = 5; return SOX_ENCODING_G723; /* read-only */
default: return SOX_ENCODING_UNKNOWN;
}
}
typedef struct { /* For G72x decoding: */
struct g72x_state state;
int (*dec_routine)(int i, int out_coding, struct g72x_state *state_ptr);
unsigned int in_buffer;
int in_bits;
} priv_t;
/*
* Unpack input codes and pass them back as bytes.
* Returns 1 if there is residual input, returns -1 if eof, else returns 0.
* (Adapted from Sun's decode.c.)
*/
static int unpack_input(sox_format_t * ft, unsigned char *code)
{
priv_t * p = (priv_t *) ft->priv;
unsigned char in_byte;
if (p->in_bits < (int)ft->encoding.bits_per_sample) {
if (lsx_read_b_buf(ft, &in_byte, (size_t) 1) != 1) {
*code = 0;
return -1;
}
p->in_buffer |= (in_byte << p->in_bits);
p->in_bits += 8;
}
*code = p->in_buffer & ((1 << ft->encoding.bits_per_sample) - 1);
p->in_buffer >>= ft->encoding.bits_per_sample;
p->in_bits -= ft->encoding.bits_per_sample;
return p->in_bits > 0;
}
static size_t dec_read(sox_format_t *ft, sox_sample_t *buf, size_t samp)
{
priv_t * p = (priv_t *)ft->priv;
unsigned char code;
size_t done;
for (done = 0; samp > 0 && unpack_input(ft, &code) >= 0; ++done, --samp)
*buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(
(*p->dec_routine)(code, AUDIO_ENCODING_LINEAR, &p->state),);
return done;
}
static int startread(sox_format_t * ft)
{
priv_t * p = (priv_t *) ft->priv;
char magic[4]; /* These 6 variables represent a Sun sound */
uint32_t hdr_size; /* header on disk. The uint32_t are written as */
uint32_t data_size; /* big-endians. At least extra bytes (totalling */
uint32_t ft_encoding; /* hdr_size - FIXED_HDR) are an "info" field of */
uint32_t rate; /* unspecified nature, usually a string. By */
uint32_t channels; /* convention the header size is a multiple of 4. */
unsigned i, bits_per_sample;
sox_encoding_t encoding;
if (lsx_readchars(ft, magic, sizeof(magic)))
return SOX_EOF;
for (i = 0; id[i].desc && memcmp(magic, id[i].str, sizeof(magic)); ++i);
if (!id[i].desc) {
lsx_fail_errno(ft, SOX_EHDR, "au: can't find Sun/NeXT/DEC identifier");
return SOX_EOF;
}
lsx_report("found %s identifier", id[i].desc);
ft->encoding.reverse_bytes = id[i].reverse_bytes;
if (lsx_readdw(ft, &hdr_size) ||
lsx_readdw(ft, &data_size) || /* Can be SUN_UNSPEC */
lsx_readdw(ft, &ft_encoding) ||
lsx_readdw(ft, &rate) ||
lsx_readdw(ft, &channels))
return SOX_EOF;
if (hdr_size < FIXED_HDR) {
lsx_fail_errno(ft, SOX_EHDR, "header size %u is too small", hdr_size);
return SOX_EOF;
}
if (hdr_size < FIXED_HDR + 4)
lsx_warn("header size %u is too small", hdr_size);
if (!(encoding = sox_enc(ft_encoding, &bits_per_sample))) {
int n = min(ft_encoding, Unknown_other);
lsx_fail_errno(ft, SOX_EFMT, "unsupported encoding `%s' (%#x)", str[n], ft_encoding);
return SOX_EOF;
}
switch (ft_encoding) {
case Adpcm_g721 : p->dec_routine = g721_decoder ; break;
case Adpcm_g723_3: p->dec_routine = g723_24_decoder; break;
case Adpcm_g723_5: p->dec_routine = g723_40_decoder; break;
}
if (p->dec_routine) {
g72x_init_state(&p->state);
ft->handler.seek = NULL;
ft->handler.read = dec_read;
}
if (hdr_size > FIXED_HDR) {
size_t info_size = hdr_size - FIXED_HDR;
char * buf = lsx_calloc(1, info_size + 1); /* +1 ensures null-terminated */
if (lsx_readchars(ft, buf, info_size) != SOX_SUCCESS) {
free(buf);
return SOX_EOF;
}
sox_append_comments(&ft->oob.comments, buf);
free(buf);
}
if (data_size == SUN_UNSPEC)
data_size = SOX_UNSPEC;
return lsx_check_read_params(ft, channels, (sox_rate_t)rate, encoding,
bits_per_sample, div_bits(data_size, bits_per_sample), sox_true);
}
static int write_header(sox_format_t * ft)
{
char * comment = lsx_cat_comments(ft->oob.comments);
size_t len = strlen(comment) + 1; /* Write out null-terminated */
size_t info_len = max(4, (len + 3) & ~3u); /* Minimum & multiple of 4 bytes */
int i = ft->encoding.reverse_bytes == MACHINE_IS_BIGENDIAN? 2 : 0;
uint64_t size64 = ft->olength ? ft->olength : ft->signal.length;
unsigned size = size64 == SOX_UNSPEC
? SUN_UNSPEC
: size64*(ft->encoding.bits_per_sample >> 3) > UINT_MAX
? SUN_UNSPEC
: (unsigned)(size64*(ft->encoding.bits_per_sample >> 3));
sox_bool error = sox_false
||lsx_writechars(ft, id[i].str, sizeof(id[i].str))
||lsx_writedw(ft, FIXED_HDR + (unsigned)info_len)
||lsx_writedw(ft, size)
||lsx_writedw(ft, ft_enc(ft->encoding.bits_per_sample, ft->encoding.encoding))
||lsx_writedw(ft, (unsigned)(ft->signal.rate + .5))
||lsx_writedw(ft, ft->signal.channels)
||lsx_writechars(ft, comment, len)
||lsx_padbytes(ft, info_len - len);
free(comment);
return error? SOX_EOF: SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(au)
{
static char const * const names[] = {"au", "snd", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_ULAW, 8, 0,
SOX_ENCODING_ALAW, 8, 0,
SOX_ENCODING_SIGN2, 8, 16, 24, 32, 0,
SOX_ENCODING_FLOAT, 32, 64, 0,
0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"PCM file format used widely on Sun systems",
names, SOX_FILE_BIG_END | SOX_FILE_REWIND,
startread, lsx_rawread, NULL,
write_header, lsx_rawwrite, NULL,
lsx_rawseek, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

287
app/src/main/cpp/sox/avr.c
Unescape View File

@ -0,0 +1,287 @@
/* AVR file format handler for SoX
* Copyright (C) 1999 Jan Paul Schmidt <jps@fundament.org>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <stdio.h>
#include <string.h>
#define AVR_MAGIC "2BIT"
/* Taken from the Audio File Formats FAQ */
typedef struct {
char magic [5]; /* 2BIT */
char name [8]; /* null-padded sample name */
unsigned short mono; /* 0 = mono, 0xffff = stereo */
unsigned short rez; /* 8 = 8 bit, 16 = 16 bit */
unsigned short sign; /* 0 = unsigned, 0xffff = signed */
unsigned short loop; /* 0 = no loop, 0xffff = looping sample */
unsigned short midi; /* 0xffff = no MIDI note assigned,
0xffXX = single key note assignment
0xLLHH = key split, low/hi note */
uint32_t rate; /* sample frequency in hertz */
uint32_t size; /* sample length in bytes or words (see rez) */
uint32_t lbeg; /* offset to start of loop in bytes or words.
set to zero if unused. */
uint32_t lend; /* offset to end of loop in bytes or words.
set to sample length if unused. */
unsigned short res1; /* Reserved, MIDI keyboard split */
unsigned short res2; /* Reserved, sample compression */
unsigned short res3; /* Reserved */
char ext[20]; /* Additional filename space, used
if (name[7] != 0) */
char user[64]; /* User defined. Typically ASCII message. */
} priv_t;
/*
* Do anything required before you start reading samples.
* Read file header.
* Find out sampling rate,
* size and encoding of samples,
* mono/stereo/quad.
*/
static int startread(sox_format_t * ft)
{
priv_t * avr = (priv_t *)ft->priv;
int rc;
lsx_reads(ft, avr->magic, (size_t)4);
if (strncmp (avr->magic, AVR_MAGIC, (size_t)4)) {
lsx_fail_errno(ft,SOX_EHDR,"AVR: unknown header");
return(SOX_EOF);
}
lsx_readbuf(ft, avr->name, sizeof(avr->name));
lsx_readw (ft, &(avr->mono));
if (avr->mono) {
ft->signal.channels = 2;
}
else {
ft->signal.channels = 1;
}
lsx_readw (ft, &(avr->rez));
if (avr->rez == 8) {
ft->encoding.bits_per_sample = 8;
}
else if (avr->rez == 16) {
ft->encoding.bits_per_sample = 16;
}
else {
lsx_fail_errno(ft,SOX_EFMT,"AVR: unsupported sample resolution");
return(SOX_EOF);
}
lsx_readw (ft, &(avr->sign));
if (avr->sign) {
ft->encoding.encoding = SOX_ENCODING_SIGN2;
}
else {
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
}
lsx_readw (ft, &(avr->loop));
lsx_readw (ft, &(avr->midi));
lsx_readdw (ft, &(avr->rate));
/*
* No support for AVRs created by ST-Replay,
* Replay Proffesional and PRO-Series 12.
*
* Just masking the upper byte out.
*/
ft->signal.rate = (avr->rate & 0x00ffffff);
lsx_readdw (ft, &(avr->size));
lsx_readdw (ft, &(avr->lbeg));
lsx_readdw (ft, &(avr->lend));
lsx_readw (ft, &(avr->res1));
lsx_readw (ft, &(avr->res2));
lsx_readw (ft, &(avr->res3));
lsx_readbuf(ft, avr->ext, sizeof(avr->ext));
lsx_readbuf(ft, avr->user, sizeof(avr->user));
rc = lsx_rawstartread (ft);
if (rc)
return rc;
return(SOX_SUCCESS);
}
static int startwrite(sox_format_t * ft)
{
priv_t * avr = (priv_t *)ft->priv;
int rc;
if (!ft->seekable) {
lsx_fail_errno(ft,SOX_EOF,"AVR: file is not seekable");
return(SOX_EOF);
}
rc = lsx_rawstartwrite (ft);
if (rc)
return rc;
/* magic */
lsx_writes(ft, AVR_MAGIC);
/* name */
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
lsx_writeb(ft, 0);
/* mono */
if (ft->signal.channels == 1) {
lsx_writew (ft, 0);
}
else if (ft->signal.channels == 2) {
lsx_writew (ft, 0xffff);
}
else {
lsx_fail_errno(ft,SOX_EFMT,"AVR: number of channels not supported");
return(0);
}
/* rez */
if (ft->encoding.bits_per_sample == 8) {
lsx_writew (ft, 8);
}
else if (ft->encoding.bits_per_sample == 16) {
lsx_writew (ft, 16);
}
else {
lsx_fail_errno(ft,SOX_EFMT,"AVR: unsupported sample resolution");
return(SOX_EOF);
}
/* sign */
if (ft->encoding.encoding == SOX_ENCODING_SIGN2) {
lsx_writew (ft, 0xffff);
}
else if (ft->encoding.encoding == SOX_ENCODING_UNSIGNED) {
lsx_writew (ft, 0);
}
else {
lsx_fail_errno(ft,SOX_EFMT,"AVR: unsupported encoding");
return(SOX_EOF);
}
/* loop */
lsx_writew (ft, 0xffff);
/* midi */
lsx_writew (ft, 0xffff);
/* rate */
lsx_writedw(ft, (unsigned)(ft->signal.rate + .5));
/* size */
/* Don't know the size yet. */
lsx_writedw (ft, 0);
/* lbeg */
lsx_writedw (ft, 0);
/* lend */
/* Don't know the size yet, so we can't set lend, either. */
lsx_writedw (ft, 0);
/* res1 */
lsx_writew (ft, 0);
/* res2 */
lsx_writew (ft, 0);
/* res3 */
lsx_writew (ft, 0);
/* ext */
lsx_writebuf(ft, "\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0", sizeof(avr->ext));
/* user */
lsx_writebuf(ft,
"\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0"
"\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0"
"\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0"
"\0\0\0\0", sizeof (avr->user));
return(SOX_SUCCESS);
}
static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t nsamp)
{
priv_t * avr = (priv_t *)ft->priv;
avr->size += nsamp;
return (lsx_rawwrite (ft, buf, nsamp));
}
static int stopwrite(sox_format_t * ft)
{
priv_t * avr = (priv_t *)ft->priv;
unsigned size = avr->size / ft->signal.channels;
/* Fix size */
lsx_seeki(ft, (off_t)26, SEEK_SET);
lsx_writedw (ft, size);
/* Fix lend */
lsx_seeki(ft, (off_t)34, SEEK_SET);
lsx_writedw (ft, size);
return(SOX_SUCCESS);
}
LSX_FORMAT_HANDLER(avr)
{
static char const * const names[] = { "avr", NULL };
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2, 16, 8, 0,
SOX_ENCODING_UNSIGNED, 16, 8, 0,
0};
static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,
"Audio Visual Research format; used on the Mac",
names, SOX_FILE_BIG_END | SOX_FILE_MONO | SOX_FILE_STEREO,
startread, lsx_rawread, NULL,
startwrite, write_samples, stopwrite,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

47
app/src/main/cpp/sox/band.h
Unescape View File

@ -0,0 +1,47 @@
/* libSoX Bandpass effect file. July 5, 1991
* Copyright 1991 Lance Norskog And Sundry Contributors
*
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Lance Norskog And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* Algorithm: 2nd order recursive filter.
* Formula stolen from MUSIC56K, a toolkit of 56000 assembler stuff.
* Quote:
* This is a 2nd order recursive band pass filter of the form.
* y(n)= a * x(n) - b * y(n-1) - c * y(n-2)
* where :
* x(n) = "IN"
* "OUT" = y(n)
* c = EXP(-2*pi*cBW/S_RATE)
* b = -4*c/(1+c)*COS(2*pi*cCF/S_RATE)
* if cSCL=2 (i.e. noise input)
* a = SQT(((1+c)*(1+c)-b*b)*(1-c)/(1+c))
* else
* a = SQT(1-b*b/(4*c))*(1-c)
* endif
* note : cCF is the center frequency in Hertz
* cBW is the band width in Hertz
* cSCL is a scale factor, use 1 for pitched sounds
* use 2 for noise.
*
*
* July 1, 1999 - Jan Paul Schmidt <jps@fundament.org>
*
* This looks like the resonator band pass in SPKit. It's a
* second order all-pole (IIR) band-pass filter described
* at the pages 186 - 189 in
* Dodge, Charles & Jerse, Thomas A. 1985:
* Computer Music -- Synthesis, Composition and Performance.
* New York: Schirmer Books.
* Reference from the SPKit manual.
*/
p->a2 = exp(-2 * M_PI * bw_Hz / effp->in_signal.rate);
p->a1 = -4 * p->a2 / (1 + p->a2) * cos(2 * M_PI * p->fc / effp->in_signal.rate);
p->b0 = sqrt(1 - p->a1 * p->a1 / (4 * p->a2)) * (1 - p->a2);
if (p->filter_type == filter_BPF_SPK_N) {
mult = sqrt(((1+p->a2) * (1+p->a2) - p->a1*p->a1) * (1-p->a2) / (1+p->a2)) / p->b0;
p->b0 *= mult;
}

325
app/src/main/cpp/sox/bend.c
Unescape View File

@ -0,0 +1,325 @@
/* libSoX effect: Pitch Bend (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* Portions based on http://www.dspdimension.com/download smbPitchShift.cpp:
*
* COPYRIGHT 1999-2006 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*
* The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
*/
#ifdef NDEBUG /* Enable assert always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox_i.h"
#include <assert.h>
#define MAX_FRAME_LENGTH 8192
typedef struct {
unsigned nbends; /* Number of bends requested */
struct {
char *str; /* Command-line argument to parse for this bend */
uint64_t start; /* Start bending when in_pos equals this */
double cents;
uint64_t duration; /* Number of samples to bend */
} *bends;
unsigned frame_rate;
size_t in_pos; /* Number of samples read from the input stream */
unsigned bends_pos; /* Number of bends completed so far */
double shift;
float gInFIFO[MAX_FRAME_LENGTH];
float gOutFIFO[MAX_FRAME_LENGTH];
double gFFTworksp[2 * MAX_FRAME_LENGTH];
float gLastPhase[MAX_FRAME_LENGTH / 2 + 1];
float gSumPhase[MAX_FRAME_LENGTH / 2 + 1];
float gOutputAccum[2 * MAX_FRAME_LENGTH];
float gAnaFreq[MAX_FRAME_LENGTH];
float gAnaMagn[MAX_FRAME_LENGTH];
float gSynFreq[MAX_FRAME_LENGTH];
float gSynMagn[MAX_FRAME_LENGTH];
long gRover;
int fftFrameSize, ovsamp;
} priv_t;
static int parse(sox_effect_t * effp, char **argv, sox_rate_t rate)
{
priv_t *p = (priv_t *) effp->priv;
size_t i;
char const *next;
uint64_t last_seen = 0;
const uint64_t in_length = argv ? 0 :
(effp->in_signal.length != SOX_UNKNOWN_LEN ?
effp->in_signal.length / effp->in_signal.channels : SOX_UNKNOWN_LEN);
for (i = 0; i < p->nbends; ++i) {
if (argv) /* 1st parse only */
p->bends[i].str = lsx_strdup(argv[i]);
next = lsx_parseposition(rate, p->bends[i].str,
argv ? NULL : &p->bends[i].start, last_seen, in_length, '+');
last_seen = p->bends[i].start;
if (next == NULL || *next != ',')
break;
p->bends[i].cents = strtod(next + 1, (char **)&next);
if (p->bends[i].cents == 0 || *next != ',')
break;
next = lsx_parseposition(rate, next + 1,
argv ? NULL : &p->bends[i].duration, last_seen, in_length, '+');
last_seen = p->bends[i].duration;
if (next == NULL || *next != '\0')
break;
/* sanity checks */
if (!argv && p->bends[i].duration < p->bends[i].start) {
lsx_fail("Bend %" PRIuPTR " has negative width", i+1);
break;
}
if (!argv && i && p->bends[i].start < p->bends[i-1].start) {
lsx_fail("Bend %" PRIuPTR " overlaps with previous one", i+1);
break;
}
p->bends[i].duration -= p->bends[i].start;
}
if (i < p->nbends)
return lsx_usage(effp);
return SOX_SUCCESS;
}
static int create(sox_effect_t * effp, int argc, char **argv)
{
priv_t *p = (priv_t *) effp->priv;
char const * opts = "f:o:";
int c;
lsx_getopt_t optstate;
lsx_getopt_init(argc, argv, opts, NULL, lsx_getopt_flag_none, 1, &optstate);
p->frame_rate = 25;
p->ovsamp = 16;
while ((c = lsx_getopt(&optstate)) != -1) switch (c) {
GETOPT_NUMERIC(optstate, 'f', frame_rate, 10 , 80)
GETOPT_NUMERIC(optstate, 'o', ovsamp, 4 , 32)
default: lsx_fail("unknown option `-%c'", optstate.opt); return lsx_usage(effp);
}
argc -= optstate.ind, argv += optstate.ind;
p->nbends = argc;
p->bends = lsx_calloc(p->nbends, sizeof(*p->bends));
return parse(effp, argv, 0.); /* No rate yet; parse with dummy */
}
static int start(sox_effect_t * effp)
{
priv_t *p = (priv_t *) effp->priv;
unsigned i;
int n = effp->in_signal.rate / p->frame_rate + .5;
for (p->fftFrameSize = 2; n > 2; p->fftFrameSize <<= 1, n >>= 1);
assert(p->fftFrameSize <= MAX_FRAME_LENGTH);
p->shift = 1;
parse(effp, 0, effp->in_signal.rate); /* Re-parse now rate is known */
p->in_pos = p->bends_pos = 0;
for (i = 0; i < p->nbends; ++i)
if (p->bends[i].duration)
return SOX_SUCCESS;
return SOX_EFF_NULL;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t *p = (priv_t *) effp->priv;
size_t i, len = *isamp = *osamp = min(*isamp, *osamp);
double magn, phase, tmp, window, real, imag;
double freqPerBin, expct;
long k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
float pitchShift = p->shift;
/* set up some handy variables */
fftFrameSize2 = p->fftFrameSize / 2;
stepSize = p->fftFrameSize / p->ovsamp;
freqPerBin = effp->in_signal.rate / p->fftFrameSize;
expct = 2. * M_PI * (double) stepSize / (double) p->fftFrameSize;
inFifoLatency = p->fftFrameSize - stepSize;
if (!p->gRover)
p->gRover = inFifoLatency;
/* main processing loop */
for (i = 0; i < len; i++) {
SOX_SAMPLE_LOCALS;
++p->in_pos;
/* As long as we have not yet collected enough data just read in */
p->gInFIFO[p->gRover] = SOX_SAMPLE_TO_FLOAT_32BIT(ibuf[i], effp->clips);
obuf[i] = SOX_FLOAT_32BIT_TO_SAMPLE(
p->gOutFIFO[p->gRover - inFifoLatency], effp->clips);
p->gRover++;
/* now we have enough data for processing */
if (p->gRover >= p->fftFrameSize) {
if (p->bends_pos != p->nbends && p->in_pos >=
p->bends[p->bends_pos].start + p->bends[p->bends_pos].duration) {
pitchShift = p->shift *= pow(2., p->bends[p->bends_pos].cents / 1200);
++p->bends_pos;
}
if (p->bends_pos != p->nbends && p->in_pos >= p->bends[p->bends_pos].start) {
double progress = (double)(p->in_pos - p->bends[p->bends_pos].start) /
p->bends[p->bends_pos].duration;
progress = 1 - cos(M_PI * progress);
progress *= p->bends[p->bends_pos].cents * (.5 / 1200);
pitchShift = p->shift * pow(2., progress);
}
p->gRover = inFifoLatency;
/* do windowing and re,im interleave */
for (k = 0; k < p->fftFrameSize; k++) {
window = -.5 * cos(2 * M_PI * k / (double) p->fftFrameSize) + .5;
p->gFFTworksp[2 * k] = p->gInFIFO[k] * window;
p->gFFTworksp[2 * k + 1] = 0.;
}
/* ***************** ANALYSIS ******************* */
lsx_safe_cdft(2 * p->fftFrameSize, 1, p->gFFTworksp);
/* this is the analysis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = p->gFFTworksp[2 * k];
imag = - p->gFFTworksp[2 * k + 1];
/* compute magnitude and phase */
magn = 2. * sqrt(real * real + imag * imag);
phase = atan2(imag, real);
/* compute phase difference */
tmp = phase - p->gLastPhase[k];
p->gLastPhase[k] = phase;
tmp -= (double) k *expct; /* subtract expected phase difference */
/* map delta phase into +/- Pi interval */
qpd = tmp / M_PI;
if (qpd >= 0)
qpd += qpd & 1;
else qpd -= qpd & 1;
tmp -= M_PI * (double) qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = p->ovsamp * tmp / (2. * M_PI);
/* compute the k-th partials' true frequency */
tmp = (double) k *freqPerBin + tmp * freqPerBin;
/* store magnitude and true frequency in analysis arrays */
p->gAnaMagn[k] = magn;
p->gAnaFreq[k] = tmp;
}
/* this does the actual pitch shifting */
memset(p->gSynMagn, 0, p->fftFrameSize * sizeof(float));
memset(p->gSynFreq, 0, p->fftFrameSize * sizeof(float));
for (k = 0; k <= fftFrameSize2; k++) {
index = k * pitchShift;
if (index <= fftFrameSize2) {
p->gSynMagn[index] += p->gAnaMagn[k];
p->gSynFreq[index] = p->gAnaFreq[k] * pitchShift;
}
}
for (k = 0; k <= fftFrameSize2; k++) { /* SYNTHESIS */
/* get magnitude and true frequency from synthesis arrays */
magn = p->gSynMagn[k], tmp = p->gSynFreq[k];
tmp -= (double) k *freqPerBin; /* subtract bin mid frequency */
tmp /= freqPerBin; /* get bin deviation from freq deviation */
tmp = 2. * M_PI * tmp / p->ovsamp; /* take p->ovsamp into account */
tmp += (double) k *expct; /* add the overlap phase advance back in */
p->gSumPhase[k] += tmp; /* accumulate delta phase to get bin phase */
phase = p->gSumPhase[k];
/* get real and imag part and re-interleave */
p->gFFTworksp[2 * k] = magn * cos(phase);
p->gFFTworksp[2 * k + 1] = - magn * sin(phase);
}
for (k = p->fftFrameSize + 2; k < 2 * p->fftFrameSize; k++)
p->gFFTworksp[k] = 0.; /* zero negative frequencies */
lsx_safe_cdft(2 * p->fftFrameSize, -1, p->gFFTworksp);
/* do windowing and add to output accumulator */
for (k = 0; k < p->fftFrameSize; k++) {
window =
-.5 * cos(2. * M_PI * (double) k / (double) p->fftFrameSize) + .5;
p->gOutputAccum[k] +=
2. * window * p->gFFTworksp[2 * k] / (fftFrameSize2 * p->ovsamp);
}
for (k = 0; k < stepSize; k++)
p->gOutFIFO[k] = p->gOutputAccum[k];
memmove(p->gOutputAccum, /* shift accumulator */
p->gOutputAccum + stepSize, p->fftFrameSize * sizeof(float));
for (k = 0; k < inFifoLatency; k++) /* move input FIFO */
p->gInFIFO[k] = p->gInFIFO[k + stepSize];
}
}
return SOX_SUCCESS;
}
static int stop(sox_effect_t * effp)
{
priv_t *p = (priv_t *) effp->priv;
if (p->bends_pos != p->nbends)
lsx_warn("Input audio too short; bends not applied: %u",
p->nbends - p->bends_pos);
return SOX_SUCCESS;
}
static int lsx_kill(sox_effect_t * effp)
{
priv_t *p = (priv_t *) effp->priv;
unsigned i;
for (i = 0; i < p->nbends; ++i)
free(p->bends[i].str);
free(p->bends);
return SOX_SUCCESS;
}
sox_effect_handler_t const *lsx_bend_effect_fn(void)
{
static sox_effect_handler_t handler = {
"bend", "[-f frame-rate(25)] [-o over-sample(16)] {start,cents,end}",
0, create, start, flow, 0, stop, lsx_kill, sizeof(priv_t)
};
return &handler;
}

178
app/src/main/cpp/sox/biquad.c
Unescape View File

@ -0,0 +1,178 @@
/* libSoX Biquad filter common functions (c) 2006-7 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "biquad.h"
#include <string.h>
typedef biquad_t priv_t;
static char const * const width_str[] = {
"band-width(Hz)",
"band-width(kHz)",
"band-width(Hz, no warp)", /* deprecated */
"band-width(octaves)",
"Q",
"slope",
};
static char const all_width_types[] = "hkboqs";
int lsx_biquad_getopts(sox_effect_t * effp, int argc, char **argv,
int min_args, int max_args, int fc_pos, int width_pos, int gain_pos,
char const * allowed_width_types, filter_t filter_type)
{
priv_t * p = (priv_t *)effp->priv;
char width_type = *allowed_width_types;
char dummy, * dummy_p; /* To check for extraneous chars. */
--argc, ++argv;
p->filter_type = filter_type;
if (argc < min_args || argc > max_args ||
(argc > fc_pos && ((p->fc = lsx_parse_frequency(argv[fc_pos], &dummy_p)) <= 0 || *dummy_p)) ||
(argc > width_pos && ((unsigned)(sscanf(argv[width_pos], "%lf%c %c", &p->width, &width_type, &dummy)-1) > 1 || p->width <= 0)) ||
(argc > gain_pos && sscanf(argv[gain_pos], "%lf %c", &p->gain, &dummy) != 1) ||
!strchr(allowed_width_types, width_type) || (width_type == 's' && p->width > 1))
return lsx_usage(effp);
p->width_type = strchr(all_width_types, width_type) - all_width_types;
if ((size_t)p->width_type >= strlen(all_width_types))
p->width_type = 0;
if (p->width_type == width_bw_kHz) {
p->width *= 1000;
p->width_type = width_bw_Hz;
}
return SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
/* Simplify: */
p->b2 /= p->a0;
p->b1 /= p->a0;
p->b0 /= p->a0;
p->a2 /= p->a0;
p->a1 /= p->a0;
p->o2 = p->o1 = p->i2 = p->i1 = 0;
return SOX_SUCCESS;
}
int lsx_biquad_start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
start(effp);
if (effp->global_info->plot == sox_plot_octave) {
printf(
"%% GNU Octave file (may also work with MATLAB(R) )\n"
"Fs=%g;minF=10;maxF=Fs/2;\n"
"sweepF=logspace(log10(minF),log10(maxF),200);\n"
"[h,w]=freqz([%.15e %.15e %.15e],[1 %.15e %.15e],sweepF,Fs);\n"
"semilogx(w,20*log10(h))\n"
"title('SoX effect: %s gain=%g frequency=%g %s=%g (rate=%g)')\n"
"xlabel('Frequency (Hz)')\n"
"ylabel('Amplitude Response (dB)')\n"
"axis([minF maxF -35 25])\n"
"grid on\n"
"disp('Hit return to continue')\n"
"pause\n"
, effp->in_signal.rate, p->b0, p->b1, p->b2, p->a1, p->a2
, effp->handler.name, p->gain, p->fc, width_str[p->width_type], p->width
, effp->in_signal.rate);
return SOX_EOF;
}
if (effp->global_info->plot == sox_plot_gnuplot) {
printf(
"# gnuplot file\n"
"set title 'SoX effect: %s gain=%g frequency=%g %s=%g (rate=%g)'\n"
"set xlabel 'Frequency (Hz)'\n"
"set ylabel 'Amplitude Response (dB)'\n"
"Fs=%g\n"
"b0=%.15e; b1=%.15e; b2=%.15e; a1=%.15e; a2=%.15e\n"
"o=2*pi/Fs\n"
"H(f)=sqrt((b0*b0+b1*b1+b2*b2+2.*(b0*b1+b1*b2)*cos(f*o)+2.*(b0*b2)*cos(2.*f*o))/(1.+a1*a1+a2*a2+2.*(a1+a1*a2)*cos(f*o)+2.*a2*cos(2.*f*o)))\n"
"set logscale x\n"
"set samples 250\n"
"set grid xtics ytics\n"
"set key off\n"
"plot [f=10:Fs/2] [-35:25] 20*log10(H(f))\n"
"pause -1 'Hit return to continue'\n"
, effp->handler.name, p->gain, p->fc, width_str[p->width_type], p->width
, effp->in_signal.rate, effp->in_signal.rate
, p->b0, p->b1, p->b2, p->a1, p->a2);
return SOX_EOF;
}
if (effp->global_info->plot == sox_plot_data) {
printf("# SoX effect: %s gain=%g frequency=%g %s=%g (rate=%g)\n"
"# IIR filter\n"
"# rate: %g\n"
"# name: b\n"
"# type: matrix\n"
"# rows: 3\n"
"# columns: 1\n"
"%24.16e\n%24.16e\n%24.16e\n"
"# name: a\n"
"# type: matrix\n"
"# rows: 3\n"
"# columns: 1\n"
"%24.16e\n%24.16e\n%24.16e\n"
, effp->handler.name, p->gain, p->fc, width_str[p->width_type], p->width
, effp->in_signal.rate, effp->in_signal.rate
, p->b0, p->b1, p->b2, 1. /* a0 */, p->a1, p->a2);
return SOX_EOF;
}
return SOX_SUCCESS;
}
int lsx_biquad_flow(sox_effect_t * effp, const sox_sample_t *ibuf,
sox_sample_t *obuf, size_t *isamp, size_t *osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len = *isamp = *osamp = min(*isamp, *osamp);
while (len--) {
double o0 = *ibuf*p->b0 + p->i1*p->b1 + p->i2*p->b2 - p->o1*p->a1 - p->o2*p->a2;
p->i2 = p->i1, p->i1 = *ibuf++;
p->o2 = p->o1, p->o1 = o0;
*obuf++ = SOX_ROUND_CLIP_COUNT(o0, effp->clips);
}
return SOX_SUCCESS;
}
static int create(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
double * d = &p->b0;
char c;
--argc, ++argv;
if (argc == 6)
for (; argc && sscanf(*argv, "%lf%c", d, &c) == 1; --argc, ++argv, ++d);
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_biquad_effect_fn(void)
{
static sox_effect_handler_t handler = {
"biquad", "b0 b1 b2 a0 a1 a2", 0,
create, lsx_biquad_start, lsx_biquad_flow, NULL, NULL, NULL, sizeof(priv_t)
};
return &handler;
}

78
app/src/main/cpp/sox/biquad.h
Unescape View File

@ -0,0 +1,78 @@
/* libSoX Biquad filter common definitions (c) 2006-7 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef biquad_included
#define biquad_included
#define LSX_EFF_ALIAS
#include "sox_i.h"
typedef enum {
filter_LPF,
filter_HPF,
filter_BPF_CSG,
filter_BPF,
filter_notch,
filter_APF,
filter_peakingEQ,
filter_lowShelf,
filter_highShelf,
filter_LPF_1,
filter_HPF_1,
filter_BPF_SPK,
filter_BPF_SPK_N,
filter_AP1,
filter_AP2,
filter_deemph,
filter_riaa
} filter_t;
typedef enum {
width_bw_Hz,
width_bw_kHz,
/* The old, non-RBJ, non-freq-warped band-pass/reject response;
* leaving here for now just in case anybody misses it: */
width_bw_old,
width_bw_oct,
width_Q,
width_slope
} width_t;
/* Private data for the biquad filter effects */
typedef struct {
double gain; /* For EQ filters */
double fc; /* Centre/corner/cutoff frequency */
double width; /* Filter width; interpreted as per width_type */
width_t width_type;
filter_t filter_type;
double b0, b1, b2; /* Filter coefficients */
double a0, a1, a2; /* Filter coefficients */
sox_sample_t i1, i2; /* Filter memory */
double o1, o2; /* Filter memory */
} biquad_t;
int lsx_biquad_getopts(sox_effect_t * effp, int n, char **argv,
int min_args, int max_args, int fc_pos, int width_pos, int gain_pos,
char const * allowed_width_types, filter_t filter_type);
int lsx_biquad_start(sox_effect_t * effp);
int lsx_biquad_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp);
#endif

416
app/src/main/cpp/sox/biquads.c
Unescape View File

@ -0,0 +1,416 @@
/* libSoX Biquad filter effects (c) 2006-8 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*
* 2-pole filters designed by Robert Bristow-Johnson <rbj@audioimagination.com>
* see https://webaudio.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html
*
* 1-pole filters based on code (c) 2000 Chris Bagwell <cbagwell@sprynet.com>
* Algorithms: Recursive single pole low/high pass filter
* Reference: The Scientist and Engineer's Guide to Digital Signal Processing
*
* low-pass: output[N] = input[N] * A + output[N-1] * B
* X = exp(-2.0 * pi * Fc)
* A = 1 - X
* B = X
* Fc = cutoff freq / sample rate
*
* Mimics an RC low-pass filter:
*
* ---/\/\/\/\----------->
* |
* --- C
* ---
* |
* |
* V
*
* high-pass: output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
* X = exp(-2.0 * pi * Fc)
* A0 = (1 + X) / 2
* A1 = -(1 + X) / 2
* B1 = X
* Fc = cutoff freq / sample rate
*
* Mimics an RC high-pass filter:
*
* || C
* ----||--------->
* || |
* <
* > R
* <
* |
* V
*/
#include "biquad.h"
#include <assert.h>
#include <string.h>
typedef biquad_t priv_t;
static int hilo1_getopts(sox_effect_t * effp, int argc, char **argv) {
return lsx_biquad_getopts(effp, argc, argv, 1, 1, 0, 1, 2, "",
*effp->handler.name == 'l'? filter_LPF_1 : filter_HPF_1);
}
static int hilo2_getopts(sox_effect_t * effp, int argc, char **argv) {
priv_t * p = (priv_t *)effp->priv;
if (argc > 1 && strcmp(argv[1], "-1") == 0)
return hilo1_getopts(effp, argc - 1, argv + 1);
if (argc > 1 && strcmp(argv[1], "-2") == 0)
++argv, --argc;
p->width = sqrt(0.5); /* Default to Butterworth */
return lsx_biquad_getopts(effp, argc, argv, 1, 2, 0, 1, 2, "qohk",
*effp->handler.name == 'l'? filter_LPF : filter_HPF);
}
static int bandpass_getopts(sox_effect_t * effp, int argc, char **argv) {
filter_t type = filter_BPF;
if (argc > 1 && strcmp(argv[1], "-c") == 0)
++argv, --argc, type = filter_BPF_CSG;
return lsx_biquad_getopts(effp, argc, argv, 2, 2, 0, 1, 2, "hkqob", type);
}
static int bandrej_getopts(sox_effect_t * effp, int argc, char **argv) {
return lsx_biquad_getopts(effp, argc, argv, 2, 2, 0, 1, 2, "hkqob", filter_notch);
}
static int allpass_getopts(sox_effect_t * effp, int argc, char **argv) {
filter_t type = filter_APF;
int m;
if (argc > 1 && strcmp(argv[1], "-1") == 0)
++argv, --argc, type = filter_AP1;
else if (argc > 1 && strcmp(argv[1], "-2") == 0)
++argv, --argc, type = filter_AP2;
m = 1 + (type == filter_APF);
return lsx_biquad_getopts(effp, argc, argv, m, m, 0, 1, 2, "hkqo", type);
}
static int tone_getopts(sox_effect_t * effp, int argc, char **argv) {
priv_t * p = (priv_t *)effp->priv;
p->width = 0.5;
p->fc = *effp->handler.name == 'b'? 100 : 3000;
return lsx_biquad_getopts(effp, argc, argv, 1, 3, 1, 2, 0, "shkqo",
*effp->handler.name == 'b'? filter_lowShelf: filter_highShelf);
}
static int equalizer_getopts(sox_effect_t * effp, int argc, char **argv) {
return lsx_biquad_getopts(effp, argc, argv, 3, 3, 0, 1, 2, "qohk", filter_peakingEQ);
}
static int band_getopts(sox_effect_t * effp, int argc, char **argv) {
filter_t type = filter_BPF_SPK;
if (argc > 1 && strcmp(argv[1], "-n") == 0)
++argv, --argc, type = filter_BPF_SPK_N;
return lsx_biquad_getopts(effp, argc, argv, 1, 2, 0, 1, 2, "hkqo", type);
}
static int deemph_getopts(sox_effect_t * effp, int argc, char **argv) {
return lsx_biquad_getopts(effp, argc, argv, 0, 0, 0, 1, 2, "s", filter_deemph);
}
static int riaa_getopts(sox_effect_t * effp, int argc, char **argv) {
priv_t * p = (priv_t *)effp->priv;
p->filter_type = filter_riaa;
(void)argv;
return --argc? lsx_usage(effp) : SOX_SUCCESS;
}
static void make_poly_from_roots(
double const * roots, size_t num_roots, double * poly)
{
size_t i, j;
poly[0] = 1;
poly[1] = -roots[0];
memset(poly + 2, 0, (num_roots + 1 - 2) * sizeof(*poly));
for (i = 1; i < num_roots; ++i)
for (j = num_roots; j > 0; --j)
poly[j] -= poly[j - 1] * roots[i];
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
double w0, A, alpha, mult;
if (p->filter_type == filter_deemph) { /* See deemph.plt for documentation */
if (effp->in_signal.rate == 44100) {
p->fc = 5283;
p->width = 0.4845;
p->gain = -9.477;
}
else if (effp->in_signal.rate == 48000) {
p->fc = 5356;
p->width = 0.479;
p->gain = -9.62;
}
else {
lsx_fail("sample rate must be 44100 (audio-CD) or 48000 (DAT)");
return SOX_EOF;
}
}
w0 = 2 * M_PI * p->fc / effp->in_signal.rate;
A = exp(p->gain / 40 * log(10.));
alpha = 0, mult = dB_to_linear(max(p->gain, 0));
if (w0 > M_PI) {
lsx_fail("frequency must be less than half the sample-rate (Nyquist rate)");
return SOX_EOF;
}
/* Set defaults: */
p->b0 = p->b1 = p->b2 = p->a1 = p->a2 = 0;
p->a0 = 1;
if (p->width) switch (p->width_type) {
case width_slope:
alpha = sin(w0)/2 * sqrt((A + 1/A)*(1/p->width - 1) + 2);
break;
case width_Q:
alpha = sin(w0)/(2*p->width);
break;
case width_bw_oct:
alpha = sin(w0)*sinh(log(2.)/2 * p->width * w0/sin(w0));
break;
case width_bw_Hz:
alpha = sin(w0)/(2*p->fc/p->width);
break;
case width_bw_kHz: assert(0); /* Shouldn't get here */
case width_bw_old:
alpha = tan(M_PI * p->width / effp->in_signal.rate);
break;
}
switch (p->filter_type) {
case filter_LPF: /* H(s) = 1 / (s^2 + s/Q + 1) */
p->b0 = (1 - cos(w0))/2;
p->b1 = 1 - cos(w0);
p->b2 = (1 - cos(w0))/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_HPF: /* H(s) = s^2 / (s^2 + s/Q + 1) */
p->b0 = (1 + cos(w0))/2;
p->b1 = -(1 + cos(w0));
p->b2 = (1 + cos(w0))/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_BPF_CSG: /* H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q) */
p->b0 = sin(w0)/2;
p->b1 = 0;
p->b2 = -sin(w0)/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_BPF: /* H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain) */
p->b0 = alpha;
p->b1 = 0;
p->b2 = -alpha;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_notch: /* H(s) = (s^2 + 1) / (s^2 + s/Q + 1) */
p->b0 = 1;
p->b1 = -2*cos(w0);
p->b2 = 1;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_APF: /* H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1) */
p->b0 = 1 - alpha;
p->b1 = -2*cos(w0);
p->b2 = 1 + alpha;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_peakingEQ: /* H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1) */
if (A == 1)
return SOX_EFF_NULL;
p->b0 = 1 + alpha*A;
p->b1 = -2*cos(w0);
p->b2 = 1 - alpha*A;
p->a0 = 1 + alpha/A;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha/A;
break;
case filter_lowShelf: /* H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1) */
if (A == 1)
return SOX_EFF_NULL;
p->b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
p->b1 = 2*A*( (A-1) - (A+1)*cos(w0) );
p->b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
p->a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
p->a1 = -2*( (A-1) + (A+1)*cos(w0) );
p->a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
break;
case filter_deemph: /* Falls through to high-shelf... */
case filter_highShelf: /* H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A) */
if (!A)
return SOX_EFF_NULL;
p->b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
p->b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
p->b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
p->a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
p->a1 = 2*( (A-1) - (A+1)*cos(w0) );
p->a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
break;
case filter_LPF_1: /* single-pole */
p->a1 = -exp(-w0);
p->b0 = 1 + p->a1;
break;
case filter_HPF_1: /* single-pole */
p->a1 = -exp(-w0);
p->b0 = (1 - p->a1)/2;
p->b1 = -p->b0;
break;
case filter_BPF_SPK: case filter_BPF_SPK_N: {
double bw_Hz;
if (!p->width)
p->width = p->fc / 2;
bw_Hz = p->width_type == width_Q? p->fc / p->width :
p->width_type == width_bw_Hz? p->width :
p->fc * (pow(2., p->width) - 1) * pow(2., -0.5 * p->width); /* bw_oct */
#include "band.h" /* Has different licence */
break;
}
case filter_AP1: /* Experimental 1-pole all-pass from Tom Erbe @ UCSD */
p->b0 = exp(-w0);
p->b1 = -1;
p->a1 = -exp(-w0);
break;
case filter_AP2: /* Experimental 2-pole all-pass from Tom Erbe @ UCSD */
p->b0 = 1 - sin(w0);
p->b1 = -2 * cos(w0);
p->b2 = 1 + sin(w0);
p->a0 = 1 + sin(w0);
p->a1 = -2 * cos(w0);
p->a2 = 1 - sin(w0);
break;
case filter_riaa: /* http://www.dsprelated.com/showmessage/73300/3.php */
if (effp->in_signal.rate == 44100) {
static const double zeros[] = {-0.2014898, 0.9233820};
static const double poles[] = {0.7083149, 0.9924091};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 48000) {
static const double zeros[] = {-0.1766069, 0.9321590};
static const double poles[] = {0.7396325, 0.9931330};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 88200) {
static const double zeros[] = {-0.1168735, 0.9648312};
static const double poles[] = {0.8590646, 0.9964002};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 96000) {
static const double zeros[] = {-0.1141486, 0.9676817};
static const double poles[] = {0.8699137, 0.9966946};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 192000) {
static const double zeros[] = {-0.1040610965, 0.9837523263};
static const double poles[] = {0.9328992971, 0.9983633125};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else {
lsx_fail("Sample rate must be 44.1k, 48k, 88.2k, 96k, or 192k");
return SOX_EOF;
}
{ /* Normalise to 0dB at 1kHz (Thanks to Glenn Davis) */
double y = 2 * M_PI * 1000 / effp->in_signal.rate;
double b_re = p->b0 + p->b1 * cos(-y) + p->b2 * cos(-2 * y);
double a_re = p->a0 + p->a1 * cos(-y) + p->a2 * cos(-2 * y);
double b_im = p->b1 * sin(-y) + p->b2 * sin(-2 * y);
double a_im = p->a1 * sin(-y) + p->a2 * sin(-2 * y);
double g = 1 / sqrt((sqr(b_re) + sqr(b_im)) / (sqr(a_re) + sqr(a_im)));
p->b0 *= g; p->b1 *= g; p->b2 *= g;
}
mult = (p->b0 + p->b1 + p->b2) / (p->a0 + p->a1 + p->a2);
lsx_debug("gain=%f", linear_to_dB(mult));
break;
}
if (effp->in_signal.mult)
*effp->in_signal.mult /= mult;
return lsx_biquad_start(effp);
}
#define BIQUAD_EFFECT(name,group,usage,flags) \
sox_effect_handler_t const * lsx_##name##_effect_fn(void) { \
static sox_effect_handler_t handler = { \
#name, usage, flags, \
group##_getopts, start, lsx_biquad_flow, 0, 0, 0, sizeof(biquad_t)\
}; \
return &handler; \
}
BIQUAD_EFFECT(highpass, hilo2, "[-1|-2] frequency [width[q|o|h|k](0.707q)]", 0)
BIQUAD_EFFECT(lowpass, hilo2, "[-1|-2] frequency [width[q|o|h|k]](0.707q)", 0)
BIQUAD_EFFECT(bandpass, bandpass, "[-c] frequency width[h|k|q|o]", 0)
BIQUAD_EFFECT(bandreject,bandrej, "frequency width[h|k|q|o]", 0)
BIQUAD_EFFECT(allpass, allpass, "frequency width[h|k|q|o]", 0)
BIQUAD_EFFECT(bass, tone, "gain [frequency(100) [width[s|h|k|q|o]](0.5s)]", 0)
BIQUAD_EFFECT(treble, tone, "gain [frequency(3000) [width[s|h|k|q|o]](0.5s)]", 0)
BIQUAD_EFFECT(equalizer, equalizer,"frequency width[q|o|h|k] gain", 0)
BIQUAD_EFFECT(band, band, "[-n] center [width[h|k|q|o]]", 0)
BIQUAD_EFFECT(deemph, deemph, NULL, 0)
BIQUAD_EFFECT(riaa, riaa, NULL, 0)

273
app/src/main/cpp/sox/bit-rot/CoreAudio/CoreAudio.h
Unescape View File

@ -0,0 +1,273 @@
#ifndef HAVE_COREAUDIO
/*
* SoX bit-rot detection file; cobbled together
*/
enum {
kAudioHardwarePropertyProcessIsMaster,
kAudioHardwarePropertyIsInitingOrExiting,
kAudioHardwarePropertyDevices,
kAudioHardwarePropertyDefaultInputDevice,
kAudioHardwarePropertyDefaultOutputDevice,
kAudioHardwarePropertyDefaultSystemOutputDevice,
kAudioHardwarePropertyDeviceForUID,
kAudioHardwarePropertySleepingIsAllowed,
kAudioHardwarePropertyUnloadingIsAllowed,
kAudioHardwarePropertyHogModeIsAllowed,
kAudioHardwarePropertyRunLoop,
kAudioHardwarePropertyPlugInForBundleID
};
enum {
kAudioObjectPropertyClass,
kAudioObjectPropertyOwner,
kAudioObjectPropertyCreator,
kAudioObjectPropertyName,
kAudioObjectPropertyManufacturer,
kAudioObjectPropertyElementName,
kAudioObjectPropertyElementCategoryName,
kAudioObjectPropertyElementNumberName,
kAudioObjectPropertyOwnedObjects,
kAudioObjectPropertyListenerAdded,
kAudioObjectPropertyListenerRemoved
};
enum {
kAudioDevicePropertyDeviceName,
kAudioDevicePropertyDeviceNameCFString = kAudioObjectPropertyName,
kAudioDevicePropertyDeviceManufacturer,
kAudioDevicePropertyDeviceManufacturerCFString =
kAudioObjectPropertyManufacturer,
kAudioDevicePropertyRegisterBufferList,
kAudioDevicePropertyBufferSize,
kAudioDevicePropertyBufferSizeRange,
kAudioDevicePropertyChannelName,
kAudioDevicePropertyChannelNameCFString = kAudioObjectPropertyElementName,
kAudioDevicePropertyChannelCategoryName,
kAudioDevicePropertyChannelCategoryNameCFString =
kAudioObjectPropertyElementCategoryName,
kAudioDevicePropertyChannelNumberName,
kAudioDevicePropertyChannelNumberNameCFString =
kAudioObjectPropertyElementNumberName,
kAudioDevicePropertySupportsMixing,
kAudioDevicePropertyStreamFormat,
kAudioDevicePropertyStreamFormats,
kAudioDevicePropertyStreamFormatSupported,
kAudioDevicePropertyStreamFormatMatch,
kAudioDevicePropertyDataSourceNameForID,
kAudioDevicePropertyClockSourceNameForID,
kAudioDevicePropertyPlayThruDestinationNameForID,
kAudioDevicePropertyChannelNominalLineLevelNameForID
};
enum {
kAudioDevicePropertyPlugIn,
kAudioDevicePropertyConfigurationApplication,
kAudioDevicePropertyDeviceUID,
kAudioDevicePropertyModelUID,
kAudioDevicePropertyTransportType,
kAudioDevicePropertyRelatedDevices,
kAudioDevicePropertyClockDomain,
kAudioDevicePropertyDeviceIsAlive,
kAudioDevicePropertyDeviceHasChanged,
kAudioDevicePropertyDeviceIsRunning,
kAudioDevicePropertyDeviceIsRunningSomewhere,
kAudioDevicePropertyDeviceCanBeDefaultDevice,
kAudioDevicePropertyDeviceCanBeDefaultSystemDevice,
kAudioDeviceProcessorOverload,
kAudioDevicePropertyHogMode,
kAudioDevicePropertyLatency,
kAudioDevicePropertyBufferFrameSize,
kAudioDevicePropertyBufferFrameSizeRange,
kAudioDevicePropertyUsesVariableBufferFrameSizes,
kAudioDevicePropertyStreams,
kAudioDevicePropertySafetyOffset,
kAudioDevicePropertyIOCycleUsage,
kAudioDevicePropertyStreamConfiguration,
kAudioDevicePropertyIOProcStreamUsage,
kAudioDevicePropertyPreferredChannelsForStereo,
kAudioDevicePropertyPreferredChannelLayout,
kAudioDevicePropertyNominalSampleRate,
kAudioDevicePropertyAvailableNominalSampleRates,
kAudioDevicePropertyActualSampleRate
};
enum {
kAudioFormatLinearPCM,
kAudioFormatAC3,
kAudioFormat60958AC3,
kAudioFormatAppleIMA4,
kAudioFormatMPEG4AAC,
kAudioFormatMPEG4CELP,
kAudioFormatMPEG4HVXC,
kAudioFormatMPEG4TwinVQ,
kAudioFormatMACE3,
kAudioFormatMACE6,
kAudioFormatULaw,
kAudioFormatALaw,
kAudioFormatQDesign,
kAudioFormatQDesign2,
kAudioFormatQUALCOMM,
kAudioFormatMPEGLayer1,
kAudioFormatMPEGLayer2,
kAudioFormatMPEGLayer3,
kAudioFormatDVAudio,
kAudioFormatVariableDurationDVAudio,
kAudioFormatTimeCode,
kAudioFormatMIDIStream,
kAudioFormatParameterValueStream,
kAudioFormatAppleLossless
};
enum {
kAudioFormatFlagIsFloat = (1L << 0),
kAudioFormatFlagIsBigEndian = (1L << 1),
kAudioFormatFlagIsSignedInteger = (1L << 2),
kAudioFormatFlagIsPacked = (1L << 3),
kAudioFormatFlagIsAlignedHigh = (1L << 4),
kAudioFormatFlagIsNonInterleaved = (1L << 5),
kAudioFormatFlagIsNonMixable = (1L << 6),
kLinearPCMFormatFlagIsFloat = kAudioFormatFlagIsFloat,
kLinearPCMFormatFlagIsBigEndian = kAudioFormatFlagIsBigEndian,
kLinearPCMFormatFlagIsSignedInteger = kAudioFormatFlagIsSignedInteger,
kLinearPCMFormatFlagIsPacked = kAudioFormatFlagIsPacked,
kLinearPCMFormatFlagIsAlignedHigh = kAudioFormatFlagIsAlignedHigh,
kLinearPCMFormatFlagIsNonInterleaved = kAudioFormatFlagIsNonInterleaved,
kLinearPCMFormatFlagIsNonMixable = kAudioFormatFlagIsNonMixable,
kAppleLosslessFormatFlag_16BitSourceData = 1,
kAppleLosslessFormatFlag_20BitSourceData = 2,
kAppleLosslessFormatFlag_24BitSourceData = 3,
kAppleLosslessFormatFlag_32BitSourceData = 4
};
enum {
kAudioFormatFlagsNativeEndian = kAudioFormatFlagIsBigEndian,
kAudioFormatFlagsNativeFloatPacked =
kAudioFormatFlagIsFloat | kAudioFormatFlagsNativeEndian |
kAudioFormatFlagIsPacked
};
enum {
kAudioDeviceUnknown
};
enum {
kVariableLengthArray = 1
};
enum {
kAudioHardwareNoError = 0,
noErr = kAudioHardwareNoError
};
enum {
false
};
typedef double Float64;
typedef float Float32;
typedef int SInt32;
typedef int Boolean;
typedef int OSErr;
typedef short SInt16;
typedef unsigned int UInt32;
typedef unsigned long int UInt64;
typedef SInt32 OSStatus;
typedef UInt32 AudioObjectID;
typedef UInt32 AudioHardwarePropertyID;
typedef UInt32 AudioDevicePropertyID;
typedef AudioObjectID AudioDeviceID;
struct AudioStreamBasicDescription {
Float64 mSampleRate;
UInt32 mFormatID;
UInt32 mFormatFlags;
UInt32 mBytesPerPacket;
UInt32 mFramesPerPacket;
UInt32 mBytesPerFrame;
UInt32 mChannelsPerFrame;
UInt32 mBitsPerChannel;
UInt32 mReserved;
};
typedef struct AudioStreamBasicDescription AudioStreamBasicDescription;
struct SMPTETime {
SInt16 mSubframes;
SInt16 mSubframeDivisor;
UInt32 mCounter;
UInt32 mType;
UInt32 mFlags;
SInt16 mHours;
SInt16 mMinutes;
SInt16 mSeconds;
SInt16 mFrames;
};
typedef struct SMPTETime SMPTETime;
struct AudioTimeStamp {
Float64 mSampleTime;
UInt64 mHostTime;
Float64 mRateScalar;
UInt64 mWordClockTime;
SMPTETime mSMPTETime;
UInt32 mFlags;
UInt32 mReserved;
};
typedef struct AudioTimeStamp AudioTimeStamp;
struct AudioBuffer {
UInt32 mNumberChannels;
UInt32 mDataByteSize;
void *mData;
};
typedef struct AudioBuffer AudioBuffer;
struct AudioBufferList {
UInt32 mNumberBuffers;
AudioBuffer mBuffers[kVariableLengthArray];
};
typedef struct AudioBufferList AudioBufferList;
typedef OSStatus(*AudioDeviceIOProc) (AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const AudioBufferList * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void *inClientData);
OSStatus AudioHardwareGetProperty(AudioHardwarePropertyID inPropertyID,
UInt32 * ioPropertyDataSize,
void *outPropertyData);
OSStatus AudioHardwareGetPropertyInfo(AudioHardwarePropertyID inPropertyID,
UInt32 * ioPropertyDataSize,
void *outPropertyData);
OSStatus AudioDeviceSetProperty(AudioDeviceID inDevice,
const AudioTimeStamp * inWhen,
UInt32 inChannel, Boolean isInput,
AudioDevicePropertyID inPropertyID,
UInt32 inPropertyDataSize,
const void *inPropertyData);
OSStatus AudioDeviceGetProperty(AudioDeviceID inDevice, UInt32 inChannel,
Boolean isInput,
AudioDevicePropertyID inPropertyID,
UInt32 * ioPropertyDataSize,
void *outPropertyData);
OSStatus AudioDeviceAddIOProc(AudioDeviceID inDevice,
AudioDeviceIOProc inProc, void *inClientData);
OSStatus AudioDeviceStart(AudioDeviceID inDevice, AudioDeviceIOProc inProc);
OSStatus AudioDeviceStop(AudioDeviceID inDevice, AudioDeviceIOProc inProc);
#endif

0
app/src/main/cpp/sox/bit-rot/mmreg.h
Unescape View File

89
app/src/main/cpp/sox/bit-rot/mmsystem.h
Unescape View File

@ -0,0 +1,89 @@
/*
* SoX bit-rot detection file; cobbled together
*/
#define HWAVEIN HANDLE
#define HWAVEOUT HANDLE
#define MMRESULT UINT
#define MMVERSION UINT
#define UNALIGNED
enum {
MMSYSERR_NOERROR,
MAXPNAMELEN,
WAVE_FORMAT_PCM,
WAVE_FORMAT_EXTENSIBLE,
WAVE_FORMAT_QUERY,
WAVE_MAPPER,
WAVERR_STILLPLAYING,
WHDR_DONE,
WHDR_INQUEUE
};
typedef struct wavehdr_tag {
LPSTR lpData;
DWORD dwBufferLength;
DWORD dwBytesRecorded;
DWORD dwUser;
DWORD dwFlags;
DWORD dwLoops;
struct wavehdr_tag *lpNext;
DWORD reserved;
} WAVEHDR,*PWAVEHDR,*LPWAVEHDR;
typedef struct _WAVEFORMATEX {
WORD wFormatTag;
WORD nChannels;
DWORD nSamplesPerSec;
DWORD nAvgBytesPerSec;
WORD nBlockAlign;
WORD wBitsPerSample;
WORD cbSize;
} WAVEFORMATEX, *PWAVEFORMATEX, *NPWAVEFORMATEX, *LPWAVEFORMATEX;
typedef struct tagWAVEINCAPSA {
WORD wMid;
WORD wPid;
MMVERSION vDriverVersion;
CHAR szPname[MAXPNAMELEN];
DWORD dwFormats;
WORD wChannels;
WORD wReserved1;
} WAVEINCAPSA,*PWAVEINCAPSA,*LPWAVEINCAPSA;
typedef struct tagWAVEOUTCAPSA {
WORD wMid;
WORD wPid;
MMVERSION vDriverVersion;
CHAR szPname[MAXPNAMELEN];
DWORD dwFormats;
WORD wChannels;
WORD wReserved1;
DWORD dwSupport;
} WAVEOUTCAPSA,*PWAVEOUTCAPSA,*LPWAVEOUTCAPSA;
MMRESULT WINAPI waveInAddBuffer(HWAVEIN,LPWAVEHDR,UINT);
MMRESULT WINAPI waveInClose(HWAVEIN);
MMRESULT WINAPI waveInGetDevCapsA(UINT,LPWAVEINCAPSA,UINT);
MMRESULT WINAPI waveInGetNumDevs(void);
MMRESULT WINAPI waveInOpen(HWAVEIN*,UINT,WAVEFORMATEX*,DWORD*,DWORD*,DWORD);
MMRESULT WINAPI waveInPrepareHeader(HWAVEIN,LPWAVEHDR,UINT);
MMRESULT WINAPI waveInReset(HWAVEIN);
MMRESULT WINAPI waveInStart(HWAVEIN);
MMRESULT WINAPI waveOutClose(HWAVEOUT);
MMRESULT WINAPI waveOutGetDevCapsA(UINT,LPWAVEOUTCAPSA,UINT);
MMRESULT WINAPI waveOutGetNumDevs(void);
MMRESULT WINAPI waveOutOpen(HWAVEOUT*,UINT,WAVEFORMATEX*,DWORD*,DWORD*,DWORD);
MMRESULT WINAPI waveOutPrepareHeader(HWAVEOUT,LPWAVEHDR,UINT);
MMRESULT WINAPI waveOutReset(HWAVEOUT);
MMRESULT WINAPI waveOutWrite(HWAVEOUT,LPWAVEHDR,UINT);
typedef struct {
WAVEFORMATEX Format;
union {
WORD wValidBitsPerSample;
WORD wSamplesPerBlock;
WORD wReserved;
} Samples;
DWORD dwChannelMask;
GUID SubFormat;
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;

160
app/src/main/cpp/sox/bit-rot/sndio.h
Unescape View File

@ -0,0 +1,160 @@
#ifndef HAVE_SNDIO
/*
* SoX bit-rot detection file, obtained from:
* http://www.openbsd.org/cgi-bin/cvsweb/src/lib/libsndio/sndio.h
*/
#if defined __GNUC__
#pragma GCC system_header
#endif
/* $OpenBSD: sndio.h,v 1.7 2009/02/03 19:44:58 ratchov Exp $ */
/*
* Copyright (c) 2008 Alexandre Ratchov <alex@caoua.org>
*
* Permission to use, copy, modify, and distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef SNDIO_H
#define SNDIO_H
#include <sys/param.h>
/*
* private ``handle'' structure
*/
struct sio_hdl;
/*
* parameters of a full-duplex stream
*/
struct sio_par {
unsigned bits; /* bits per sample */
unsigned bps; /* bytes per sample */
unsigned sig; /* 1 = signed, 0 = unsigned */
unsigned le; /* 1 = LE, 0 = BE byte order */
unsigned msb; /* 1 = MSB, 0 = LSB aligned */
unsigned rchan; /* number channels for recording direction */
unsigned pchan; /* number channels for playback direction */
unsigned rate; /* frames per second */
unsigned bufsz; /* end-to-end buffer size */
#define SIO_IGNORE 0 /* pause during xrun */
#define SIO_SYNC 1 /* resync after xrun */
#define SIO_ERROR 2 /* terminate on xrun */
unsigned xrun; /* what to do on overruns/underruns */
unsigned round; /* optimal bufsz divisor */
unsigned appbufsz; /* minimum buffer size */
int __pad[3]; /* for future use */
int __magic; /* for internal/debug purposes only */
};
/*
* capabilities of a stream
*/
struct sio_cap {
#define SIO_NENC 8
#define SIO_NCHAN 8
#define SIO_NRATE 16
#define SIO_NCONF 4
struct sio_enc { /* allowed sample encodings */
unsigned bits;
unsigned bps;
unsigned sig;
unsigned le;
unsigned msb;
} enc[SIO_NENC];
unsigned rchan[SIO_NCHAN]; /* allowed values for rchan */
unsigned pchan[SIO_NCHAN]; /* allowed values for pchan */
unsigned rate[SIO_NRATE]; /* allowed rates */
int __pad[7]; /* for future use */
unsigned nconf; /* number of elements in confs[] */
struct sio_conf {
unsigned enc; /* mask of enc[] indexes */
unsigned rchan; /* mask of chan[] indexes (rec) */
unsigned pchan; /* mask of chan[] indexes (play) */
unsigned rate; /* mask of rate[] indexes */
} confs[SIO_NCONF];
};
#define SIO_XSTRINGS { "ignore", "sync", "error" }
/*
* mode bitmap
*/
#define SIO_PLAY 1
#define SIO_REC 2
/*
* maximum size of the encording string (the longest possible
* encoding is ``s24le3msb'')
*/
#define SIO_ENCMAX 10
/*
* default bytes per sample for the given bits per sample
*/
#define SIO_BPS(bits) (((bits) <= 8) ? 1 : (((bits) <= 16) ? 2 : 4))
/*
* default value of "sio_par->le" flag
*/
#if BYTE_ORDER == LITTLE_ENDIAN
#define SIO_LE_NATIVE 1
#else
#define SIO_LE_NATIVE 0
#endif
/*
* default device for the sun audio(4) back-end
*/
#define SIO_SUN_PATH "/dev/audio"
/*
* default socket name for the aucat(1) back-end
*/
#define SIO_AUCAT_PATH "default"
/*
* maximum value of volume, eg. for sio_setvol()
*/
#define SIO_MAXVOL 127
#ifdef __cplusplus
extern "C" {
#endif
int sio_strtoenc(struct sio_par *, char *);
int sio_enctostr(struct sio_par *, char *);
void sio_initpar(struct sio_par *);
struct sio_hdl *sio_open(char *, unsigned, int);
void sio_close(struct sio_hdl *);
int sio_setpar(struct sio_hdl *, struct sio_par *);
int sio_getpar(struct sio_hdl *, struct sio_par *);
int sio_getcap(struct sio_hdl *, struct sio_cap *);
void sio_onmove(struct sio_hdl *, void (*)(void *, int), void *);
size_t sio_write(struct sio_hdl *, void *, size_t);
size_t sio_read(struct sio_hdl *, void *, size_t);
int sio_start(struct sio_hdl *);
int sio_stop(struct sio_hdl *);
int sio_nfds(struct sio_hdl *);
int sio_pollfd(struct sio_hdl *, struct pollfd *, int);
int sio_revents(struct sio_hdl *, struct pollfd *);
int sio_eof(struct sio_hdl *);
int sio_setvol(struct sio_hdl *, unsigned);
void sio_onvol(struct sio_hdl *, void (*)(void *, unsigned), void *);
#ifdef __cplusplus
}
#endif
#endif /* !defined(SNDIO_H) */
#endif

289
app/src/main/cpp/sox/bit-rot/sys/audioio.h
Unescape View File

@ -0,0 +1,289 @@
#ifndef HAVE_SUN_AUDIO
/*
* SoX bit-rot detection file, obtained from: Solaris 9 /usr/include/sys
*/
#if defined __GNUC__
#pragma GCC system_header
#endif
/*
* Copyright (c) 1995-2001 by Sun Microsystems, Inc.
* All rights reserved.
*/
#ifndef _SYS_AUDIOIO_H
#define _SYS_AUDIOIO_H
#pragma ident "@(#)audioio.h 1.30 01/01/10 SMI"
#include <sys/types.h>
#if 0
#include <sys/types32.h>
#include <sys/time.h>
#include <sys/ioccom.h>
#else
#define ushort_t unsigned short
#define uint_t unsigned int
#define uchar_t unsigned char
struct timeval32
{
unsigned tv_sec;
unsigned tv_usec;
};
#endif
/*
* These are the ioctl calls for all Solaris audio devices, including
* the x86 and SPARCstation audio devices.
*
* You are encouraged to design your code in a modular fashion so that
* future changes to the interface can be incorporated with little
* trouble.
*/
#ifdef __cplusplus
extern "C" {
#endif
/*
* This structure contains state information for audio device IO streams.
*/
struct audio_prinfo {
/*
* The following values describe the audio data encoding.
*/
uint_t sample_rate; /* samples per second */
uint_t channels; /* number of interleaved channels */
uint_t precision; /* bit-width of each sample */
uint_t encoding; /* data encoding method */
/*
* The following values control audio device configuration
*/
uint_t gain; /* gain level: 0 - 255 */
uint_t port; /* selected I/O port (see below) */
uint_t avail_ports; /* available I/O ports (see below) */
uint_t mod_ports; /* I/O ports that are modifiable (see below) */
uint_t _xxx; /* Reserved for future use */
uint_t buffer_size; /* I/O buffer size */
/*
* The following values describe driver state
*/
uint_t samples; /* number of samples converted */
uint_t eof; /* End Of File counter (play only) */
uchar_t pause; /* non-zero for pause, zero to resume */
uchar_t error; /* non-zero if overflow/underflow */
uchar_t waiting; /* non-zero if a process wants access */
uchar_t balance; /* stereo channel balance */
ushort_t minordev;
/*
* The following values are read-only state flags
*/
uchar_t open; /* non-zero if open access permitted */
uchar_t active; /* non-zero if I/O is active */
};
typedef struct audio_prinfo audio_prinfo_t;
/*
* This structure describes the current state of the audio device.
*/
struct audio_info {
/*
* Per-stream information
*/
audio_prinfo_t play; /* output status information */
audio_prinfo_t record; /* input status information */
/*
* Per-unit/channel information
*/
uint_t monitor_gain; /* input to output mix: 0 - 255 */
uchar_t output_muted; /* non-zero if output is muted */
uchar_t ref_cnt; /* driver reference count, read only */
uchar_t _xxx[2]; /* Reserved for future use */
uint_t hw_features; /* hardware features this driver supports */
uint_t sw_features; /* supported SW features */
uint_t sw_features_enabled; /* supported SW feat. enabled */
};
typedef struct audio_info audio_info_t;
/*
* Audio encoding types
*/
#define AUDIO_ENCODING_NONE (0) /* no encoding assigned */
#define AUDIO_ENCODING_ULAW (1) /* u-law encoding */
#define AUDIO_ENCODING_ALAW (2) /* A-law encoding */
#define AUDIO_ENCODING_LINEAR (3) /* Signed Linear PCM encoding */
#define AUDIO_ENCODING_DVI (104) /* DVI ADPCM */
#define AUDIO_ENCODING_LINEAR8 (105) /* 8 bit UNSIGNED */
/*
* These ranges apply to record, play, and monitor gain values
*/
#define AUDIO_MIN_GAIN (0) /* minimum gain value */
#define AUDIO_MAX_GAIN (255) /* maximum gain value */
#define AUDIO_MID_GAIN (AUDIO_MAX_GAIN / 2)
/*
* These values apply to the balance field to adjust channel gain values
*/
#define AUDIO_LEFT_BALANCE (0) /* left channel only */
#define AUDIO_MID_BALANCE (32) /* equal left/right channel */
#define AUDIO_RIGHT_BALANCE (64) /* right channel only */
#define AUDIO_BALANCE_SHIFT (3)
/*
* Generic minimum/maximum limits for number of channels, both modes
*/
#define AUDIO_CHANNELS_MONO (1)
#define AUDIO_CHANNELS_STEREO (2)
#define AUDIO_MIN_PLAY_CHANNELS (AUDIO_CHANNELS_MONO)
#define AUDIO_MAX_PLAY_CHANNELS (AUDIO_CHANNELS_STEREO)
#define AUDIO_MIN_REC_CHANNELS (AUDIO_CHANNELS_MONO)
#define AUDIO_MAX_REC_CHANNELS (AUDIO_CHANNELS_STEREO)
/*
* Generic minimum/maximum limits for sample precision
*/
#define AUDIO_PRECISION_8 (8)
#define AUDIO_PRECISION_16 (16)
#define AUDIO_MIN_PLAY_PRECISION (8)
#define AUDIO_MAX_PLAY_PRECISION (32)
#define AUDIO_MIN_REC_PRECISION (8)
#define AUDIO_MAX_REC_PRECISION (32)
/*
* Define some convenient names for typical audio ports
*/
#define AUDIO_NONE 0x00 /* all ports off */
/*
* output ports (several may be enabled simultaneously)
*/
#define AUDIO_SPEAKER 0x01 /* output to built-in speaker */
#define AUDIO_HEADPHONE 0x02 /* output to headphone jack */
#define AUDIO_LINE_OUT 0x04 /* output to line out */
#define AUDIO_SPDIF_OUT 0x08 /* output to SPDIF port */
#define AUDIO_AUX1_OUT 0x10 /* output to aux1 out */
#define AUDIO_AUX2_OUT 0x20 /* output to aux2 out */
/*
* input ports (usually only one at a time)
*/
#define AUDIO_MICROPHONE 0x01 /* input from microphone */
#define AUDIO_LINE_IN 0x02 /* input from line in */
#define AUDIO_CD 0x04 /* input from on-board CD inputs */
#define AUDIO_INTERNAL_CD_IN AUDIO_CD /* input from internal CDROM */
#define AUDIO_SPDIF_IN 0x08 /* input from SPDIF port */
#define AUDIO_AUX1_IN 0x10 /* input from aux1 in */
#define AUDIO_AUX2_IN 0x20 /* input from aux2 in */
#define AUDIO_CODEC_LOOPB_IN 0x40 /* input from Codec internal loopback */
#define AUDIO_SUNVTS 0x80 /* SunVTS input setting-internal LB */
/*
* Define the hw_features
*/
#define AUDIO_HWFEATURE_DUPLEX 0x00000001u /* simult. play & rec support */
#define AUDIO_HWFEATURE_MSCODEC 0x00000002u /* multi-stream Codec */
#define AUDIO_HWFEATURE_IN2OUT 0x00000004u /* input to output loopback */
#define AUDIO_HWFEATURE_PLAY 0x00000008u /* device supports play */
#define AUDIO_HWFEATURE_RECORD 0x00000010u /* device supports record */
/*
* Define the sw_features
*/
#define AUDIO_SWFEATURE_MIXER 0x00000001u /* audio mixer audio pers mod */
/*
* This macro initializes an audio_info structure to 'harmless' values.
* Note that (~0) might not be a harmless value for a flag that was
* a signed int.
*/
#define AUDIO_INITINFO(i) { \
uint_t *__x__; \
for (__x__ = (uint_t *)(i); \
(char *)__x__ < (((char *)(i)) + sizeof (audio_info_t)); \
*__x__++ = ~0); \
}
/*
* Parameter for the AUDIO_GETDEV ioctl to determine current
* audio devices.
*/
#define MAX_AUDIO_DEV_LEN (16)
struct audio_device {
char name[MAX_AUDIO_DEV_LEN];
char version[MAX_AUDIO_DEV_LEN];
char config[MAX_AUDIO_DEV_LEN];
};
typedef struct audio_device audio_device_t;
/*
* Ioctl calls for the audio device.
*/
/*
* AUDIO_GETINFO retrieves the current state of the audio device.
*
* AUDIO_SETINFO copies all fields of the audio_info structure whose
* values are not set to the initialized value (-1) to the device state.
* It performs an implicit AUDIO_GETINFO to return the new state of the
* device. Note that the record.samples and play.samples fields are set
* to the last value before the AUDIO_SETINFO took effect. This allows
* an application to reset the counters while atomically retrieving the
* last value.
*
* AUDIO_DRAIN suspends the calling process until the write buffers are
* empty.
*
* AUDIO_GETDEV returns a structure of type audio_device_t which contains
* three strings. The string "name" is a short identifying string (for
* example, the SBus Fcode name string), the string "version" identifies
* the current version of the device, and the "config" string identifies
* the specific configuration of the audio stream. All fields are
* device-dependent -- see the device specific manual pages for details.
*/
#define AUDIO_GETINFO _IOR('A', 1, audio_info_t)
#define AUDIO_SETINFO _IOWR('A', 2, audio_info_t)
#define AUDIO_DRAIN _IO('A', 3)
#define AUDIO_GETDEV _IOR('A', 4, audio_device_t)
/*
* The following ioctl sets the audio device into an internal loopback mode,
* if the hardware supports this. The argument is TRUE to set loopback,
* FALSE to reset to normal operation. If the hardware does not support
* internal loopback, the ioctl should fail with EINVAL.
*/
#define AUDIO_DIAG_LOOPBACK _IOW('A', 101, int)
/*
* Structure sent up as a M_PROTO message on trace streams
*/
struct audtrace_hdr {
uint_t seq; /* Sequence number (per-aud_stream) */
int type; /* device-dependent */
#if defined(_LP64) || defined(_I32LPx)
struct timeval32 timestamp;
#else
struct timeval timestamp;
#endif
char _f[8]; /* filler */
};
typedef struct audtrace_hdr audtrace_hdr_t;
#ifdef __cplusplus
}
#endif
#endif /* _SYS_AUDIOIO_H */
#endif

34
app/src/main/cpp/sox/bit-rot/windows.h
Unescape View File

@ -0,0 +1,34 @@
/*
* SoX bit-rot detection file; cobbled together
*/
#define BYTE uint8_t
#define CHAR char
#define DWORD_PTR DWORD *
#define DWORD uint32_t
#define HANDLE void *
#define LPCSTR char *
#define LPCVOID void *
#define LPDWORD DWORD *
#define LPSTR char const *
#define UINT DWORD
#define WCHAR int16_t
#define WINAPI
#define WIN_BOOL int
#define WORD uint16_t
typedef char GUID[16];
enum {
FALSE,
TRUE,
FORMAT_MESSAGE_FROM_SYSTEM,
FORMAT_MESSAGE_IGNORE_INSERTS,
INFINITE,
CALLBACK_EVENT
};
DWORD CloseHandle(HANDLE);
DWORD FormatMessageA(DWORD,LPCVOID,DWORD,DWORD,LPSTR, DWORD,LPDWORD);
DWORD GetLastError(void);
DWORD WaitForSingleObject(HANDLE, DWORD);
HANDLE CreateEventA(LPCVOID,WIN_BOOL,WIN_BOOL,LPCSTR);

35
app/src/main/cpp/sox/caf.c
Unescape View File

@ -0,0 +1,35 @@
/* libSoX file format: CAF Copyright (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
LSX_FORMAT_HANDLER(caf)
{
static char const * const names[] = {"caf", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2, 16, 24, 32, 8, 0,
SOX_ENCODING_FLOAT, 32, 64, 0,
SOX_ENCODING_ALAW, 8, 0,
SOX_ENCODING_ULAW, 8, 0,
0};
static sox_format_handler_t handler;
handler = *lsx_sndfile_format_fn();
handler.description = "Apples's Core Audio Format";
handler.names = names;
handler.write_formats = write_encodings;
return &handler;
}

49
app/src/main/cpp/sox/cdr.c
Unescape View File

@ -0,0 +1,49 @@
/* libSoX file format: cdda (c) 2006-8 SoX contributors
* Based on an original idea by David Elliott
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
static int start(sox_format_t * ft)
{
return lsx_check_read_params(ft, 2, 44100., SOX_ENCODING_SIGN2, 16, (uint64_t)0, sox_true);
}
static int stopwrite(sox_format_t * ft)
{
unsigned const sector_num_samples = 588 * ft->signal.channels;
unsigned i = ft->olength % sector_num_samples;
if (i) while (i++ < sector_num_samples) /* Pad with silence to multiple */
lsx_writew(ft, 0); /* of 1/75th of a second. */
return SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(cdr)
{
static char const * const names[] = {"cdda", "cdr", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_SIGN2, 16, 0, 0};
static sox_rate_t const write_rates[] = {44100, 0};
static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,
"Red Book Compact Disc Digital Audio",
names, SOX_FILE_BIG_END|SOX_FILE_STEREO,
start, lsx_rawread, NULL,
NULL, lsx_rawwrite, stopwrite,
lsx_rawseek, write_encodings, write_rates, 0
};
return &handler;
}

351
app/src/main/cpp/sox/chorus.c
Unescape View File

@ -0,0 +1,351 @@
/* August 24, 1998
* Copyright (C) 1998 Juergen Mueller And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Juergen Mueller And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
/*
* Chorus effect.
*
* Flow diagram scheme for n delays ( 1 <= n <= MAX_CHORUS ):
*
* * gain-in ___
* ibuff -----+--------------------------------------------->| |
* | _________ | |
* | | | * decay 1 | |
* +---->| delay 1 |----------------------------->| |
* | |_________| | |
* | /|\ | |
* : | | |
* : +-----------------+ +--------------+ | + |
* : | Delay control 1 |<--| mod. speed 1 | | |
* : +-----------------+ +--------------+ | |
* | _________ | |
* | | | * decay n | |
* +---->| delay n |----------------------------->| |
* |_________| | |
* /|\ |___|
* | |
* +-----------------+ +--------------+ | * gain-out
* | Delay control n |<--| mod. speed n | |
* +-----------------+ +--------------+ +----->obuff
*
*
* The delay i is controled by a sine or triangle modulation i ( 1 <= i <= n).
*
* Usage:
* chorus gain-in gain-out delay-1 decay-1 speed-1 depth-1 -s1|t1 [
* delay-2 decay-2 speed-2 depth-2 -s2|-t2 ... ]
*
* Where:
* gain-in, decay-1 ... decay-n : 0.0 ... 1.0 volume
* gain-out : 0.0 ... volume
* delay-1 ... delay-n : 20.0 ... 100.0 msec
* speed-1 ... speed-n : 0.1 ... 5.0 Hz modulation 1 ... n
* depth-1 ... depth-n : 0.0 ... 10.0 msec modulated delay 1 ... n
* -s1 ... -sn : modulation by sine 1 ... n
* -t1 ... -tn : modulation by triangle 1 ... n
*
* Note:
* when decay is close to 1.0, the samples can begin clipping and the output
* can saturate!
*
* Hint:
* 1 / out-gain < gain-in ( 1 + decay-1 + ... + decay-n )
*
*/
/*
* libSoX chorus effect file.
*/
#include "sox_i.h"
#include <stdlib.h> /* Harmless, and prototypes atof() etc. --dgc */
#include <string.h>
#define MOD_SINE 0
#define MOD_TRIANGLE 1
#define MAX_CHORUS 7
typedef struct {
int num_chorus;
int modulation[MAX_CHORUS];
int counter;
long phase[MAX_CHORUS];
float *chorusbuf;
float in_gain, out_gain;
float delay[MAX_CHORUS], decay[MAX_CHORUS];
float speed[MAX_CHORUS], depth[MAX_CHORUS];
long length[MAX_CHORUS];
int *lookup_tab[MAX_CHORUS];
int depth_samples[MAX_CHORUS], samples[MAX_CHORUS];
int maxsamples;
unsigned int fade_out;
} priv_t;
/*
* Process options
*/
static int sox_chorus_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * chorus = (priv_t *) effp->priv;
int i;
--argc, ++argv;
chorus->num_chorus = 0;
i = 0;
if ( ( argc < 7 ) || (( argc - 2 ) % 5 ) )
return lsx_usage(effp);
sscanf(argv[i++], "%f", &chorus->in_gain);
sscanf(argv[i++], "%f", &chorus->out_gain);
while ( i < argc ) {
if ( chorus->num_chorus > MAX_CHORUS )
{
lsx_fail("chorus: to many delays, use less than %i delays", MAX_CHORUS);
return (SOX_EOF);
}
sscanf(argv[i++], "%f", &chorus->delay[chorus->num_chorus]);
sscanf(argv[i++], "%f", &chorus->decay[chorus->num_chorus]);
sscanf(argv[i++], "%f", &chorus->speed[chorus->num_chorus]);
sscanf(argv[i++], "%f", &chorus->depth[chorus->num_chorus]);
if ( !strcmp(argv[i], "-s"))
chorus->modulation[chorus->num_chorus] = MOD_SINE;
else if ( ! strcmp(argv[i], "-t"))
chorus->modulation[chorus->num_chorus] = MOD_TRIANGLE;
else
return lsx_usage(effp);
i++;
chorus->num_chorus++;
}
return (SOX_SUCCESS);
}
/*
* Prepare for processing.
*/
static int sox_chorus_start(sox_effect_t * effp)
{
priv_t * chorus = (priv_t *) effp->priv;
int i;
float sum_in_volume;
chorus->maxsamples = 0;
if ( chorus->in_gain < 0.0 )
{
lsx_fail("chorus: gain-in must be positive!");
return (SOX_EOF);
}
if ( chorus->in_gain > 1.0 )
{
lsx_fail("chorus: gain-in must be less than 1.0!");
return (SOX_EOF);
}
if ( chorus->out_gain < 0.0 )
{
lsx_fail("chorus: gain-out must be positive!");
return (SOX_EOF);
}
for ( i = 0; i < chorus->num_chorus; i++ ) {
chorus->samples[i] = (int) ( ( chorus->delay[i] +
chorus->depth[i] ) * effp->in_signal.rate / 1000.0);
chorus->depth_samples[i] = (int) (chorus->depth[i] *
effp->in_signal.rate / 1000.0);
if ( chorus->delay[i] < 20.0 )
{
lsx_fail("chorus: delay must be more than 20.0 msec!");
return (SOX_EOF);
}
if ( chorus->delay[i] > 100.0 )
{
lsx_fail("chorus: delay must be less than 100.0 msec!");
return (SOX_EOF);
}
if ( chorus->speed[i] < 0.1 )
{
lsx_fail("chorus: speed must be more than 0.1 Hz!");
return (SOX_EOF);
}
if ( chorus->speed[i] > 5.0 )
{
lsx_fail("chorus: speed must be less than 5.0 Hz!");
return (SOX_EOF);
}
if ( chorus->depth[i] < 0.0 )
{
lsx_fail("chorus: delay must be more positive!");
return (SOX_EOF);
}
if ( chorus->depth[i] > 10.0 )
{
lsx_fail("chorus: delay must be less than 10.0 msec!");
return (SOX_EOF);
}
if ( chorus->decay[i] < 0.0 )
{
lsx_fail("chorus: decay must be positive!" );
return (SOX_EOF);
}
if ( chorus->decay[i] > 1.0 )
{
lsx_fail("chorus: decay must be less that 1.0!" );
return (SOX_EOF);
}
chorus->length[i] = effp->in_signal.rate / chorus->speed[i];
chorus->lookup_tab[i] = lsx_malloc(sizeof (int) * chorus->length[i]);
if (chorus->modulation[i] == MOD_SINE)
lsx_generate_wave_table(SOX_WAVE_SINE, SOX_INT, chorus->lookup_tab[i],
(size_t)chorus->length[i], 0., (double)chorus->depth_samples[i], 0.);
else
lsx_generate_wave_table(SOX_WAVE_TRIANGLE, SOX_INT, chorus->lookup_tab[i],
(size_t)chorus->length[i],
(double)(chorus->samples[i] - 1 - 2 * chorus->depth_samples[i]),
(double)(chorus->samples[i] - 1), 3 * M_PI_2);
chorus->phase[i] = 0;
if ( chorus->samples[i] > chorus->maxsamples )
chorus->maxsamples = chorus->samples[i];
}
/* Be nice and check the hint with warning, if... */
sum_in_volume = 1.0;
for ( i = 0; i < chorus->num_chorus; i++ )
sum_in_volume += chorus->decay[i];
if ( chorus->in_gain * ( sum_in_volume ) > 1.0 / chorus->out_gain )
lsx_warn("chorus: warning >>> gain-out can cause saturation or clipping of output <<<");
chorus->chorusbuf = lsx_malloc(sizeof (float) * chorus->maxsamples);
for ( i = 0; i < chorus->maxsamples; i++ )
chorus->chorusbuf[i] = 0.0;
chorus->counter = 0;
chorus->fade_out = chorus->maxsamples;
effp->out_signal.length = SOX_UNKNOWN_LEN; /* TODO: calculate actual length */
return (SOX_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int sox_chorus_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * chorus = (priv_t *) effp->priv;
int i;
float d_in, d_out;
sox_sample_t out;
size_t len = min(*isamp, *osamp);
*isamp = *osamp = len;
while (len--) {
/* Store delays as 24-bit signed longs */
d_in = (float) *ibuf++ / 256;
/* Compute output first */
d_out = d_in * chorus->in_gain;
for ( i = 0; i < chorus->num_chorus; i++ )
d_out += chorus->chorusbuf[(chorus->maxsamples +
chorus->counter - chorus->lookup_tab[i][chorus->phase[i]]) %
chorus->maxsamples] * chorus->decay[i];
/* Adjust the output volume and size to 24 bit */
d_out = d_out * chorus->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Mix decay of delay and input */
chorus->chorusbuf[chorus->counter] = d_in;
chorus->counter =
( chorus->counter + 1 ) % chorus->maxsamples;
for ( i = 0; i < chorus->num_chorus; i++ )
chorus->phase[i] =
( chorus->phase[i] + 1 ) % chorus->length[i];
}
/* processed all samples */
return (SOX_SUCCESS);
}
/*
* Drain out reverb lines.
*/
static int sox_chorus_drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * chorus = (priv_t *) effp->priv;
size_t done;
int i;
float d_in, d_out;
sox_sample_t out;
done = 0;
while ( ( done < *osamp ) && ( done < chorus->fade_out ) ) {
d_in = 0;
d_out = 0;
/* Compute output first */
for ( i = 0; i < chorus->num_chorus; i++ )
d_out += chorus->chorusbuf[(chorus->maxsamples +
chorus->counter - chorus->lookup_tab[i][chorus->phase[i]]) %
chorus->maxsamples] * chorus->decay[i];
/* Adjust the output volume and size to 24 bit */
d_out = d_out * chorus->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Mix decay of delay and input */
chorus->chorusbuf[chorus->counter] = d_in;
chorus->counter =
( chorus->counter + 1 ) % chorus->maxsamples;
for ( i = 0; i < chorus->num_chorus; i++ )
chorus->phase[i] =
( chorus->phase[i] + 1 ) % chorus->length[i];
done++;
chorus->fade_out--;
}
/* samples played, it remains */
*osamp = done;
if (chorus->fade_out == 0)
return SOX_EOF;
else
return SOX_SUCCESS;
}
/*
* Clean up chorus effect.
*/
static int sox_chorus_stop(sox_effect_t * effp)
{
priv_t * chorus = (priv_t *) effp->priv;
int i;
free(chorus->chorusbuf);
chorus->chorusbuf = NULL;
for ( i = 0; i < chorus->num_chorus; i++ ) {
free(chorus->lookup_tab[i]);
chorus->lookup_tab[i] = NULL;
}
return (SOX_SUCCESS);
}
static sox_effect_handler_t sox_chorus_effect = {
"chorus",
"gain-in gain-out delay decay speed depth [ -s | -t ]",
SOX_EFF_LENGTH | SOX_EFF_GAIN,
sox_chorus_getopts,
sox_chorus_start,
sox_chorus_flow,
sox_chorus_drain,
sox_chorus_stop,
NULL, sizeof(priv_t)
};
const sox_effect_handler_t *lsx_chorus_effect_fn(void)
{
return &sox_chorus_effect;
}

293
app/src/main/cpp/sox/compand.c
Unescape View File

@ -0,0 +1,293 @@
/* libSoX compander effect
*
* Written by Nick Bailey (nick@bailey-family.org.uk or
* n.bailey@elec.gla.ac.uk)
*
* Copyright 1999 Chris Bagwell And Nick Bailey
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Chris Bagwell And Nick Bailey are not responsible for
* the consequences of using this software.
*/
#include "sox_i.h"
#include <string.h>
#include <stdlib.h>
#include "compandt.h"
/*
* Compressor/expander effect for libSoX.
*
* Flow diagram for one channel:
*
* ------------ ---------------
* | | | | ---
* ibuff ---+---| integrator |--->| transfer func |--->| |
* | | | | | | |
* | ------------ --------------- | | * gain
* | | * |----------->obuff
* | ------- | |
* | | | | |
* +----->| delay |-------------------------->| |
* | | ---
* -------
*/
#define compand_usage \
"attack1,decay1{,attack2,decay2} [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2} [gain [initial-volume-dB [delay]]]\n" \
"\twhere {} means optional and repeatable and [] means optional.\n" \
"\tdB values are floating point or -inf'; times are in seconds."
/*
* Note: clipping can occur if the transfer function pushes things too
* close to 0 dB. In that case, use a negative gain, or reduce the
* output level of the transfer function.
*/
typedef struct {
sox_compandt_t transfer_fn;
struct {
double attack_times[2]; /* 0:attack_time, 1:decay_time */
double volume; /* Current "volume" of each channel */
} * channels;
unsigned expectedChannels;/* Also flags that channels aren't to be treated
individually when = 1 and input not mono */
double delay; /* Delay to apply before companding */
sox_sample_t *delay_buf; /* Old samples, used for delay processing */
ptrdiff_t delay_buf_size;/* Size of delay_buf in samples */
ptrdiff_t delay_buf_index; /* Index into delay_buf */
ptrdiff_t delay_buf_cnt; /* No. of active entries in delay_buf */
int delay_buf_full; /* Shows buffer situation (important for drain) */
char *arg0; /* copies of arguments, so that they may be modified */
char *arg1;
char *arg2;
} priv_t;
static int getopts(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * l = (priv_t *) effp->priv;
char * s;
char dummy; /* To check for extraneous chars. */
unsigned pairs, i, j, commas;
--argc, ++argv;
if (argc < 2 || argc > 5)
return lsx_usage(effp);
l->arg0 = lsx_strdup(argv[0]);
l->arg1 = lsx_strdup(argv[1]);
l->arg2 = argc > 2 ? lsx_strdup(argv[2]) : NULL;
/* Start by checking the attack and decay rates */
for (s = l->arg0, commas = 0; *s; ++s) if (*s == ',') ++commas;
if ((commas % 2) == 0) {
lsx_fail("there must be an even number of attack/decay parameters");
return SOX_EOF;
}
pairs = 1 + commas/2;
l->channels = lsx_calloc(pairs, sizeof(*l->channels));
l->expectedChannels = pairs;
/* Now tokenise the rates string and set up these arrays. Keep
them in seconds at the moment: we don't know the sample rate yet. */
for (i = 0, s = strtok(l->arg0, ","); s != NULL; ++i) {
for (j = 0; j < 2; ++j) {
if (sscanf(s, "%lf %c", &l->channels[i].attack_times[j], &dummy) != 1) {
lsx_fail("syntax error trying to read attack/decay time");
return SOX_EOF;
} else if (l->channels[i].attack_times[j] < 0) {
lsx_fail("attack & decay times can't be less than 0 seconds");
return SOX_EOF;
}
s = strtok(NULL, ",");
}
}
if (!lsx_compandt_parse(&l->transfer_fn, l->arg1, l->arg2))
return SOX_EOF;
/* Set the initial "volume" to be attibuted to the input channels.
Unless specified, choose 0dB otherwise clipping will
result if the user has seleced a long attack time */
for (i = 0; i < l->expectedChannels; ++i) {
double init_vol_dB = 0;
if (argc > 3 && sscanf(argv[3], "%lf %c", &init_vol_dB, &dummy) != 1) {
lsx_fail("syntax error trying to read initial volume");
return SOX_EOF;
} else if (init_vol_dB > 0) {
lsx_fail("initial volume is relative to maximum volume so can't exceed 0dB");
return SOX_EOF;
}
l->channels[i].volume = pow(10., init_vol_dB / 20);
}
/* If there is a delay, store it. */
if (argc > 4 && sscanf(argv[4], "%lf %c", &l->delay, &dummy) != 1) {
lsx_fail("syntax error trying to read delay value");
return SOX_EOF;
} else if (l->delay < 0) {
lsx_fail("delay can't be less than 0 seconds");
return SOX_EOF;
}
return SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * l = (priv_t *) effp->priv;
unsigned i, j;
lsx_debug("%i input channel(s) expected: actually %i",
l->expectedChannels, effp->out_signal.channels);
for (i = 0; i < l->expectedChannels; ++i)
lsx_debug("Channel %i: attack = %g decay = %g", i,
l->channels[i].attack_times[0], l->channels[i].attack_times[1]);
if (!lsx_compandt_show(&l->transfer_fn, effp->global_info->plot))
return SOX_EOF;
/* Convert attack and decay rates using number of samples */
for (i = 0; i < l->expectedChannels; ++i)
for (j = 0; j < 2; ++j)
if (l->channels[i].attack_times[j] > 1.0/effp->out_signal.rate)
l->channels[i].attack_times[j] = 1.0 -
exp(-1.0/(effp->out_signal.rate * l->channels[i].attack_times[j]));
else
l->channels[i].attack_times[j] = 1.0;
/* Allocate the delay buffer */
l->delay_buf_size = l->delay * effp->out_signal.rate * effp->out_signal.channels;
if (l->delay_buf_size > 0)
l->delay_buf = lsx_calloc((size_t)l->delay_buf_size, sizeof(*l->delay_buf));
l->delay_buf_index = 0;
l->delay_buf_cnt = 0;
l->delay_buf_full= 0;
return SOX_SUCCESS;
}
/*
* Update a volume value using the given sample
* value, the attack rate and decay rate
*/
static void doVolume(double *v, double samp, priv_t * l, int chan)
{
double s = -samp / SOX_SAMPLE_MIN;
double delta = s - *v;
if (delta > 0.0) /* increase volume according to attack rate */
*v += delta * l->channels[chan].attack_times[0];
else /* reduce volume according to decay rate */
*v += delta * l->channels[chan].attack_times[1];
}
static int flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * l = (priv_t *) effp->priv;
int len = (*isamp > *osamp) ? *osamp : *isamp;
int filechans = effp->out_signal.channels;
int idone,odone;
for (idone = 0,odone = 0; idone < len; ibuf += filechans) {
int chan;
/* Maintain the volume fields by simulating a leaky pump circuit */
for (chan = 0; chan < filechans; ++chan) {
if (l->expectedChannels == 1 && filechans > 1) {
/* User is expecting same compander for all channels */
int i;
double maxsamp = 0.0;
for (i = 0; i < filechans; ++i) {
double rect = fabs((double)ibuf[i]);
if (rect > maxsamp) maxsamp = rect;
}
doVolume(&l->channels[0].volume, maxsamp, l, 0);
break;
} else
doVolume(&l->channels[chan].volume, fabs((double)ibuf[chan]), l, chan);
}
/* Volume memory is updated: perform compand */
for (chan = 0; chan < filechans; ++chan) {
int ch = l->expectedChannels > 1 ? chan : 0;
double level_in_lin = l->channels[ch].volume;
double level_out_lin = lsx_compandt(&l->transfer_fn, level_in_lin);
double checkbuf;
if (l->delay_buf_size <= 0) {
checkbuf = ibuf[chan] * level_out_lin;
SOX_SAMPLE_CLIP_COUNT(checkbuf, effp->clips);
obuf[odone++] = checkbuf;
idone++;
} else {
if (l->delay_buf_cnt >= l->delay_buf_size) {
l->delay_buf_full=1; /* delay buffer is now definitely full */
checkbuf = l->delay_buf[l->delay_buf_index] * level_out_lin;
SOX_SAMPLE_CLIP_COUNT(checkbuf, effp->clips);
obuf[odone] = checkbuf;
odone++;
idone++;
} else {
l->delay_buf_cnt++;
idone++; /* no "odone++" because we did not fill obuf[...] */
}
l->delay_buf[l->delay_buf_index++] = ibuf[chan];
l->delay_buf_index %= l->delay_buf_size;
}
}
}
*isamp = idone; *osamp = odone;
return (SOX_SUCCESS);
}
static int drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * l = (priv_t *) effp->priv;
size_t chan, done = 0;
if (l->delay_buf_full == 0)
l->delay_buf_index = 0;
while (done+effp->out_signal.channels <= *osamp && l->delay_buf_cnt > 0)
for (chan = 0; chan < effp->out_signal.channels; ++chan) {
int c = l->expectedChannels > 1 ? chan : 0;
double level_in_lin = l->channels[c].volume;
double level_out_lin = lsx_compandt(&l->transfer_fn, level_in_lin);
obuf[done++] = l->delay_buf[l->delay_buf_index++] * level_out_lin;
l->delay_buf_index %= l->delay_buf_size;
l->delay_buf_cnt--;
}
*osamp = done;
return l->delay_buf_cnt > 0 ? SOX_SUCCESS : SOX_EOF;
}
static int stop(sox_effect_t * effp)
{
priv_t * l = (priv_t *) effp->priv;
free(l->delay_buf);
return SOX_SUCCESS;
}
static int lsx_kill(sox_effect_t * effp)
{
priv_t * l = (priv_t *) effp->priv;
lsx_compandt_kill(&l->transfer_fn);
free(l->channels);
free(l->arg0);
free(l->arg1);
free(l->arg2);
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_compand_effect_fn(void)
{
static sox_effect_handler_t handler = {
"compand", compand_usage, SOX_EFF_MCHAN | SOX_EFF_GAIN,
getopts, start, flow, drain, stop, lsx_kill, sizeof(priv_t)
};
return &handler;
}

229
app/src/main/cpp/sox/compandt.c
Unescape View File

@ -0,0 +1,229 @@
/* libSoX Compander Transfer Function: (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "compandt.h"
#include <string.h>
#define LOG_TO_LOG10(x) ((x) * 20 / M_LN10)
sox_bool lsx_compandt_show(sox_compandt_t * t, sox_plot_t plot)
{
int i;
for (i = 1; t->segments[i-1].x; ++i)
lsx_debug("TF: %g %g %g %g",
LOG_TO_LOG10(t->segments[i].x),
LOG_TO_LOG10(t->segments[i].y),
LOG_TO_LOG10(t->segments[i].a),
LOG_TO_LOG10(t->segments[i].b));
if (plot == sox_plot_octave) {
printf(
"%% GNU Octave file (may also work with MATLAB(R) )\n"
"in=linspace(-99.5,0,200);\n"
"out=[");
for (i = -199; i <= 0; ++i) {
double in = i/2.;
double in_lin = pow(10., in/20);
printf("%g ", in + 20 * log10(lsx_compandt(t, in_lin)));
}
printf(
"];\n"
"plot(in,out)\n"
"title('SoX effect: compand')\n"
"xlabel('Input level (dB)')\n"
"ylabel('Output level (dB)')\n"
"grid on\n"
"disp('Hit return to continue')\n"
"pause\n");
return sox_false;
}
if (plot == sox_plot_gnuplot) {
printf(
"# gnuplot file\n"
"set title 'SoX effect: compand'\n"
"set xlabel 'Input level (dB)'\n"
"set ylabel 'Output level (dB)'\n"
"set grid xtics ytics\n"
"set key off\n"
"plot '-' with lines\n");
for (i = -199; i <= 0; ++i) {
double in = i/2.;
double in_lin = pow(10., in/20);
printf("%g %g\n", in, in + 20 * log10(lsx_compandt(t, in_lin)));
}
printf(
"e\n"
"pause -1 'Hit return to continue'\n");
return sox_false;
}
return sox_true;
}
static void prepare_transfer_fn(sox_compandt_t * t)
{
int i;
double radius = t->curve_dB * M_LN10 / 20;
for (i = 0; !i || t->segments[i-2].x; i += 2) {
t->segments[i].y += t->outgain_dB;
t->segments[i].x *= M_LN10 / 20; /* Convert to natural logs */
t->segments[i].y *= M_LN10 / 20;
}
#define line1 t->segments[i - 4]
#define curve t->segments[i - 3]
#define line2 t->segments[i - 2]
#define line3 t->segments[i - 0]
for (i = 4; t->segments[i - 2].x; i += 2) {
double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
line1.a = 0;
line1.b = (line2.y - line1.y) / (line2.x - line1.x);
line2.a = 0;
line2.b = (line3.y - line2.y) / (line3.x - line2.x);
theta = atan2(line2.y - line1.y, line2.x - line1.x);
len = sqrt(pow(line2.x - line1.x, 2.) + pow(line2.y - line1.y, 2.));
r = min(radius, len);
curve.x = line2.x - r * cos(theta);
curve.y = line2.y - r * sin(theta);
theta = atan2(line3.y - line2.y, line3.x - line2.x);
len = sqrt(pow(line3.x - line2.x, 2.) + pow(line3.y - line2.y, 2.));
r = min(radius, len / 2);
x = line2.x + r * cos(theta);
y = line2.y + r * sin(theta);
cx = (curve.x + line2.x + x) / 3;
cy = (curve.y + line2.y + y) / 3;
line2.x = x;
line2.y = y;
in1 = cx - curve.x;
out1 = cy - curve.y;
in2 = line2.x - curve.x;
out2 = line2.y - curve.y;
curve.a = (out2/in2 - out1/in1) / (in2-in1);
curve.b = out1/in1 - curve.a*in1;
}
#undef line1
#undef curve
#undef line2
#undef line3
t->segments[i - 3].x = 0;
t->segments[i - 3].y = t->segments[i - 2].y;
t->in_min_lin = exp(t->segments[1].x);
t->out_min_lin= exp(t->segments[1].y);
}
static sox_bool parse_transfer_value(char const * text, double * value)
{
char dummy; /* To check for extraneous chars. */
if (!text) {
lsx_fail("syntax error trying to read transfer function value");
return sox_false;
}
if (!strcmp(text, "-inf"))
*value = -20 * log10(-(double)SOX_SAMPLE_MIN);
else if (sscanf(text, "%lf %c", value, &dummy) != 1) {
lsx_fail("syntax error trying to read transfer function value");
return sox_false;
}
else if (*value > 0) {
lsx_fail("transfer function values are relative to maximum volume so can't exceed 0dB");
return sox_false;
}
return sox_true;
}
sox_bool lsx_compandt_parse(sox_compandt_t * t, char * points, char * gain)
{
char const * text = points;
unsigned i, j, num, pairs, commas = 0;
char dummy; /* To check for extraneous chars. */
if (sscanf(points, "%lf %c", &t->curve_dB, &dummy) == 2 && dummy == ':')
points = strchr(points, ':') + 1;
else t->curve_dB = 0;
t->curve_dB = max(t->curve_dB, .01);
while (*text) commas += *text++ == ',';
pairs = 1 + commas / 2;
++pairs; /* allow room for extra pair at the beginning */
pairs *= 2; /* allow room for the auto-curves */
++pairs; /* allow room for 0,0 at end */
t->segments = lsx_calloc(pairs, sizeof(*t->segments));
#define s(n) t->segments[2*((n)+1)]
for (i = 0, text = strtok(points, ","); text != NULL; ++i) {
if (!parse_transfer_value(text, &s(i).x))
return sox_false;
if (i && s(i-1).x > s(i).x) {
lsx_fail("transfer function input values must be strictly increasing");
return sox_false;
}
if (i || (commas & 1)) {
text = strtok(NULL, ",");
if (!parse_transfer_value(text, &s(i).y))
return sox_false;
s(i).y -= s(i).x;
}
text = strtok(NULL, ",");
}
num = i;
if (num == 0 || s(num-1).x) /* Add 0,0 if necessary */
++num;
#undef s
if (gain && sscanf(gain, "%lf %c", &t->outgain_dB, &dummy) != 1) {
lsx_fail("syntax error trying to read post-processing gain value");
return sox_false;
}
#define s(n) t->segments[2*(n)]
s(0).x = s(1).x - 2 * t->curve_dB; /* Add a tail off segment at the start */
s(0).y = s(1).y;
++num;
for (i = 2; i < num; ++i) { /* Join adjacent colinear segments */
double g1 = (s(i-1).y - s(i-2).y) * (s(i-0).x - s(i-1).x);
double g2 = (s(i-0).y - s(i-1).y) * (s(i-1).x - s(i-2).x);
if (fabs(g1 - g2)) /* fabs stops epsilon problems */
continue;
--num;
for (j = --i; j < num; ++j)
s(j) = s(j+1);
}
#undef s
prepare_transfer_fn(t);
return sox_true;
}
void lsx_compandt_kill(sox_compandt_t * p)
{
free(p->segments);
}

52
app/src/main/cpp/sox/compandt.h
Unescape View File

@ -0,0 +1,52 @@
/* libSoX Compander Transfer Function (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
typedef struct {
struct sox_compandt_segment {
double x, y; /* 1st point in segment */
double a, b; /* Quadratic coeffecients for rest of segment */
} * segments;
double in_min_lin;
double out_min_lin;
double outgain_dB; /* Post processor gain */
double curve_dB;
} sox_compandt_t;
sox_bool lsx_compandt_parse(sox_compandt_t * t, char * points, char * gain);
sox_bool lsx_compandt_show(sox_compandt_t * t, sox_plot_t plot);
void lsx_compandt_kill(sox_compandt_t * p);
/* Place in header to allow in-lining */
static double lsx_compandt(sox_compandt_t * t, double in_lin)
{
struct sox_compandt_segment * s;
double in_log, out_log;
if (in_lin <= t->in_min_lin)
return t->out_min_lin;
in_log = log(in_lin);
for (s = t->segments + 1; in_log > s[1].x; ++s);
in_log -= s->x;
out_log = s->y + in_log * (s->a * in_log + s->b);
return exp(out_log);
}

49
app/src/main/cpp/sox/contrast.c
Unescape View File

@ -0,0 +1,49 @@
/* libSoX effect: Contrast Enhancement (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
typedef struct {double contrast;} priv_t;
static int create(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
p->contrast = 75;
--argc, ++argv;
do {NUMERIC_PARAMETER(contrast, 0, 100)} while (0);
p->contrast /= 750; /* shift range to 0 to 0.1333, default 0.1 */
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len = *isamp = *osamp = min(*isamp, *osamp);
while (len--) {
double d = *ibuf++ * (-M_PI_2 / SOX_SAMPLE_MIN);
*obuf++ = sin(d + p->contrast * sin(d * 4)) * SOX_SAMPLE_MAX;
}
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_contrast_effect_fn(void)
{
static sox_effect_handler_t handler = {"contrast", "[enhancement (75)]",
0, create, NULL, flow, NULL, NULL, NULL, sizeof(priv_t)};
return &handler;
}

418
app/src/main/cpp/sox/coreaudio.c
Unescape View File

@ -0,0 +1,418 @@
/* AudioCore sound handler
*
* Copyright 2008 Chris Bagwell And Sundry Contributors
*/
#include "sox_i.h"
#include <CoreAudio/CoreAudio.h>
#include <pthread.h>
#define Buffactor 4
typedef struct {
AudioDeviceID adid;
pthread_mutex_t mutex;
pthread_cond_t cond;
int device_started;
size_t bufsize;
size_t bufrd;
size_t bufwr;
size_t bufrdavail;
float *buf;
} priv_t;
static OSStatus PlaybackIOProc(AudioDeviceID inDevice UNUSED,
const AudioTimeStamp *inNow UNUSED,
const AudioBufferList *inInputData UNUSED,
const AudioTimeStamp *inInputTime UNUSED,
AudioBufferList *outOutputData,
const AudioTimeStamp *inOutputTime UNUSED,
void *inClientData)
{
priv_t *ac = (priv_t*)((sox_format_t*)inClientData)->priv;
AudioBuffer *buf;
size_t copylen, avail;
pthread_mutex_lock(&ac->mutex);
for(buf = outOutputData->mBuffers;
buf != outOutputData->mBuffers + outOutputData->mNumberBuffers;
buf++){
copylen = buf->mDataByteSize / sizeof(float);
if(copylen > ac->bufrdavail)
copylen = ac->bufrdavail;
avail = ac->bufsize - ac->bufrd;
if(buf->mData == NULL){
/*do nothing-hardware can't play audio*/
}else if(copylen > avail){
memcpy(buf->mData, ac->buf + ac->bufrd, avail * sizeof(float));
memcpy((float*)buf->mData + avail, ac->buf, (copylen - avail) * sizeof(float));
}else{
memcpy(buf->mData, ac->buf + ac->bufrd, copylen * sizeof(float));
}
buf->mDataByteSize = copylen * sizeof(float);
ac->bufrd += copylen;
if(ac->bufrd >= ac->bufsize)
ac->bufrd -= ac->bufsize;
ac->bufrdavail -= copylen;
}
pthread_cond_signal(&ac->cond);
pthread_mutex_unlock(&ac->mutex);
return kAudioHardwareNoError;
}
static OSStatus RecIOProc(AudioDeviceID inDevice UNUSED,
const AudioTimeStamp *inNow UNUSED,
const AudioBufferList *inInputData,
const AudioTimeStamp *inInputTime UNUSED,
AudioBufferList *outOutputData UNUSED,
const AudioTimeStamp *inOutputTime UNUSED,
void *inClientData)
{
priv_t *ac = (priv_t *)((sox_format_t*)inClientData)->priv;
AudioBuffer const *buf;
size_t nfree, copylen, avail;
pthread_mutex_lock(&ac->mutex);
for(buf = inInputData->mBuffers;
buf != inInputData->mBuffers + inInputData->mNumberBuffers;
buf++){
if(buf->mData == NULL)
continue;
copylen = buf->mDataByteSize / sizeof(float);
nfree = ac->bufsize - ac->bufrdavail - 1;
if(nfree == 0)
lsx_warn("coreaudio: unhandled buffer overrun. Data discarded.");
if(copylen > nfree)
copylen = nfree;
avail = ac->bufsize - ac->bufwr;
if(copylen > avail){
memcpy(ac->buf + ac->bufwr, buf->mData, avail * sizeof(float));
memcpy(ac->buf, (float*)buf->mData + avail, (copylen - avail) * sizeof(float));
}else{
memcpy(ac->buf + ac->bufwr, buf->mData, copylen * sizeof(float));
}
ac->bufwr += copylen;
if(ac->bufwr >= ac->bufsize)
ac->bufwr -= ac->bufsize;
ac->bufrdavail += copylen;
}
pthread_cond_signal(&ac->cond);
pthread_mutex_unlock(&ac->mutex);
return kAudioHardwareNoError;
}
static int setup(sox_format_t *ft, int is_input)
{
priv_t *ac = (priv_t *)ft->priv;
OSStatus status;
UInt32 property_size;
struct AudioStreamBasicDescription stream_desc;
int32_t buf_size;
int rc;
if (strncmp(ft->filename, "default", (size_t)7) == 0)
{
property_size = sizeof(ac->adid);
if (is_input)
status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &property_size, &ac->adid);
else
status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &property_size, &ac->adid);
}
else
{
Boolean is_writable;
status = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &property_size, &is_writable);
if (status == noErr)
{
int device_count = property_size/sizeof(AudioDeviceID);
AudioDeviceID *devices;
devices = malloc(property_size);
status = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &property_size, devices);
if (status == noErr)
{
int i;
for (i = 0; i < device_count; i++)
{
char name[256];
status = AudioDeviceGetProperty(devices[i],0,false,kAudioDevicePropertyDeviceName,&property_size,&name);
lsx_report("Found Audio Device \"%s\"\n",name);
/* String returned from OS is truncated so only compare
* as much as returned.
*/
if (strncmp(name,ft->filename,strlen(name)) == 0)
{
ac->adid = devices[i];
break;
}
}
}
free(devices);
}
}
if (status || ac->adid == kAudioDeviceUnknown)
{
lsx_fail_errno(ft, SOX_EPERM, "can not open audio device");
return SOX_EOF;
}
/* Query device to get initial values */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceGetProperty(ac->adid, 0, is_input,
kAudioDevicePropertyStreamFormat,
&property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not get audio device properties");
return SOX_EOF;
}
if (!(stream_desc.mFormatFlags & kLinearPCMFormatFlagIsFloat))
{
lsx_fail_errno(ft, SOX_EPERM, "audio device does not accept floats");
return SOX_EOF;
}
/* OS X effectively only supports these values. */
ft->signal.channels = 2;
ft->signal.rate = 44100;
ft->encoding.bits_per_sample = 32;
/* TODO: My limited experience with hardware can only get floats working
* withh a fixed sample rate and stereo. I know that is a limitiation of
* audio device I have so this may not be standard operating orders.
* If some hardware supports setting sample rates and channel counts
* then should do that over resampling and mixing.
*/
#if 0
stream_desc.mSampleRate = ft->signal.rate;
stream_desc.mChannelsPerFrame = ft->signal.channels;
/* Write them back */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceSetProperty(ac->adid, NULL, 0, is_input,
kAudioDevicePropertyStreamFormat,
property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not set audio device properties");
return SOX_EOF;
}
/* Query device to see if it worked */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceGetProperty(ac->adid, 0, is_input,
kAudioDevicePropertyStreamFormat,
&property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not get audio device properties");
return SOX_EOF;
}
#endif
if (stream_desc.mChannelsPerFrame != ft->signal.channels)
{
lsx_debug("audio device did not accept %d channels. Use %d channels instead.", (int)ft->signal.channels,
(int)stream_desc.mChannelsPerFrame);
ft->signal.channels = stream_desc.mChannelsPerFrame;
}
if (stream_desc.mSampleRate != ft->signal.rate)
{
lsx_debug("audio device did not accept %d sample rate. Use %d instead.", (int)ft->signal.rate,
(int)stream_desc.mSampleRate);
ft->signal.rate = stream_desc.mSampleRate;
}
ac->bufsize = sox_globals.bufsiz / sizeof(sox_sample_t) * Buffactor;
ac->bufrd = 0;
ac->bufwr = 0;
ac->bufrdavail = 0;
ac->buf = lsx_malloc(ac->bufsize * sizeof(float));
buf_size = sox_globals.bufsiz / sizeof(sox_sample_t) * sizeof(float);
property_size = sizeof(buf_size);
status = AudioDeviceSetProperty(ac->adid, NULL, 0, is_input,
kAudioDevicePropertyBufferSize,
property_size, &buf_size);
rc = pthread_mutex_init(&ac->mutex, NULL);
if (rc)
{
lsx_fail_errno(ft, SOX_EPERM, "failed initializing mutex");
return SOX_EOF;
}
rc = pthread_cond_init(&ac->cond, NULL);
if (rc)
{
lsx_fail_errno(ft, SOX_EPERM, "failed initializing condition");
return SOX_EOF;
}
ac->device_started = 0;
/* Registers callback with the device without activating it. */
if (is_input)
status = AudioDeviceAddIOProc(ac->adid, RecIOProc, (void *)ft);
else
status = AudioDeviceAddIOProc(ac->adid, PlaybackIOProc, (void *)ft);
return SOX_SUCCESS;
}
static int startread(sox_format_t *ft)
{
return setup(ft, 1);
}
static size_t read_samples(sox_format_t *ft, sox_sample_t *buf, size_t nsamp)
{
priv_t *ac = (priv_t *)ft->priv;
size_t len;
SOX_SAMPLE_LOCALS;
if (!ac->device_started) {
AudioDeviceStart(ac->adid, RecIOProc);
ac->device_started = 1;
}
pthread_mutex_lock(&ac->mutex);
/* Wait until input buffer has been filled by device driver */
while (ac->bufrdavail == 0)
pthread_cond_wait(&ac->cond, &ac->mutex);
len = 0;
while(len < nsamp && ac->bufrdavail > 0){
buf[len] = SOX_FLOAT_32BIT_TO_SAMPLE(ac->buf[ac->bufrd], ft->clips);
len++;
ac->bufrd++;
if(ac->bufrd == ac->bufsize)
ac->bufrd = 0;
ac->bufrdavail--;
}
pthread_mutex_unlock(&ac->mutex);
return len;
}
static int stopread(sox_format_t * ft)
{
priv_t *ac = (priv_t *)ft->priv;
AudioDeviceStop(ac->adid, RecIOProc);
AudioDeviceRemoveIOProc(ac->adid, RecIOProc);
pthread_cond_destroy(&ac->cond);
pthread_mutex_destroy(&ac->mutex);
free(ac->buf);
return SOX_SUCCESS;
}
static int startwrite(sox_format_t * ft)
{
return setup(ft, 0);
}
static size_t write_samples(sox_format_t *ft, const sox_sample_t *buf, size_t nsamp)
{
priv_t *ac = (priv_t *)ft->priv;
size_t i;
SOX_SAMPLE_LOCALS;
pthread_mutex_lock(&ac->mutex);
/* Wait to start until mutex is locked to help prevent callback
* getting zero samples.
*/
if(!ac->device_started){
if(AudioDeviceStart(ac->adid, PlaybackIOProc)){
pthread_mutex_unlock(&ac->mutex);
return SOX_EOF;
}
ac->device_started = 1;
}
/* globals.bufsize is in samples
* buf_offset is in bytes
* buf_size is in bytes
*/
for(i = 0; i < nsamp; i++){
while(ac->bufrdavail == ac->bufsize - 1)
pthread_cond_wait(&ac->cond, &ac->mutex);
ac->buf[ac->bufwr] = SOX_SAMPLE_TO_FLOAT_32BIT(buf[i], ft->clips);
ac->bufwr++;
if(ac->bufwr == ac->bufsize)
ac->bufwr = 0;
ac->bufrdavail++;
}
pthread_mutex_unlock(&ac->mutex);
return nsamp;
}
static int stopwrite(sox_format_t * ft)
{
priv_t *ac = (priv_t *)ft->priv;
if(ac->device_started){
pthread_mutex_lock(&ac->mutex);
while (ac->bufrdavail > 0)
pthread_cond_wait(&ac->cond, &ac->mutex);
pthread_mutex_unlock(&ac->mutex);
AudioDeviceStop(ac->adid, PlaybackIOProc);
}
AudioDeviceRemoveIOProc(ac->adid, PlaybackIOProc);
pthread_cond_destroy(&ac->cond);
pthread_mutex_destroy(&ac->mutex);
free(ac->buf);
return SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(coreaudio)
{
static char const *const names[] = { "coreaudio", NULL };
static unsigned const write_encodings[] = {
SOX_ENCODING_FLOAT, 32, 0,
0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Mac AudioCore device driver",
names, SOX_FILE_DEVICE | SOX_FILE_NOSTDIO,
startread, read_samples, stopread,
startwrite, write_samples, stopwrite,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

114
app/src/main/cpp/sox/cvsd-fmt.c
Unescape View File

@ -0,0 +1,114 @@
/* libSoX file format: CVSD (see cvsd.c) (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "cvsd.h"
LSX_FORMAT_HANDLER(cvsd)
{
static char const * const names[] = {"cvsd", "cvs", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_CVSD, 1, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Headerless MIL Std 188 113 Continuously Variable Slope Delta modulation",
names, SOX_FILE_MONO,
lsx_cvsdstartread, lsx_cvsdread, lsx_cvsdstopread,
lsx_cvsdstartwrite, lsx_cvsdwrite, lsx_cvsdstopwrite,
lsx_rawseek, write_encodings, NULL, sizeof(cvsd_priv_t)
};
return &handler;
}
/* libSoX file format: CVU (c) 2008 robs@users.sourceforge.net
* Unfiltered, therefore, on decode, use with either filter -4k or rate 8k */
typedef struct {
double sample, step, step_mult, step_add;
unsigned last_n_bits;
unsigned char byte;
off_t bit_count;
} priv_t;
static int start(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
ft->signal.channels = 1;
lsx_rawstart(ft, sox_true, sox_false, sox_true, SOX_ENCODING_CVSD, 1);
p->last_n_bits = 5; /* 101 */
p->step_mult = exp((-1 / .005 / ft->signal.rate));
p->step_add = (1 - p->step_mult) * (.1 * SOX_SAMPLE_MAX);
lsx_debug("step_mult=%g step_add=%f", p->step_mult, p->step_add);
return SOX_SUCCESS;
}
static void decode(priv_t * p, int bit)
{
p->last_n_bits = ((p->last_n_bits << 1) | bit) & 7;
p->step *= p->step_mult;
if (p->last_n_bits == 0 || p->last_n_bits == 7)
p->step += p->step_add;
if (p->last_n_bits & 1)
p->sample = min(p->step_mult * p->sample + p->step, SOX_SAMPLE_MAX);
else
p->sample = max(p->step_mult * p->sample - p->step, SOX_SAMPLE_MIN);
}
static size_t cvsdread(sox_format_t * ft, sox_sample_t * buf, size_t len)
{
priv_t *p = (priv_t *) ft->priv;
size_t i;
for (i = 0; i < len; ++i) {
if (!(p->bit_count & 7))
if (lsx_read_b_buf(ft, &p->byte, (size_t)1) != 1)
break;
++p->bit_count;
decode(p, p->byte & 1);
p->byte >>= 1;
*buf++ = floor(p->sample + .5);
}
return i;
}
static size_t cvsdwrite(sox_format_t * ft, sox_sample_t const * buf, size_t len)
{
priv_t *p = (priv_t *) ft->priv;
size_t i;
for (i = 0; i < len; ++i) {
decode(p, *buf++ > p->sample);
p->byte >>= 1;
p->byte |= p->last_n_bits << 7;
if (!(++p->bit_count & 7))
if (lsx_writeb(ft, p->byte) != SOX_SUCCESS)
break;
}
return len;
}
LSX_FORMAT_HANDLER(cvu)
{
static char const * const names[] = {"cvu", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_CVSD, 1, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Headerless Continuously Variable Slope Delta modulation (unfiltered)",
names, SOX_FILE_MONO, start, cvsdread, NULL, start, cvsdwrite, NULL,
lsx_rawseek, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

680
app/src/main/cpp/sox/cvsd.c
Unescape View File

@ -0,0 +1,680 @@
/* libSoX CVSD (Continuously Variable Slope Delta modulation)
* conversion routines
*
* The CVSD format is described in the MIL Std 188 113, which is
* available from http://bbs.itsi.disa.mil:5580/T3564
*
* Copyright (C) 1996
* Thomas Sailer (sailer@ife.ee.ethz.ch) (HB9JNX/AE4WA)
* Swiss Federal Institute of Technology, Electronics Lab
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
* Change History:
*
* June 1, 1998 - Chris Bagwell (cbagwell@sprynet.com)
* Fixed compile warnings reported by Kjetil Torgrim Homme
* <kjetilho@ifi.uio.no>
*
* June 20, 2006 - Kimberly Rockwell (pyxis13317 (at) yahoo.com)
* Speed optimization: Unrolled float_conv() loop in seperate
* functions for encoding and decoding. 15% speed up decoding.
*
* Aug. 24, 2009 - P. Chaintreuil (sox-cvsd-peep (at) parallaxshift.com)
* Speed optimization: Replaced calls to memmove() with a
* mirrored circular buffer. This doubles the size of the
* dec.output_filter (48 -> 96 floats) and enc.input_filter
* (16 -> 32 floats), but keeps the memory from having to
* be copied so many times. 56% speed increase decoding;
* less than 5% encoding speed increase.
*/
#include "sox_i.h"
#include "cvsd.h"
#include "cvsdfilt.h"
#include <string.h>
#include <time.h>
/* ---------------------------------------------------------------------- */
/*
* private data structures
*/
typedef cvsd_priv_t priv_t;
static int debug_count = 0;
/* ---------------------------------------------------------------------- */
/* This float_conv() function is not used as more specialized/optimized
* versions exist below. However, those new versions are tied to
* very percise filters defined in cvsdfilt.h. If those are modified
* or different filters are found to be required, this function may
* be needed. Thus I leave it here for possible future use, but commented
* out to avoid compiler warnings about it not being used.
static float float_conv(float const *fp1, float const *fp2,int n)
{
float res = 0;
for(; n > 0; n--)
res += (*fp1++) * (*fp2++);
return res;
}
*/
static float float_conv_enc(float const *fp1, float const *fp2)
{
/* This is a specialzed version of float_conv() for encoding
* which simply assumes a CVSD_ENC_FILTERLEN (16) length of
* the two arrays and unrolls that loop.
*
* fp1 should be the enc.input_filter array and must be
* CVSD_ENC_FILTERLEN (16) long.
*
* fp2 should be one of the enc_filter_xx_y() tables listed
* in cvsdfilt.h. At minimum, fp2 must be CVSD_ENC_FILTERLEN
* (16) entries long.
*/
float res = 0;
/* unrolling loop */
res += fp1[0] * fp2[0];
res += fp1[1] * fp2[1];
res += fp1[2] * fp2[2];
res += fp1[3] * fp2[3];
res += fp1[4] * fp2[4];
res += fp1[5] * fp2[5];
res += fp1[6] * fp2[6];
res += fp1[7] * fp2[7];
res += fp1[8] * fp2[8];
res += fp1[9] * fp2[9];
res += fp1[10] * fp2[10];
res += fp1[11] * fp2[11];
res += fp1[12] * fp2[12];
res += fp1[13] * fp2[13];
res += fp1[14] * fp2[14];
res += fp1[15] * fp2[15];
return res;
}
static float float_conv_dec(float const *fp1, float const *fp2)
{
/* This is a specialzed version of float_conv() for decoding
* which assumes a specific length and structure to the data
* in fp2.
*
* fp1 should be the dec.output_filter array and must be
* CVSD_DEC_FILTERLEN (48) long.
*
* fp2 should be one of the dec_filter_xx() tables listed
* in cvsdfilt.h. fp2 is assumed to be CVSD_DEC_FILTERLEN
* (48) entries long, is assumed to have 0.0 in the last
* entry, and is a symmetrical mirror around fp2[23] (ie,
* fp2[22] == fp2[24], fp2[0] == fp2[47], etc).
*/
float res = 0;
/* unrolling loop, also taking advantage of the symmetry
* of the sampling rate array*/
res += (fp1[0] + fp1[46]) * fp2[0];
res += (fp1[1] + fp1[45]) * fp2[1];
res += (fp1[2] + fp1[44]) * fp2[2];
res += (fp1[3] + fp1[43]) * fp2[3];
res += (fp1[4] + fp1[42]) * fp2[4];
res += (fp1[5] + fp1[41]) * fp2[5];
res += (fp1[6] + fp1[40]) * fp2[6];
res += (fp1[7] + fp1[39]) * fp2[7];
res += (fp1[8] + fp1[38]) * fp2[8];
res += (fp1[9] + fp1[37]) * fp2[9];
res += (fp1[10] + fp1[36]) * fp2[10];
res += (fp1[11] + fp1[35]) * fp2[11];
res += (fp1[12] + fp1[34]) * fp2[12];
res += (fp1[13] + fp1[33]) * fp2[13];
res += (fp1[14] + fp1[32]) * fp2[14];
res += (fp1[15] + fp1[31]) * fp2[15];
res += (fp1[16] + fp1[30]) * fp2[16];
res += (fp1[17] + fp1[29]) * fp2[17];
res += (fp1[18] + fp1[28]) * fp2[18];
res += (fp1[19] + fp1[27]) * fp2[19];
res += (fp1[20] + fp1[26]) * fp2[20];
res += (fp1[21] + fp1[25]) * fp2[21];
res += (fp1[22] + fp1[24]) * fp2[22];
res += (fp1[23]) * fp2[23];
return res;
}
/* ---------------------------------------------------------------------- */
/*
* some remarks about the implementation of the CVSD decoder
* the principal integrator is integrated into the output filter
* to achieve this, the coefficients of the output filter are multiplied
* with (1/(1-1/z)) in the initialisation code.
* the output filter must have a sharp zero at f=0 (i.e. the sum of the
* filter parameters must be zero). This prevents an accumulation of
* DC voltage at the principal integration.
*/
/* ---------------------------------------------------------------------- */
static void cvsdstartcommon(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
p->cvsd_rate = (ft->signal.rate <= 24000) ? 16000 : 32000;
ft->signal.rate = 8000;
ft->signal.channels = 1;
lsx_rawstart(ft, sox_true, sox_false, sox_true, SOX_ENCODING_CVSD, 1);
/*
* initialize the decoder
*/
p->com.overload = 0x5;
p->com.mla_int = 0;
/*
* timeconst = (1/e)^(200 / SR) = exp(-200/SR)
* SR is the sampling rate
*/
p->com.mla_tc0 = exp((-200.0)/((float)(p->cvsd_rate)));
/*
* phase_inc = 32000 / SR
*/
p->com.phase_inc = 32000 / p->cvsd_rate;
/*
* initialize bit shift register
*/
p->bit.shreg = p->bit.cnt = 0;
p->bit.mask = 1;
/*
* count the bytes written
*/
p->bytes_written = 0;
p->com.v_min = 1;
p->com.v_max = -1;
lsx_report("cvsd: bit rate %dbit/s, bits from %s", p->cvsd_rate,
ft->encoding.reverse_bits ? "msb to lsb" : "lsb to msb");
}
/* ---------------------------------------------------------------------- */
int lsx_cvsdstartread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
float *fp1;
int i;
cvsdstartcommon(ft);
p->com.mla_tc1 = 0.1 * (1 - p->com.mla_tc0);
p->com.phase = 0;
/*
* initialize the output filter coeffs (i.e. multiply
* the coeffs with (1/(1-1/z)) to achieve integration
* this is now done in the filter parameter generation utility
*/
/*
* zero the filter
*/
for(fp1 = p->c.dec.output_filter, i = CVSD_DEC_FILTERLEN*2; i > 0; i--)
*fp1++ = 0;
/* initialize mirror circular buffer offset to anything sane. */
p->c.dec.offset = CVSD_DEC_FILTERLEN - 1;
return (SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
int lsx_cvsdstartwrite(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
float *fp1;
int i;
cvsdstartcommon(ft);
p->com.mla_tc1 = 0.1 * (1 - p->com.mla_tc0);
p->com.phase = 4;
/*
* zero the filter
*/
for(fp1 = p->c.enc.input_filter, i = CVSD_ENC_FILTERLEN*2; i > 0; i--)
*fp1++ = 0;
p->c.enc.recon_int = 0;
/* initialize mirror circular buffer offset to anything sane. */
p->c.enc.offset = CVSD_ENC_FILTERLEN - 1;
return(SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
int lsx_cvsdstopwrite(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
if (p->bit.cnt) {
lsx_writeb(ft, p->bit.shreg);
p->bytes_written++;
}
lsx_debug("cvsd: min slope %f, max slope %f",
p->com.v_min, p->com.v_max);
return (SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
int lsx_cvsdstopread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
lsx_debug("cvsd: min value %f, max value %f",
p->com.v_min, p->com.v_max);
return(SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
size_t lsx_cvsdread(sox_format_t * ft, sox_sample_t *buf, size_t nsamp)
{
priv_t *p = (priv_t *) ft->priv;
size_t done = 0;
float oval;
while (done < nsamp) {
if (!p->bit.cnt) {
if (lsx_read_b_buf(ft, &(p->bit.shreg), (size_t) 1) != 1)
return done;
p->bit.cnt = 8;
p->bit.mask = 1;
}
/*
* handle one bit
*/
p->bit.cnt--;
p->com.overload = ((p->com.overload << 1) |
(!!(p->bit.shreg & p->bit.mask))) & 7;
p->bit.mask <<= 1;
p->com.mla_int *= p->com.mla_tc0;
if ((p->com.overload == 0) || (p->com.overload == 7))
p->com.mla_int += p->com.mla_tc1;
/* shift output filter window in mirror cirular buffer. */
if (p->c.dec.offset != 0)
--p->c.dec.offset;
else p->c.dec.offset = CVSD_DEC_FILTERLEN - 1;
/* write into both halves of the mirror circular buffer */
if (p->com.overload & 1)
{
p->c.dec.output_filter[p->c.dec.offset] = p->com.mla_int;
p->c.dec.output_filter[p->c.dec.offset + CVSD_DEC_FILTERLEN] = p->com.mla_int;
}
else
{
p->c.dec.output_filter[p->c.dec.offset] = -p->com.mla_int;
p->c.dec.output_filter[p->c.dec.offset + CVSD_DEC_FILTERLEN] = -p->com.mla_int;
}
/*
* check if the next output is due
*/
p->com.phase += p->com.phase_inc;
if (p->com.phase >= 4) {
oval = float_conv_dec(
p->c.dec.output_filter + p->c.dec.offset,
(p->cvsd_rate < 24000) ?
dec_filter_16 : dec_filter_32);
lsx_debug_more("input %d %f\n", debug_count, p->com.mla_int);
lsx_debug_more("recon %d %f\n", debug_count, oval);
debug_count++;
if (oval > p->com.v_max)
p->com.v_max = oval;
if (oval < p->com.v_min)
p->com.v_min = oval;
*buf++ = (oval * ((float)SOX_SAMPLE_MAX));
done++;
}
p->com.phase &= 3;
}
return done;
}
/* ---------------------------------------------------------------------- */
size_t lsx_cvsdwrite(sox_format_t * ft, const sox_sample_t *buf, size_t nsamp)
{
priv_t *p = (priv_t *) ft->priv;
size_t done = 0;
float inval;
for(;;) {
/*
* check if the next input is due
*/
if (p->com.phase >= 4) {
if (done >= nsamp)
return done;
/* shift input filter window in mirror cirular buffer. */
if (p->c.enc.offset != 0)
--p->c.enc.offset;
else p->c.enc.offset = CVSD_ENC_FILTERLEN - 1;
/* write into both halves of the mirror circular buffer */
p->c.enc.input_filter[p->c.enc.offset] =
p->c.enc.input_filter[p->c.enc.offset
+ CVSD_ENC_FILTERLEN] =
(*buf++) /
((float)SOX_SAMPLE_MAX);
done++;
}
p->com.phase &= 3;
/* insert input filter here! */
inval = float_conv_enc(
p->c.enc.input_filter + p->c.enc.offset,
(p->cvsd_rate < 24000) ?
(enc_filter_16[(p->com.phase >= 2)]) :
(enc_filter_32[p->com.phase]));
/*
* encode one bit
*/
p->com.overload = (((p->com.overload << 1) |
(inval > p->c.enc.recon_int)) & 7);
p->com.mla_int *= p->com.mla_tc0;
if ((p->com.overload == 0) || (p->com.overload == 7))
p->com.mla_int += p->com.mla_tc1;
if (p->com.mla_int > p->com.v_max)
p->com.v_max = p->com.mla_int;
if (p->com.mla_int < p->com.v_min)
p->com.v_min = p->com.mla_int;
if (p->com.overload & 1) {
p->c.enc.recon_int += p->com.mla_int;
p->bit.shreg |= p->bit.mask;
} else
p->c.enc.recon_int -= p->com.mla_int;
if ((++(p->bit.cnt)) >= 8) {
lsx_writeb(ft, p->bit.shreg);
p->bytes_written++;
p->bit.shreg = p->bit.cnt = 0;
p->bit.mask = 1;
} else
p->bit.mask <<= 1;
p->com.phase += p->com.phase_inc;
lsx_debug_more("input %d %f\n", debug_count, inval);
lsx_debug_more("recon %d %f\n", debug_count, p->c.enc.recon_int);
debug_count++;
}
}
/* ---------------------------------------------------------------------- */
/*
* DVMS file header
*/
/* FIXME: eliminate these 4 functions */
static uint32_t get32_le(unsigned char **p)
{
uint32_t val = (((*p)[3]) << 24) | (((*p)[2]) << 16) |
(((*p)[1]) << 8) | (**p);
(*p) += 4;
return val;
}
static uint16_t get16_le(unsigned char **p)
{
unsigned val = (((*p)[1]) << 8) | (**p);
(*p) += 2;
return val;
}
static void put32_le(unsigned char **p, uint32_t val)
{
*(*p)++ = val & 0xff;
*(*p)++ = (val >> 8) & 0xff;
*(*p)++ = (val >> 16) & 0xff;
*(*p)++ = (val >> 24) & 0xff;
}
static void put16_le(unsigned char **p, unsigned val)
{
*(*p)++ = val & 0xff;
*(*p)++ = (val >> 8) & 0xff;
}
struct dvms_header {
char Filename[14];
unsigned Id;
unsigned State;
time_t Unixtime;
unsigned Usender;
unsigned Ureceiver;
size_t Length;
unsigned Srate;
unsigned Days;
unsigned Custom1;
unsigned Custom2;
char Info[16];
char extend[64];
unsigned Crc;
};
#define DVMS_HEADER_LEN 120
/* ---------------------------------------------------------------------- */
static int dvms_read_header(sox_format_t * ft, struct dvms_header *hdr)
{
unsigned char hdrbuf[DVMS_HEADER_LEN];
unsigned char *pch = hdrbuf;
int i;
unsigned sum;
if (lsx_readbuf(ft, hdrbuf, sizeof(hdrbuf)) != sizeof(hdrbuf))
{
return (SOX_EOF);
}
for(i = sizeof(hdrbuf), sum = 0; i > /*2*/3; i--) /* Deti bug */
sum += *pch++;
pch = hdrbuf;
memcpy(hdr->Filename, pch, sizeof(hdr->Filename));
pch += sizeof(hdr->Filename);
hdr->Id = get16_le(&pch);
hdr->State = get16_le(&pch);
hdr->Unixtime = get32_le(&pch);
hdr->Usender = get16_le(&pch);
hdr->Ureceiver = get16_le(&pch);
hdr->Length = get32_le(&pch);
hdr->Srate = get16_le(&pch);
hdr->Days = get16_le(&pch);
hdr->Custom1 = get16_le(&pch);
hdr->Custom2 = get16_le(&pch);
memcpy(hdr->Info, pch, sizeof(hdr->Info));
pch += sizeof(hdr->Info);
memcpy(hdr->extend, pch, sizeof(hdr->extend));
pch += sizeof(hdr->extend);
hdr->Crc = get16_le(&pch);
if (sum != hdr->Crc)
{
lsx_report("DVMS header checksum error, read %u, calculated %u",
hdr->Crc, sum);
return (SOX_EOF);
}
return (SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
/*
* note! file must be seekable
*/
static int dvms_write_header(sox_format_t * ft, struct dvms_header *hdr)
{
unsigned char hdrbuf[DVMS_HEADER_LEN];
unsigned char *pch = hdrbuf;
unsigned char *pchs = hdrbuf;
int i;
unsigned sum;
memcpy(pch, hdr->Filename, sizeof(hdr->Filename));
pch += sizeof(hdr->Filename);
put16_le(&pch, hdr->Id);
put16_le(&pch, hdr->State);
put32_le(&pch, (unsigned)hdr->Unixtime);
put16_le(&pch, hdr->Usender);
put16_le(&pch, hdr->Ureceiver);
put32_le(&pch, (unsigned) hdr->Length);
put16_le(&pch, hdr->Srate);
put16_le(&pch, hdr->Days);
put16_le(&pch, hdr->Custom1);
put16_le(&pch, hdr->Custom2);
memcpy(pch, hdr->Info, sizeof(hdr->Info));
pch += sizeof(hdr->Info);
memcpy(pch, hdr->extend, sizeof(hdr->extend));
pch += sizeof(hdr->extend);
for(i = sizeof(hdrbuf), sum = 0; i > /*2*/3; i--) /* Deti bug */
sum += *pchs++;
hdr->Crc = sum;
put16_le(&pch, hdr->Crc);
if (lsx_seeki(ft, (off_t)0, SEEK_SET) < 0)
{
lsx_report("seek failed\n: %s",strerror(errno));
return (SOX_EOF);
}
if (lsx_writebuf(ft, hdrbuf, sizeof(hdrbuf)) != sizeof(hdrbuf))
{
lsx_report("%s",strerror(errno));
return (SOX_EOF);
}
return (SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
static void make_dvms_hdr(sox_format_t * ft, struct dvms_header *hdr)
{
priv_t *p = (priv_t *) ft->priv;
size_t len;
char * comment = lsx_cat_comments(ft->oob.comments);
memset(hdr->Filename, 0, sizeof(hdr->Filename));
len = strlen(ft->filename);
if (len >= sizeof(hdr->Filename))
len = sizeof(hdr->Filename)-1;
memcpy(hdr->Filename, ft->filename, len);
hdr->Id = hdr->State = 0;
hdr->Unixtime = sox_globals.repeatable? 0 : time(NULL);
hdr->Usender = hdr->Ureceiver = 0;
hdr->Length = p->bytes_written;
hdr->Srate = p->cvsd_rate/100;
hdr->Days = hdr->Custom1 = hdr->Custom2 = 0;
memset(hdr->Info, 0, sizeof(hdr->Info));
len = strlen(comment);
if (len >= sizeof(hdr->Info))
len = sizeof(hdr->Info)-1;
memcpy(hdr->Info, comment, len);
memset(hdr->extend, 0, sizeof(hdr->extend));
free(comment);
}
/* ---------------------------------------------------------------------- */
int lsx_dvmsstartread(sox_format_t * ft)
{
struct dvms_header hdr;
int rc;
rc = dvms_read_header(ft, &hdr);
if (rc){
lsx_fail_errno(ft,SOX_EHDR,"unable to read DVMS header");
return rc;
}
lsx_debug("DVMS header of source file \"%s\":", ft->filename);
lsx_debug(" filename \"%.14s\"", hdr.Filename);
lsx_debug(" id 0x%x", hdr.Id);
lsx_debug(" state 0x%x", hdr.State);
lsx_debug(" time %s", ctime(&hdr.Unixtime)); /* ctime generates lf */
lsx_debug(" usender %u", hdr.Usender);
lsx_debug(" ureceiver %u", hdr.Ureceiver);
lsx_debug(" length %" PRIuPTR, hdr.Length);
lsx_debug(" srate %u", hdr.Srate);
lsx_debug(" days %u", hdr.Days);
lsx_debug(" custom1 %u", hdr.Custom1);
lsx_debug(" custom2 %u", hdr.Custom2);
lsx_debug(" info \"%.16s\"", hdr.Info);
ft->signal.rate = (hdr.Srate < 240) ? 16000 : 32000;
lsx_debug("DVMS rate %dbit/s using %gbit/s deviation %g%%",
hdr.Srate*100, ft->signal.rate,
((ft->signal.rate - hdr.Srate*100) * 100) / ft->signal.rate);
rc = lsx_cvsdstartread(ft);
if (rc)
return rc;
return(SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
int lsx_dvmsstartwrite(sox_format_t * ft)
{
struct dvms_header hdr;
int rc;
rc = lsx_cvsdstartwrite(ft);
if (rc)
return rc;
make_dvms_hdr(ft, &hdr);
rc = dvms_write_header(ft, &hdr);
if (rc){
lsx_fail_errno(ft,rc,"cannot write DVMS header");
return rc;
}
if (!ft->seekable)
lsx_warn("Length in output .DVMS header will wrong since can't seek to fix it");
return(SOX_SUCCESS);
}
/* ---------------------------------------------------------------------- */
int lsx_dvmsstopwrite(sox_format_t * ft)
{
struct dvms_header hdr;
int rc;
lsx_cvsdstopwrite(ft);
if (!ft->seekable)
{
lsx_warn("File not seekable");
return (SOX_EOF);
}
if (lsx_seeki(ft, (off_t)0, 0) != 0)
{
lsx_fail_errno(ft,errno,"Can't rewind output file to rewrite DVMS header.");
return(SOX_EOF);
}
make_dvms_hdr(ft, &hdr);
rc = dvms_write_header(ft, &hdr);
if(rc){
lsx_fail_errno(ft,rc,"cannot write DVMS header");
return rc;
}
return rc;
}

68
app/src/main/cpp/sox/cvsd.h
Unescape View File

@ -0,0 +1,68 @@
/* libSoX DVMS format module (implementation in cvsd.c)
*
* Copyright (C) 1996-2007 Thomas Sailer and SoX Contributors
* Thomas Sailer (sailer@ife.ee.ethz.ch) (HB9JNX/AE4WA)
* Swiss Federal Institute of Technology, Electronics Lab
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#include "sox_i.h"
#define CVSD_ENC_FILTERLEN 16 /* PCM sampling rate */
#define CVSD_DEC_FILTERLEN 48 /* CVSD sampling rate */
typedef struct {
struct {
unsigned overload;
float mla_int;
float mla_tc0;
float mla_tc1;
unsigned phase;
unsigned phase_inc;
float v_min, v_max;
} com;
union {
struct {
/* mirror circular buffer */
float output_filter[CVSD_DEC_FILTERLEN*2];
unsigned offset; /* into output_filter; always in first half */
} dec;
struct {
float recon_int;
/* mirror circular buffer */
float input_filter[CVSD_ENC_FILTERLEN*2];
unsigned offset; /* into input_filter; always in first half */
} enc;
} c;
struct {
unsigned char shreg;
unsigned mask;
unsigned cnt;
} bit;
unsigned bytes_written;
unsigned cvsd_rate;
} cvsd_priv_t;
int lsx_cvsdstartread(sox_format_t * ft);
int lsx_cvsdstartwrite(sox_format_t * ft);
size_t lsx_cvsdread(sox_format_t * ft, sox_sample_t *buf, size_t nsamp);
size_t lsx_cvsdwrite(sox_format_t * ft, const sox_sample_t *buf, size_t nsamp);
int lsx_cvsdstopread(sox_format_t * ft);
int lsx_cvsdstopwrite(sox_format_t * ft);
int lsx_dvmsstartread(sox_format_t * ft);
int lsx_dvmsstartwrite(sox_format_t * ft);
int lsx_dvmsstopwrite(sox_format_t * ft);

112
app/src/main/cpp/sox/cvsdfilt.h
Unescape View File

@ -0,0 +1,112 @@
/* libSoX CVSD (Continuously Variable Slope Delta modulation)
* conversion routines
*
* The CVSD format is described in the MIL Std 188 113, which is
* available from http://bbs.itsi.disa.mil:5580/T3564
*
* Copyright (C) 1996
* Thomas Sailer (sailer@ife.ee.ethz.ch) (HB9JNX/AE4WA)
* Swiss Federal Institute of Technology, Electronics Lab
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
static const float dec_filter_16[48] = {
0.001102, 0.001159, 0.000187, -0.000175,
0.002097, 0.006543, 0.009384, 0.008004,
0.006562, 0.013569, 0.030745, 0.047053,
0.050491, 0.047388, 0.062171, 0.109115,
0.167120, 0.197144, 0.195471, 0.222098,
0.354745, 0.599184, 0.849632, 0.956536,
0.849632, 0.599184, 0.354745, 0.222098,
0.195471, 0.197144, 0.167120, 0.109115,
0.062171, 0.047388, 0.050491, 0.047053,
0.030745, 0.013569, 0.006562, 0.008004,
0.009384, 0.006543, 0.002097, -0.000175,
0.000187, 0.001159, 0.001102, 0.000000
};
/* ---------------------------------------------------------------------- */
static const float dec_filter_32[48] = {
0.001950, 0.004180, 0.006331, 0.007907,
0.008510, 0.008342, 0.008678, 0.011827,
0.020282, 0.035231, 0.055200, 0.075849,
0.091585, 0.098745, 0.099031, 0.101287,
0.120058, 0.170672, 0.262333, 0.392047,
0.542347, 0.684488, 0.786557, 0.823702,
0.786557, 0.684488, 0.542347, 0.392047,
0.262333, 0.170672, 0.120058, 0.101287,
0.099031, 0.098745, 0.091585, 0.075849,
0.055200, 0.035231, 0.020282, 0.011827,
0.008678, 0.008342, 0.008510, 0.007907,
0.006331, 0.004180, 0.001950, -0.000000
};
/* ---------------------------------------------------------------------- */
static const float enc_filter_16_0[16] = {
-0.000362, 0.004648, 0.001381, 0.008312,
0.041490, -0.001410, 0.124061, 0.247446,
-0.106761, -0.236326, -0.023798, -0.023506,
-0.030097, 0.001493, -0.005363, -0.001672
};
static const float enc_filter_16_1[16] = {
0.001672, 0.005363, -0.001493, 0.030097,
0.023506, 0.023798, 0.236326, 0.106761,
-0.247446, -0.124061, 0.001410, -0.041490,
-0.008312, -0.001381, -0.004648, 0.000362
};
static const float *enc_filter_16[2] = {
enc_filter_16_0, enc_filter_16_1
};
/* ---------------------------------------------------------------------- */
static const float enc_filter_32_0[16] = {
-0.000289, 0.002112, 0.001421, 0.002235,
0.021003, 0.001237, 0.047132, 0.129636,
-0.027328, -0.126462, -0.021456, -0.008069,
-0.017959, 0.000301, -0.002538, -0.001278
};
static const float enc_filter_32_1[16] = {
-0.000010, 0.002787, 0.000055, 0.006813,
0.020249, -0.000995, 0.077912, 0.112870,
-0.076980, -0.106971, -0.005096, -0.015449,
-0.012591, 0.000813, -0.003003, -0.000527
};
static const float enc_filter_32_2[16] = {
0.000527, 0.003003, -0.000813, 0.012591,
0.015449, 0.005096, 0.106971, 0.076980,
-0.112870, -0.077912, 0.000995, -0.020249,
-0.006813, -0.000055, -0.002787, 0.000010
};
static const float enc_filter_32_3[16] = {
0.001278, 0.002538, -0.000301, 0.017959,
0.008069, 0.021456, 0.126462, 0.027328,
-0.129636, -0.047132, -0.001237, -0.021003,
-0.002235, -0.001421, -0.002112, 0.000289
};
static const float *enc_filter_32[4] = {
enc_filter_32_0, enc_filter_32_1, enc_filter_32_2, enc_filter_32_3
};
/* ---------------------------------------------------------------------- */

162
app/src/main/cpp/sox/dat.c
Unescape View File

@ -0,0 +1,162 @@
/* libSoX text format file. Tom Littlejohn, March 93.
*
* Reads/writes sound files as text.
*
* Copyright 1998-2006 Chris Bagwell and SoX Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Lance Norskog And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
#include "sox_i.h"
#include <string.h>
#define LINEWIDTH (size_t)256
/* Private data for dat file */
typedef struct {
double timevalue, deltat;
int buffered;
char prevline[LINEWIDTH];
} priv_t;
static int sox_datstartread(sox_format_t * ft)
{
char inpstr[LINEWIDTH];
long rate;
int chan;
int status;
char sc;
/* Read lines until EOF or first non-comment line */
while ((status = lsx_reads(ft, inpstr, LINEWIDTH-1)) != SOX_EOF) {
inpstr[LINEWIDTH-1] = 0;
if ((sscanf(inpstr," %c", &sc) != 0) && (sc != ';')) break;
if (sscanf(inpstr," ; Sample Rate %ld", &rate)) {
ft->signal.rate=rate;
} else if (sscanf(inpstr," ; Channels %d", &chan)) {
ft->signal.channels=chan;
}
}
/* Hold a copy of the last line we read (first non-comment) */
if (status != SOX_EOF) {
strncpy(((priv_t *)ft->priv)->prevline, inpstr, (size_t)LINEWIDTH);
((priv_t *)ft->priv)->buffered = 1;
} else {
((priv_t *)ft->priv)->buffered = 0;
}
/* Default channels to 1 if not found */
if (ft->signal.channels == 0)
ft->signal.channels = 1;
ft->encoding.encoding = SOX_ENCODING_FLOAT_TEXT;
return (SOX_SUCCESS);
}
static int sox_datstartwrite(sox_format_t * ft)
{
priv_t * dat = (priv_t *) ft->priv;
char s[LINEWIDTH];
dat->timevalue = 0.0;
dat->deltat = 1.0 / (double)ft->signal.rate;
/* Write format comments to start of file */
sprintf(s,"; Sample Rate %ld\015\n", (long)ft->signal.rate);
lsx_writes(ft, s);
sprintf(s,"; Channels %d\015\n", (int)ft->signal.channels);
lsx_writes(ft, s);
return (SOX_SUCCESS);
}
static size_t sox_datread(sox_format_t * ft, sox_sample_t *buf, size_t nsamp)
{
char inpstr[LINEWIDTH];
int inpPtr = 0;
int inpPtrInc = 0;
double sampval = 0.0;
int retc = 0;
char sc = 0;
size_t done = 0;
size_t i=0;
/* Always read a complete set of channels */
nsamp -= (nsamp % ft->signal.channels);
while (done < nsamp) {
/* Read a line or grab the buffered first line */
if (((priv_t *)ft->priv)->buffered) {
strncpy(inpstr, ((priv_t *)ft->priv)->prevline, (size_t)LINEWIDTH);
inpstr[LINEWIDTH-1] = 0;
((priv_t *)ft->priv)->buffered=0;
} else {
lsx_reads(ft, inpstr, LINEWIDTH-1);
inpstr[LINEWIDTH-1] = 0;
if (lsx_eof(ft)) return (done);
}
/* Skip over comments - ie. 0 or more whitespace, then ';' */
if ((sscanf(inpstr," %c", &sc) != 0) && (sc==';')) continue;
/* Read a complete set of channels */
sscanf(inpstr," %*s%n", &inpPtr);
for (i=0; i<ft->signal.channels; i++) {
SOX_SAMPLE_LOCALS;
retc = sscanf(&inpstr[inpPtr]," %lg%n", &sampval, &inpPtrInc);
inpPtr += inpPtrInc;
if (retc != 1) {
lsx_fail_errno(ft,SOX_EOF,"Unable to read sample.");
return 0;
}
*buf++ = SOX_FLOAT_64BIT_TO_SAMPLE(sampval, ft->clips);
done++;
}
}
return (done);
}
static size_t sox_datwrite(sox_format_t * ft, const sox_sample_t *buf, size_t nsamp)
{
priv_t * dat = (priv_t *) ft->priv;
size_t done = 0;
double sampval=0.0;
char s[LINEWIDTH];
size_t i=0;
/* Always write a complete set of channels */
nsamp -= (nsamp % ft->signal.channels);
/* Write time, then sample values, then CRLF newline */
while(done < nsamp) {
sprintf(s," %15.8g ",dat->timevalue);
lsx_writes(ft, s);
for (i=0; i<ft->signal.channels; i++) {
sampval = SOX_SAMPLE_TO_FLOAT_64BIT(*buf++, ft->clips);
sprintf(s," %15.11g", sampval);
lsx_writes(ft, s);
done++;
}
sprintf(s," \r\n");
lsx_writes(ft, s);
dat->timevalue += dat->deltat;
}
return done;
}
LSX_FORMAT_HANDLER(dat)
{
static char const * const names[] = {"dat", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_FLOAT_TEXT, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Textual representation of the sampled audio", names, 0,
sox_datstartread, sox_datread, NULL,
sox_datstartwrite, sox_datwrite, NULL,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}

165
app/src/main/cpp/sox/dcshift.c
Unescape View File

@ -0,0 +1,165 @@
/* libSoX dcshift.c
* (c) 2000.04.15 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
*
* based on vol.c which is
* (c) 20/03/2000 Fabien COELHO <fabien@coelho.net> for sox.
*
* DC shift a sound file, with basic linear amplitude formula.
* Beware of saturations! clipping is checked and reported.
* Cannot handle different number of channels.
* Cannot handle rate change.
*/
#include "sox_i.h"
typedef struct {
double dcshift; /* DC shift. */
int uselimiter; /* boolean: are we using the limiter? */
double limiterthreshhold;
double limitergain; /* limiter gain. */
uint64_t limited; /* number of limited values to report. */
uint64_t totalprocessed;
} priv_t;
/*
* Process options: dcshift (double) type (amplitude, power, dB)
*/
static int sox_dcshift_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * dcs = (priv_t *) effp->priv;
dcs->dcshift = 1.0; /* default is no change */
dcs->uselimiter = 0; /* default is no limiter */
--argc, ++argv;
if (argc < 1)
return lsx_usage(effp);
if (argc && (!sscanf(argv[0], "%lf", &dcs->dcshift)))
return lsx_usage(effp);
if (argc>1)
{
if (!sscanf(argv[1], "%lf", &dcs->limitergain))
return lsx_usage(effp);
dcs->uselimiter = 1; /* ok, we'll use it */
/* The following equation is derived so that there is no
* discontinuity in output amplitudes */
/* and a SOX_SAMPLE_MAX input always maps to a SOX_SAMPLE_MAX output
* when the limiter is activated. */
/* (NOTE: There **WILL** be a discontinuity in the slope of the
* output amplitudes when using the limiter.) */
dcs->limiterthreshhold = SOX_SAMPLE_MAX * (1.0 - (fabs(dcs->dcshift) - dcs->limitergain));
}
return SOX_SUCCESS;
}
/*
* Start processing
*/
static int sox_dcshift_start(sox_effect_t * effp)
{
priv_t * dcs = (priv_t *) effp->priv;
if (dcs->dcshift == 0)
return SOX_EFF_NULL;
dcs->limited = 0;
dcs->totalprocessed = 0;
return SOX_SUCCESS;
}
/*
* Process data.
*/
static int sox_dcshift_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * dcs = (priv_t *) effp->priv;
double dcshift = dcs->dcshift;
double limitergain = dcs->limitergain;
double limiterthreshhold = dcs->limiterthreshhold;
double sample;
size_t len;
len = min(*osamp, *isamp);
/* report back dealt with amount. */
*isamp = len; *osamp = len;
if (dcs->uselimiter)
{
dcs->totalprocessed += len;
for (;len>0; len--)
{
sample = *ibuf++;
if (sample > limiterthreshhold && dcshift > 0)
{
sample = (sample - limiterthreshhold) * limitergain / (SOX_SAMPLE_MAX - limiterthreshhold) + limiterthreshhold + dcshift;
dcs->limited++;
}
else if (sample < -limiterthreshhold && dcshift < 0)
{
/* Note this should really be SOX_SAMPLE_MIN but
* the clip() below will take care of the overflow.
*/
sample = (sample + limiterthreshhold) * limitergain / (SOX_SAMPLE_MAX - limiterthreshhold) - limiterthreshhold + dcshift;
dcs->limited++;
}
else
{
/* Note this should consider SOX_SAMPLE_MIN but
* the clip() below will take care of the overflow.
*/
sample = dcshift * SOX_SAMPLE_MAX + sample;
}
SOX_SAMPLE_CLIP_COUNT(sample, effp->clips);
*obuf++ = sample;
}
}
else for (; len > 0; --len) { /* quite basic, with clipping */
double d = dcshift * (SOX_SAMPLE_MAX + 1.) + *ibuf++;
*obuf++ = SOX_ROUND_CLIP_COUNT(d, effp->clips);
}
return SOX_SUCCESS;
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
static int sox_dcshift_stop(sox_effect_t * effp)
{
priv_t * dcs = (priv_t *) effp->priv;
if (dcs->limited)
{
lsx_warn("DCSHIFT limited %" PRIu64 " values (%d percent).",
dcs->limited, (int) (dcs->limited * 100.0 / dcs->totalprocessed));
}
return SOX_SUCCESS;
}
static sox_effect_handler_t sox_dcshift_effect = {
"dcshift",
"shift [ limitergain ]\n"
"\tThe peak limiter has a gain much less than 1.0 (ie 0.05 or 0.02) which\n"
"\tis only used on peaks to prevent clipping. (default is no limiter)",
SOX_EFF_MCHAN | SOX_EFF_GAIN,
sox_dcshift_getopts,
sox_dcshift_start,
sox_dcshift_flow,
NULL,
sox_dcshift_stop,
NULL, sizeof(priv_t)
};
const sox_effect_handler_t *lsx_dcshift_effect_fn(void)
{
return &sox_dcshift_effect;
}

118
app/src/main/cpp/sox/deemph.plt
Unescape View File

@ -0,0 +1,118 @@
# 15/50us EIAJ de-emphasis filter for CD/DAT
#
# 09/02/98 (c) Heiko Eissfeldt
#
# 18/03/07 robs@users.sourceforge.net: changed to biquad for slightly
# better accuracy.
#
# License: LGPL (Lesser Gnu Public License)
#
# This implements the inverse filter of the optional pre-emphasis stage
# as defined by IEC 60908 (describing the audio cd format).
#
# Background: In the early days of audio cds, there were recording
# problems with noise (for example in classical recordings). The high
# dynamics of audio cds exposed these recording errors a lot.
#
# The commonly used solution at that time was to 'pre-emphasize' the
# trebles to have a better signal-noise-ratio. That is trebles were
# amplified before recording, so that they would give a stronger signal
# compared to the underlying (tape) noise.
#
# For that purpose the audio signal was prefiltered with the following
# frequency response (simple first order filter):
#
# V (in dB)
# ^
# |
# |~10dB _________________
# | /
# | / |
# | 20dB / decade ->/ |
# | / |
# |____________________/_ _ |_ _ _ _ _ _ _ _ _ Frequency
# |0 dB | |
# | | |
# | | |
# 3.1kHz ~10kHz
#
# So the recorded audio signal has amplified trebles compared to the
# original. HiFi cd players do correct this by applying an inverse
# filter automatically, the cd-rom drives or cd burners used by digital
# sampling programs (like cdda2wav) however do not.
#
# So, this is what this effect does.
#
# This is the gnuplot file for the frequency response of the deemphasis.
#
# The absolute error is <=0.04dB up to ~12kHz, and <=0.06dB up to 20kHz.
# First define the ideal filter:
# Filter parameters
T = 1. / 441000. # we use the tenfold sampling frequency
OmegaU = 1. / 15e-6
OmegaL = 15. / 50. * OmegaU
# Calculate filter coefficients
V0 = OmegaL / OmegaU
H0 = V0 - 1.
B = V0 * tan(OmegaU * T / 2.)
A1 = (B - 1.) / (B + 1.)
B0 = (1. + (1. - A1) * H0 / 2.)
B1 = (A1 + (A1 - 1.) * H0 / 2.)
# helper variables
D = B1 / B0
O = 2 * pi * T
# Ideal transfer function
Hi(f) = B0*sqrt((1 + 2*cos(f*O)*D + D*D)/(1 + 2*cos(f*O)*A1 + A1*A1))
# Now use a biquad (RBJ high shelf) with sampling frequency of 44100Hz
# to approximate the ideal curve:
# Filter parameters
t = 1. / 44100.
gain = -9.477
slope = .4845
f0 = 5283
# Calculate filter coefficients
A = exp(gain / 40. * log(10.))
w0 = 2. * pi * f0 * t
alpha = sin(w0) / 2. * sqrt((A + 1. / A) * (1. / slope - 1.) + 2.)
b0 = A * ((A + 1.) + (A - 1.) * cos(w0) + 2. * sqrt(A) * alpha)
b1 = -2. * A * ((A - 1.) + (A + 1.) * cos(w0))
b2 = A * ((A + 1.) + (A - 1.) * cos(w0) - 2. * sqrt(A) * alpha)
a0 = (A + 1.) - (A - 1.) * cos(w0) + 2. * sqrt(A) * alpha
a1 = 2. * ((A - 1.) - (A + 1.) * cos(w0))
a2 = (A + 1.) - (A - 1.) * cos(w0) - 2. * sqrt(A) * alpha
b2 = b2 / a0
b1 = b1 / a0
b0 = b0 / a0
a2 = a2 / a0
a1 = a1 / a0
# helper variables
o = 2 * pi * t
# Best fit transfer function
Hb(f) = sqrt((b0*b0 + b1*b1 + b2*b2 +\
2.*(b0*b1 + b1*b2)*cos(f*o) + 2.*(b0*b2)* cos(2.*f*o)) /\
(1. + a1*a1 + a2*a2 + 2.*(a1 + a1*a2)*cos(f*o) + 2.*a2*cos(2.*f*o)))
# plot real, best, ideal, level with halved attenuation,
# level at full attentuation, 10fold magnified error
set logscale x
set grid xtics ytics mxtics mytics
set key left bottom
plot [f=1000:20000] [-12:2] \
20 * log10(Hi(f)),\
20 * log10(Hb(f)),\
20 * log10(OmegaL/(2 * pi * f)),\
.5 * 20 * log10(V0),\
20 * log10(V0),\
200 * log10(Hb(f)/Hi(f))
pause -1 "Hit return to continue"

163
app/src/main/cpp/sox/delay.c
Unescape View File

@ -0,0 +1,163 @@
/* libSoX effect: Delay one or more channels (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <string.h>
typedef struct {
size_t argc;
struct { char *str; uint64_t delay; } *args;
uint64_t *max_delay;
uint64_t delay, pre_pad, pad;
size_t buffer_size, buffer_index;
sox_sample_t * buffer;
sox_bool drain_started;
} priv_t;
static int lsx_kill(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
unsigned i;
for (i = 0; i < p->argc; ++i)
free(p->args[i].str);
free(p->args);
free(p->max_delay);
return SOX_SUCCESS;
}
static int create(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
unsigned i;
--argc, ++argv;
p->argc = argc;
p->args = lsx_calloc(p->argc, sizeof(*p->args));
p->max_delay = lsx_malloc(sizeof(*p->max_delay));
for (i = 0; i < p->argc; ++i) {
char const * next = lsx_parseposition(0., p->args[i].str = lsx_strdup(argv[i]), NULL, (uint64_t)0, (uint64_t)0, '=');
if (!next || *next) {
lsx_kill(effp);
return lsx_usage(effp);
}
}
return SOX_SUCCESS;
}
static int stop(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
free(p->buffer);
return SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
uint64_t max_delay = 0, last_seen = 0, delay;
uint64_t in_length = effp->in_signal.length != SOX_UNKNOWN_LEN ?
effp->in_signal.length / effp->in_signal.channels : SOX_UNKNOWN_LEN;
if (effp->flow == 0) {
unsigned i;
if (p->argc > effp->in_signal.channels) {
lsx_fail("too few input channels");
return SOX_EOF;
}
for (i = 0; i < p->argc; ++i) {
if (!lsx_parseposition(effp->in_signal.rate, p->args[i].str, &delay, last_seen, in_length, '=') || delay == SOX_UNKNOWN_LEN) {
lsx_fail("Position relative to end of audio specified, but audio length is unknown");
return SOX_EOF;
}
p->args[i].delay = last_seen = delay;
if (delay > max_delay) {
max_delay = delay;
}
}
*p->max_delay = max_delay;
if (max_delay == 0)
return SOX_EFF_NULL;
effp->out_signal.length = effp->in_signal.length != SOX_UNKNOWN_LEN ?
effp->in_signal.length + max_delay * effp->in_signal.channels :
SOX_UNKNOWN_LEN;
lsx_debug("extending audio by %" PRIu64 " samples", max_delay);
}
max_delay = *p->max_delay;
if (effp->flow < p->argc)
p->buffer_size = p->args[effp->flow].delay;
p->buffer_index = p->delay = p->pre_pad = 0;
p->pad = max_delay - p->buffer_size;
p->buffer = lsx_malloc(p->buffer_size * sizeof(*p->buffer));
p->drain_started = sox_false;
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len = *isamp = *osamp = min(*isamp, *osamp);
if (!p->buffer_size)
memcpy(obuf, ibuf, len * sizeof(*obuf));
else for (; len; --len) {
if (p->delay < p->buffer_size) {
p->buffer[p->delay++] = *ibuf++;
*obuf++ = 0;
} else {
*obuf++ = p->buffer[p->buffer_index];
p->buffer[p->buffer_index++] = *ibuf++;
p->buffer_index %= p->buffer_size;
}
}
return SOX_SUCCESS;
}
static int drain(sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len;
if (! p->drain_started) {
p->drain_started = sox_true;
p->pre_pad = p->buffer_size - p->delay;
/* If the input was too short to fill the buffer completely,
flow() has not yet output enough silence to reach the
desired delay. */
}
len = *osamp = min(p->pre_pad + p->delay + p->pad, *osamp);
for (; p->pre_pad && len; --p->pre_pad, --len)
*obuf++ = 0;
for (; p->delay && len; --p->delay, --len) {
*obuf++ = p->buffer[p->buffer_index++];
p->buffer_index %= p->buffer_size;
}
for (; p->pad && len; --p->pad, --len)
*obuf++ = 0;
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_delay_effect_fn(void)
{
static sox_effect_handler_t handler = {
"delay", "{position}", SOX_EFF_LENGTH | SOX_EFF_MODIFY,
create, start, flow, drain, stop, lsx_kill, sizeof(priv_t)
};
return &handler;
}

137
app/src/main/cpp/sox/dft_filter.c
Unescape View File

@ -0,0 +1,137 @@
/* Abstract effect: dft filter Copyright (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "fft4g.h"
#include "dft_filter.h"
#include <string.h>
typedef dft_filter_t filter_t;
typedef dft_filter_priv_t priv_t;
void lsx_set_dft_filter(dft_filter_t *f, double *h, int n, int post_peak)
{
int i;
f->num_taps = n;
f->post_peak = post_peak;
f->dft_length = lsx_set_dft_length(f->num_taps);
f->coefs = lsx_calloc(f->dft_length, sizeof(*f->coefs));
for (i = 0; i < f->num_taps; ++i)
f->coefs[(i + f->dft_length - f->num_taps + 1) & (f->dft_length - 1)] = h[i] / f->dft_length * 2;
lsx_safe_rdft(f->dft_length, 1, f->coefs);
free(h);
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
fifo_create(&p->input_fifo, (int)sizeof(double));
memset(fifo_reserve(&p->input_fifo,
p->filter_ptr->post_peak), 0, sizeof(double) * p->filter_ptr->post_peak);
fifo_create(&p->output_fifo, (int)sizeof(double));
return SOX_SUCCESS;
}
static void filter(priv_t * p)
{
int i, num_in = max(0, fifo_occupancy(&p->input_fifo));
filter_t const * f = p->filter_ptr;
int const overlap = f->num_taps - 1;
double * output;
while (num_in >= f->dft_length) {
double const * input = fifo_read_ptr(&p->input_fifo);
fifo_read(&p->input_fifo, f->dft_length - overlap, NULL);
num_in -= f->dft_length - overlap;
output = fifo_reserve(&p->output_fifo, f->dft_length);
fifo_trim_by(&p->output_fifo, overlap);
memcpy(output, input, f->dft_length * sizeof(*output));
lsx_safe_rdft(f->dft_length, 1, output);
output[0] *= f->coefs[0];
output[1] *= f->coefs[1];
for (i = 2; i < f->dft_length; i += 2) {
double tmp = output[i];
output[i ] = f->coefs[i ] * tmp - f->coefs[i+1] * output[i+1];
output[i+1] = f->coefs[i+1] * tmp + f->coefs[i ] * output[i+1];
}
lsx_safe_rdft(f->dft_length, -1, output);
}
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t odone = min(*osamp, (size_t)fifo_occupancy(&p->output_fifo));
double const * s = fifo_read(&p->output_fifo, (int)odone, NULL);
lsx_save_samples(obuf, s, odone, &effp->clips);
p->samples_out += odone;
if (*isamp && odone < *osamp) {
double * t = fifo_write(&p->input_fifo, (int)*isamp, NULL);
p->samples_in += *isamp;
lsx_load_samples(t, ibuf, *isamp);
filter(p);
}
else *isamp = 0;
*osamp = odone;
return SOX_SUCCESS;
}
static int drain(sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
static size_t isamp = 0;
size_t remaining = p->samples_in > p->samples_out ?
(size_t)(p->samples_in - p->samples_out) : 0;
double * buff = lsx_calloc(1024, sizeof(*buff));
if (remaining > 0) {
while ((size_t)fifo_occupancy(&p->output_fifo) < remaining) {
fifo_write(&p->input_fifo, 1024, buff);
p->samples_in += 1024;
filter(p);
}
fifo_trim_to(&p->output_fifo, (int)remaining);
p->samples_in = 0;
}
free(buff);
return flow(effp, 0, obuf, &isamp, osamp);
}
static int stop(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
fifo_delete(&p->input_fifo);
fifo_delete(&p->output_fifo);
free(p->filter_ptr->coefs);
memset(p->filter_ptr, 0, sizeof(*p->filter_ptr));
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_dft_filter_effect_fn(void)
{
static sox_effect_handler_t handler = {
NULL, NULL, SOX_EFF_GAIN, NULL, start, flow, drain, stop, NULL, 0
};
return &handler;
}

16
app/src/main/cpp/sox/dft_filter.h
Unescape View File

@ -0,0 +1,16 @@
#include "fft4g.h"
#define FIFO_SIZE_T int
#include "fifo.h"
typedef struct {
int dft_length, num_taps, post_peak;
double * coefs;
} dft_filter_t;
typedef struct {
uint64_t samples_in, samples_out;
fifo_t input_fifo, output_fifo;
dft_filter_t filter, * filter_ptr;
} dft_filter_priv_t;
void lsx_set_dft_filter(dft_filter_t * f, double * h, int n, int post_peak);

436
app/src/main/cpp/sox/dither.c
Unescape View File

@ -0,0 +1,436 @@
/* Effect: dither/noise-shape Copyright (c) 2008-9 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifdef NDEBUG /* Enable assert always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox_i.h"
#include <assert.h>
#undef RANQD1
#define RANQD1 ranqd1(p->ranqd1)
typedef enum { /* Collection of various filters from the net */
Shape_none, Shape_lipshitz, Shape_f_weighted, Shape_modified_e_weighted,
Shape_improved_e_weighted, Shape_gesemann, Shape_shibata, Shape_low_shibata, Shape_high_shibata
} filter_name_t;
static lsx_enum_item const filter_names[] = {
LSX_ENUM_ITEM(Shape_,none)
LSX_ENUM_ITEM(Shape_,lipshitz)
{"f-weighted", Shape_f_weighted},
{"modified-e-weighted", Shape_modified_e_weighted},
{"improved-e-weighted", Shape_improved_e_weighted},
LSX_ENUM_ITEM(Shape_,gesemann)
LSX_ENUM_ITEM(Shape_,shibata)
{"low-shibata", Shape_low_shibata},
{"high-shibata", Shape_high_shibata},
{0, 0}};
typedef struct {
sox_rate_t rate;
enum {fir, iir} type;
size_t len;
int gain_cB; /* Chosen so clips are few if any, but not guaranteed none. */
double const * coefs;
filter_name_t name;
} filter_t;
static double const lip44[] = {2.033, -2.165, 1.959, -1.590, .6149};
static double const fwe44[] = {
2.412, -3.370, 3.937, -4.174, 3.353, -2.205, 1.281, -.569, .0847};
static double const mew44[] = {
1.662, -1.263, .4827, -.2913, .1268, -.1124, .03252, -.01265, -.03524};
static double const iew44[] = {
2.847, -4.685, 6.214, -7.184, 6.639, -5.032, 3.263, -1.632, .4191};
static double const ges44[] = {
2.2061, -.4706, -.2534, -.6214, 1.0587, .0676, -.6054, -.2738};
static double const ges48[] = {
2.2374, -.7339, -.1251, -.6033, .903, .0116, -.5853, -.2571};
static double const shi48[] = {
2.8720729351043701172, -5.0413231849670410156, 6.2442994117736816406,
-5.8483986854553222656, 3.7067542076110839844, -1.0495119094848632812,
-1.1830236911773681641, 2.1126792430877685547, -1.9094531536102294922,
0.99913084506988525391, -0.17090806365013122559, -0.32615602016448974609,
0.39127644896507263184, -0.26876461505889892578, 0.097676105797290802002,
-0.023473845794796943665,
};
static double const shi44[] = {
2.6773197650909423828, -4.8308925628662109375, 6.570110321044921875,
-7.4572014808654785156, 6.7263274192810058594, -4.8481650352478027344,
2.0412089824676513672, 0.7006359100341796875, -2.9537565708160400391,
4.0800385475158691406, -4.1845216751098632812, 3.3311812877655029297,
-2.1179926395416259766, 0.879302978515625, -0.031759146600961685181,
-0.42382788658142089844, 0.47882103919982910156, -0.35490813851356506348,
0.17496839165687561035, -0.060908168554306030273,
};
static double const shi38[] = {
1.6335992813110351562, -2.2615492343902587891, 2.4077029228210449219,
-2.6341717243194580078, 2.1440362930297851562, -1.8153258562088012695,
1.0816224813461303711, -0.70302653312683105469, 0.15991993248462677002,
0.041549518704414367676, -0.29416576027870178223, 0.2518316805362701416,
-0.27766478061676025391, 0.15785403549671173096, -0.10165894031524658203,
0.016833892092108726501,
};
static double const shi32[] =
{ /* dmaker 32000: bestmax=4.99659 (inverted) */
0.82118552923202515,
-1.0063692331314087,
0.62341964244842529,
-1.0447187423706055,
0.64532512426376343,
-0.87615132331848145,
0.52219754457473755,
-0.67434263229370117,
0.44954317808151245,
-0.52557498216629028,
0.34567299485206604,
-0.39618203043937683,
0.26791760325431824,
-0.28936097025871277,
0.1883765310049057,
-0.19097308814525604,
0.10431359708309174,
-0.10633844882249832,
0.046832218766212463,
-0.039653312414884567,
};
static double const shi22[] =
{ /* dmaker 22050: bestmax=5.77762 (inverted) */
0.056581053882837296,
-0.56956905126571655,
-0.40727734565734863,
-0.33870288729667664,
-0.29810553789138794,
-0.19039161503314972,
-0.16510021686553955,
-0.13468159735202789,
-0.096633769571781158,
-0.081049129366874695,
-0.064953058958053589,
-0.054459091275930405,
-0.043378707021474838,
-0.03660014271736145,
-0.026256965473294258,
-0.018786206841468811,
-0.013387725688517094,
-0.0090983230620622635,
-0.0026585909072309732,
-0.00042083300650119781,
};
static double const shi16[] =
{ /* dmaker 16000: bestmax=5.97128 (inverted) */
-0.37251132726669312,
-0.81423574686050415,
-0.55010956525802612,
-0.47405767440795898,
-0.32624706625938416,
-0.3161766529083252,
-0.2286367267370224,
-0.22916607558727264,
-0.19565616548061371,
-0.18160104751586914,
-0.15423151850700378,
-0.14104481041431427,
-0.11844276636838913,
-0.097583092749118805,
-0.076493598520755768,
-0.068106919527053833,
-0.041881654411554337,
-0.036922425031661987,
-0.019364040344953537,
-0.014994367957115173,
};
static double const shi11[] =
{ /* dmaker 11025: bestmax=5.9406 (inverted) */
-0.9264228343963623,
-0.98695987462997437,
-0.631156325340271,
-0.51966935396194458,
-0.39738872647285461,
-0.35679301619529724,
-0.29720726609230042,
-0.26310476660728455,
-0.21719355881214142,
-0.18561814725399017,
-0.15404847264289856,
-0.12687471508979797,
-0.10339745879173279,
-0.083688631653785706,
-0.05875682458281517,
-0.046893671154975891,
-0.027950936928391457,
-0.020740609616041183,
-0.009366452693939209,
-0.0060260160826146603,
};
static double const shi08[] =
{ /* dmaker 8000: bestmax=5.56234 (inverted) */
-1.202863335609436,
-0.94103097915649414,
-0.67878556251525879,
-0.57650017738342285,
-0.50004476308822632,
-0.44349345564842224,
-0.37833768129348755,
-0.34028723835945129,
-0.29413089156150818,
-0.24994957447052002,
-0.21715600788593292,
-0.18792112171649933,
-0.15268312394618988,
-0.12135542929172516,
-0.099610626697540283,
-0.075273610651493073,
-0.048787496984004974,
-0.042586319148540497,
-0.028991291299462318,
-0.011869125068187714,
};
static double const shl48[] = {
2.3925774097442626953, -3.4350297451019287109, 3.1853709220886230469,
-1.8117271661758422852, -0.20124770700931549072, 1.4759907722473144531,
-1.7210904359817504883, 0.97746700048446655273, -0.13790138065814971924,
-0.38185903429985046387, 0.27421241998672485352, 0.066584214568138122559,
-0.35223302245140075684, 0.37672343850135803223, -0.23964276909828186035,
0.068674825131893157959,
};
static double const shl44[] = {
2.0833916664123535156, -3.0418450832366943359, 3.2047898769378662109,
-2.7571926116943359375, 1.4978630542755126953, -0.3427594602108001709,
-0.71733748912811279297, 1.0737057924270629883, -1.0225815773010253906,
0.56649994850158691406, -0.20968692004680633545, -0.065378531813621520996,
0.10322438180446624756, -0.067442022264003753662, -0.00495197344571352005,
};
static double const shh44[] = {
3.0259189605712890625, -6.0268716812133789062, 9.195003509521484375,
-11.824929237365722656, 12.767142295837402344, -11.917946815490722656,
9.1739168167114257812, -5.3712320327758789062, 1.1393624544143676758,
2.4484779834747314453, -4.9719839096069335938, 6.0392003059387207031,
-5.9359521865844726562, 4.903278350830078125, -3.5527443885803222656,
2.1909697055816650391, -1.1672389507293701172, 0.4903914332389831543,
-0.16519790887832641602, 0.023217858746647834778,
};
static const filter_t filters[] = {
{44100, fir, 5, 210, lip44, Shape_lipshitz},
{46000, fir, 9, 276, fwe44, Shape_f_weighted},
{46000, fir, 9, 160, mew44, Shape_modified_e_weighted},
{46000, fir, 9, 321, iew44, Shape_improved_e_weighted},
{48000, iir, 4, 220, ges48, Shape_gesemann},
{44100, iir, 4, 230, ges44, Shape_gesemann},
{48000, fir, 16, 301, shi48, Shape_shibata},
{44100, fir, 20, 333, shi44, Shape_shibata},
{37800, fir, 16, 240, shi38, Shape_shibata},
{32000, fir, 20, 240/*TBD*/, shi32, Shape_shibata},
{22050, fir, 20, 240/*TBD*/, shi22, Shape_shibata},
{16000, fir, 20, 240/*TBD*/, shi16, Shape_shibata},
{11025, fir, 20, 240/*TBD*/, shi11, Shape_shibata},
{ 8000, fir, 20, 240/*TBD*/, shi08, Shape_shibata},
{48000, fir, 16, 250, shl48, Shape_low_shibata},
{44100, fir, 15, 250, shl44, Shape_low_shibata},
{44100, fir, 20, 383, shh44, Shape_high_shibata},
{ 0, fir, 0, 0, NULL, Shape_none},
};
#define MAX_N 20
typedef struct {
filter_name_t filter_name;
sox_bool auto_detect, alt_tpdf;
double dummy;
double previous_errors[MAX_N * 2];
double previous_outputs[MAX_N * 2];
size_t pos, prec;
uint64_t num_output;
int32_t history, ranqd1, r;
double const * coefs;
sox_bool dither_off;
sox_effect_handler_flow flow;
} priv_t;
#define CONVOLVE _ _ _ _
#define NAME flow_iir_4
#define IIR
#define N 4
#include "dither.h"
#undef IIR
#define CONVOLVE _ _ _ _ _
#define NAME flow_fir_5
#define N 5
#include "dither.h"
#define CONVOLVE _ _ _ _ _ _ _ _ _
#define NAME flow_fir_9
#define N 9
#include "dither.h"
#define CONVOLVE _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
#define NAME flow_fir_15
#define N 15
#include "dither.h"
#define CONVOLVE _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
#define NAME flow_fir_16
#define N 16
#include "dither.h"
#define CONVOLVE _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
#define NAME flow_fir_20
#define N 20
#include "dither.h"
static int flow_no_shape(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len = *isamp = *osamp = min(*isamp, *osamp);
while (len--) {
if (p->auto_detect) {
p->history = (p->history << 1) +
!!(*ibuf & (((unsigned)-1) >> p->prec));
if (p->history && p->dither_off) {
p->dither_off = sox_false;
lsx_debug("flow %" PRIuPTR ": on @ %" PRIu64, effp->flow, p->num_output);
} else if (!p->history && !p->dither_off) {
p->dither_off = sox_true;
lsx_debug("flow %" PRIuPTR ": off @ %" PRIu64, effp->flow, p->num_output);
}
}
if (!p->dither_off) {
int32_t r = RANQD1 >> p->prec;
double d = ((double)*ibuf++ + r + (p->alt_tpdf? -p->r : (RANQD1 >> p->prec))) / (1 << (32 - p->prec));
int i = d < 0? d - .5 : d + .5;
p->r = r;
if (i <= (-1 << (p->prec-1)))
++effp->clips, *obuf = SOX_SAMPLE_MIN;
else if (i > (int)SOX_INT_MAX(p->prec))
++effp->clips, *obuf = SOX_INT_MAX(p->prec) << (32 - p->prec);
else *obuf = i << (32 - p->prec);
++obuf;
}
else
*obuf++ = *ibuf++;
++p->num_output;
}
return SOX_SUCCESS;
}
static int getopts(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
int c;
lsx_getopt_t optstate;
lsx_getopt_init(argc, argv, "+aSsf:p:", NULL, lsx_getopt_flag_none, 1, &optstate);
while ((c = lsx_getopt(&optstate)) != -1) switch (c) {
case 'a': p->auto_detect = sox_true; break;
case 'S': p->alt_tpdf = sox_true; break;
case 's': p->filter_name = Shape_shibata; break;
case 'f':
p->filter_name = lsx_enum_option(c, optstate.arg, filter_names);
if (p->filter_name == INT_MAX)
return SOX_EOF;
break;
GETOPT_NUMERIC(optstate, 'p', prec, 1, 24)
default: lsx_fail("invalid option `-%c'", optstate.opt); return lsx_usage(effp);
}
argc -= optstate.ind, argv += optstate.ind;
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
double mult = 1; /* Amount the noise shaping multiplies up the TPDF (+/-1) */
if (p->prec == 0)
p->prec = effp->out_signal.precision;
if (effp->in_signal.precision <= p->prec || p->prec > 24)
return SOX_EFF_NULL; /* Dithering not needed at this resolution */
if (p->prec == 1) {
/* The general dither routines don't work in this case, so notify
user and leave it at that for now.
TODO: Some special-case treatment of 1-bit noise shaping will be
needed for meaningful DSD write support. */
lsx_warn("Dithering/noise-shaping to 1 bit is currently not supported.");
return SOX_EFF_NULL;
}
effp->out_signal.precision = p->prec;
p->flow = flow_no_shape;
if (p->filter_name) {
filter_t const * f;
for (f = filters; f->len && (f->name != p->filter_name || fabs(effp->in_signal.rate - f->rate) / f->rate > .05); ++f); /* 5% leeway on frequency */
if (!f->len) {
p->alt_tpdf |= effp->in_signal.rate >= 22050;
if (!effp->flow)
lsx_warn("no `%s' filter is available for rate %g; using %s TPDF",
lsx_find_enum_value(p->filter_name, filter_names)->text,
effp->in_signal.rate, p->alt_tpdf? "sloped" : "plain");
}
else {
assert(f->len <= MAX_N);
if (f->type == fir) switch(f->len) {
case 5: p->flow = flow_fir_5 ; break;
case 9: p->flow = flow_fir_9 ; break;
case 15: p->flow = flow_fir_15; break;
case 16: p->flow = flow_fir_16; break;
case 20: p->flow = flow_fir_20; break;
default: assert(sox_false);
} else switch(f->len) {
case 4: p->flow = flow_iir_4 ; break;
default: assert(sox_false);
}
p->coefs = f->coefs;
mult = dB_to_linear(f->gain_cB * 0.1);
}
}
p->ranqd1 = ranqd1(sox_globals.ranqd1) + effp->flow;
if (effp->in_signal.mult) /* (Takes account of ostart mult (sox.c). */
*effp->in_signal.mult *= (SOX_SAMPLE_MAX - (1 << (31 - p->prec)) *
(2 * mult + 1)) / (SOX_SAMPLE_MAX - (1 << (31 - p->prec)));
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
return p->flow(effp, ibuf, obuf, isamp, osamp);
}
sox_effect_handler_t const * lsx_dither_effect_fn(void)
{
static sox_effect_handler_t handler = {
"dither", "[-S|-s|-f filter] [-a] [-p precision]"
"\n (none) Use TPDF"
"\n -S Use sloped TPDF (without noise shaping)"
"\n -s Shape noise (with shibata filter)"
"\n -f name Set shaping filter to one of: lipshitz, f-weighted,"
"\n modified-e-weighted, improved-e-weighted, gesemann,"
"\n shibata, low-shibata, high-shibata."
"\n -a Automatically turn on & off dithering as needed (use with caution!)"
"\n -p bits Override the target sample precision",
SOX_EFF_PREC, getopts, start, flow, 0, 0, 0, sizeof(priv_t)
};
return &handler;
}

63
app/src/main/cpp/sox/dither.h
Unescape View File

@ -0,0 +1,63 @@
#ifdef IIR
#define _ output += p->coefs[j] * p->previous_errors[p->pos + j] \
- p->coefs[N + j] * p->previous_outputs[p->pos + j], ++j;
#else
#define _ d -= p->coefs[j] * p->previous_errors[p->pos + j], ++j;
#endif
static int NAME(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len = *isamp = *osamp = min(*isamp, *osamp);
while (len--) {
if (p->auto_detect) {
p->history = (p->history << 1) +
!!(*ibuf & (((unsigned)-1) >> p->prec));
if (p->history && p->dither_off) {
p->dither_off = sox_false;
lsx_debug("flow %" PRIuPTR ": on @ %" PRIu64, effp->flow, p->num_output);
} else if (!p->history && !p->dither_off) {
p->dither_off = sox_true;
memset(p->previous_errors, 0, sizeof(p->previous_errors));
memset(p->previous_outputs, 0, sizeof(p->previous_outputs));
lsx_debug("flow %" PRIuPTR ": off @ %" PRIu64, effp->flow, p->num_output);
}
}
if (!p->dither_off) {
int32_t r1 = RANQD1 >> p->prec, r2 = RANQD1 >> p->prec; /* Defer add! */
#ifdef IIR
double d1, d, output = 0;
#else
double d1, d = *ibuf++;
#endif
int i, j = 0;
CONVOLVE
assert(j == N);
p->pos = p->pos? p->pos - 1 : p->pos - 1 + N;
#ifdef IIR
d = *ibuf++ - output;
p->previous_outputs[p->pos + N] = p->previous_outputs[p->pos] = output;
#endif
d1 = (d + r1 + r2) / (1 << (32 - p->prec));
i = d1 < 0? d1 - .5 : d1 + .5;
p->previous_errors[p->pos + N] = p->previous_errors[p->pos] =
(double)i * (1 << (32 - p->prec)) - d;
if (i < (-1 << (p->prec-1)))
++effp->clips, *obuf = SOX_SAMPLE_MIN;
else if (i > (int)SOX_INT_MAX(p->prec))
++effp->clips, *obuf = SOX_INT_MAX(p->prec) << (32 - p->prec);
else *obuf = i << (32 - p->prec);
++obuf;
}
else
*obuf++ = *ibuf++;
++p->num_output;
}
return SOX_SUCCESS;
}
#undef CONVOLVE
#undef _
#undef NAME
#undef N

73
app/src/main/cpp/sox/divide.c
Unescape View File

@ -0,0 +1,73 @@
/* libSoX effect: divide Copyright (c) 2009 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* This is W.I.P. hence marked SOX_EFF_ALPHA for now.
* Needs better handling of when the divisor approaches or is zero; some
* sort of interpolation of the output values perhaps.
*/
#include "sox_i.h"
#include <string.h>
typedef struct {
sox_sample_t * last;
} priv_t;
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
p->last = lsx_calloc(effp->in_signal.channels, sizeof(*p->last));
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t i, len = min(*isamp, *osamp) / effp->in_signal.channels;
*osamp = *isamp = len * effp->in_signal.channels;
while (len--) {
double divisor = *obuf++ = *ibuf++;
if (divisor) {
double out, mult = 1. / SOX_SAMPLE_TO_FLOAT_64BIT(divisor,);
for (i = 1; i < effp->in_signal.channels; ++i) {
out = *ibuf++ * mult;
p->last[i] = *obuf++ = SOX_ROUND_CLIP_COUNT(out, effp->clips);
}
}
else for (i = 1; i < effp->in_signal.channels; ++i, ++ibuf)
*obuf++ = p->last[i];
}
return SOX_SUCCESS;
}
static int stop(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
free(p->last);
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_divide_effect_fn(void)
{
static sox_effect_handler_t handler = {
"divide", NULL, SOX_EFF_MCHAN | SOX_EFF_GAIN | SOX_EFF_ALPHA,
NULL, start, flow, NULL, stop, NULL, sizeof(priv_t)
};
return &handler;
}

85
app/src/main/cpp/sox/downsample.c
Unescape View File

@ -0,0 +1,85 @@
/* libSoX effect: Downsample
*
* First version of this effect written 11/2011 by Ulrich Klauer.
*
* Copyright 2011 Chris Bagwell and SoX Contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
typedef struct {
unsigned int factor;
unsigned int carry; /* number of samples still to be discarded,
carried over from last block */
} priv_t;
static int create(sox_effect_t *effp, int argc, char **argv)
{
priv_t *p = (priv_t*)effp->priv;
p->factor = 2;
--argc, ++argv;
do { /* break-able block */
NUMERIC_PARAMETER(factor, 1, 16384)
} while (0);
return argc ? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t *effp)
{
priv_t *p = (priv_t*) effp->priv;
effp->out_signal.rate = effp->in_signal.rate / p->factor;
return p->factor == 1 ? SOX_EFF_NULL : SOX_SUCCESS;
}
static int flow(sox_effect_t *effp, const sox_sample_t *ibuf,
sox_sample_t *obuf, size_t *isamp, size_t *osamp)
{
priv_t *p = (priv_t*)effp->priv;
size_t ilen = *isamp, olen = *osamp;
size_t t;
t = min(p->carry, ilen);
p->carry -= t;
ibuf += t; ilen -= t;
/* NB: either p->carry (usually) or ilen is now zero; hence, a
non-zero value of ilen implies p->carry == 0, and there is no
need to test for this in the following while and if. */
while (ilen >= p->factor && olen) {
*obuf++ = *ibuf;
ibuf += p->factor;
olen--; ilen -= p->factor;
}
if (ilen && olen) {
*obuf++ = *ibuf;
p->carry = p->factor - ilen;
olen--; ilen = 0;
}
*isamp -= ilen, *osamp -= olen;
return SOX_SUCCESS;
}
sox_effect_handler_t const *lsx_downsample_effect_fn(void)
{
static sox_effect_handler_t handler = {"downsample", "[factor (2)]",
SOX_EFF_RATE | SOX_EFF_MODIFY,
create, start, flow, NULL, NULL, NULL, sizeof(priv_t)};
return &handler;
}

32
app/src/main/cpp/sox/dvms-fmt.c
Unescape View File

@ -0,0 +1,32 @@
/* libSoX file format: DVMS (see cvsd.c) (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "cvsd.h"
LSX_FORMAT_HANDLER(dvms)
{
static char const * const names[] = {"dvms", "vms", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_CVSD, 1, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"MIL Std 188 113 Continuously Variable Slope Delta modulation with header",
names, SOX_FILE_MONO,
lsx_dvmsstartread, lsx_cvsdread, lsx_cvsdstopread,
lsx_dvmsstartwrite, lsx_cvsdwrite, lsx_dvmsstopwrite,
NULL, write_encodings, NULL, sizeof(cvsd_priv_t)
};
return &handler;
}

97
app/src/main/cpp/sox/earwax.c
Unescape View File

@ -0,0 +1,97 @@
/* libSoX earwax - makes listening to headphones easier November 9, 2000
*
* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
* This source code is freely redistributable and may be used for any purpose.
* This copyright notice must be maintained. Edward Beingessner And Sundry
* Contributors are not responsible for the consequences of using this
* software.
*
* This effect takes a 44.1kHz stereo (CD format) signal that is meant to be
* listened to on headphones, and adds audio cues to move the soundstage from
* inside your head (standard for headphones) to outside and in front of the
* listener (standard for speakers). This makes the sound much easier to listen
* to on headphones.
*/
#include "sox_i.h"
#include <string.h>
static const sox_sample_t filt[32 * 2] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
-1, -5, /* signal was from 30 degrees */
3, 3, /* from the ear, the other as */
-2, 5, /* if 330 degrees. */
-5, 0,
9, 1,
6, 3, /* Input */
-4, -1, /* Left Right */
-5, -3, /* __________ __________ */
-2, -5, /* | | | | */
-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
6, -7, /* / |__________| |__________| \ */
30, -29, /* / \ / \ */
12, -3, /* / X \ */
-11, 4, /* / / \ \ */
-3, 7, /* ____V_____ __________V V__________ _____V____ */
-20, 23, /* | | | | | | | | */
2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
1, -6, /* |__________| |__________| |__________| |__________| */
-14, -5, /* \ ___ / \ ___ / */
15, -18, /* \ / \ / _____ \ / \ / */
6, 7, /* `->| + |<--' / \ `-->| + |<-' */
15, -10, /* \___/ _/ \_ \___/ */
-14, 22, /* \ / \ / \ / */
-7, -2, /* `--->| | | |<---' */
-4, 9, /* \_/ \_/ */
6, -12, /* */
6, -6, /* Headphones */
0, -11,
0, -5,
4, 0};
#define NUMTAPS array_length(filt)
typedef struct {sox_sample_t tap[NUMTAPS];} priv_t; /* FIR filter z^-1 delays */
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
if (effp->in_signal.rate != 44100 || effp->in_signal.channels != 2) {
lsx_fail("works only with stereo audio sampled at 44100Hz (i.e. CDDA)");
return SOX_EOF;
}
memset(p->tap, 0, NUMTAPS * sizeof(*p->tap)); /* zero tap memory */
if (effp->in_signal.mult)
*effp->in_signal.mult *= dB_to_linear(-4.4);
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t i, len = *isamp = *osamp = min(*isamp, *osamp);
while (len--) { /* update taps and calculate output */
double output = 0;
for (i = NUMTAPS - 1; i; --i) {
p->tap[i] = p->tap[i - 1];
output += p->tap[i] * filt[i];
}
p->tap[0] = *ibuf++ / 64; /* scale output */
output += p->tap[0] * filt[0];
*obuf++ = SOX_ROUND_CLIP_COUNT(output, effp->clips);
}
return SOX_SUCCESS;
}
/* No drain: preserve audio file length; it's only 32 samples anyway. */
sox_effect_handler_t const *lsx_earwax_effect_fn(void)
{
static sox_effect_handler_t handler = {"earwax", NULL, SOX_EFF_MCHAN,
NULL, start, flow, NULL, NULL, NULL, sizeof(priv_t)};
return &handler;
}

266
app/src/main/cpp/sox/echo.c
Unescape View File

@ -0,0 +1,266 @@
/* libSoX Echo effect August 24, 1998
*
* Copyright (C) 1998 Juergen Mueller And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Juergen Mueller And Sundry Contributors are not responsible for
* the consequences of using this software.
*
*
* Flow diagram scheme for n delays ( 1 <= n <= MAX_ECHOS ):
*
* * gain-in ___
* ibuff -----------+------------------------------------------>| |
* | _________ | |
* | | | * decay 1 | |
* +----->| delay 1 |------------------------->| |
* | |_________| | |
* | _________ | + |
* | | | * decay 2 | |
* +---------->| delay 2 |-------------------->| |
* | |_________| | |
* : _________ | |
* | | | * decay n | |
* +--------------->| delay n |--------------->|___|
* |_________| |
* | * gain-out
* |
* +----->obuff
* Usage:
* echo gain-in gain-out delay-1 decay-1 [delay-2 decay-2 ... delay-n decay-n]
*
* Where:
* gain-in, decay-1 ... decay-n : 0.0 ... 1.0 volume
* gain-out : 0.0 ... volume
* delay-1 ... delay-n : > 0.0 msec
*
* Note:
* when decay is close to 1.0, the samples can begin clipping and the output
* can saturate!
*
* Hint:
* 1 / out-gain > gain-in ( 1 + decay-1 + ... + decay-n )
*/
#include "sox_i.h"
#include <stdlib.h> /* Harmless, and prototypes atof() etc. --dgc */
#define DELAY_BUFSIZ ( 50 * 50U * 1024 )
#define MAX_ECHOS 7 /* 24 bit x ( 1 + MAX_ECHOS ) = */
/* 24 bit x 8 = 32 bit !!! */
/* Private data for SKEL file */
typedef struct {
int counter;
int num_delays;
double *delay_buf;
float in_gain, out_gain;
float delay[MAX_ECHOS], decay[MAX_ECHOS];
ptrdiff_t samples[MAX_ECHOS], maxsamples;
size_t fade_out;
} priv_t;
/* Private data for SKEL file */
/*
* Process options
*/
static int sox_echo_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * echo = (priv_t *) effp->priv;
int i;
--argc, ++argv;
echo->num_delays = 0;
if ((argc < 4) || (argc % 2))
return lsx_usage(effp);
i = 0;
sscanf(argv[i++], "%f", &echo->in_gain);
sscanf(argv[i++], "%f", &echo->out_gain);
while (i < argc) {
if ( echo->num_delays >= MAX_ECHOS )
lsx_fail("echo: to many delays, use less than %i delays",
MAX_ECHOS);
/* Linux bug and it's cleaner. */
sscanf(argv[i++], "%f", &echo->delay[echo->num_delays]);
sscanf(argv[i++], "%f", &echo->decay[echo->num_delays]);
echo->num_delays++;
}
return (SOX_SUCCESS);
}
/*
* Prepare for processing.
*/
static int sox_echo_start(sox_effect_t * effp)
{
priv_t * echo = (priv_t *) effp->priv;
int i;
float sum_in_volume;
long j;
echo->maxsamples = 0;
if ( echo->in_gain < 0.0 )
{
lsx_fail("echo: gain-in must be positive!");
return (SOX_EOF);
}
if ( echo->in_gain > 1.0 )
{
lsx_fail("echo: gain-in must be less than 1.0!");
return (SOX_EOF);
}
if ( echo->out_gain < 0.0 )
{
lsx_fail("echo: gain-in must be positive!");
return (SOX_EOF);
}
for ( i = 0; i < echo->num_delays; i++ ) {
echo->samples[i] = echo->delay[i] * effp->in_signal.rate / 1000.0;
if ( echo->samples[i] < 1 )
{
lsx_fail("echo: delay must be positive!");
return (SOX_EOF);
}
if ( echo->samples[i] > (ptrdiff_t)DELAY_BUFSIZ )
{
lsx_fail("echo: delay must be less than %g seconds!",
DELAY_BUFSIZ / effp->in_signal.rate );
return (SOX_EOF);
}
if ( echo->decay[i] < 0.0 )
{
lsx_fail("echo: decay must be positive!" );
return (SOX_EOF);
}
if ( echo->decay[i] > 1.0 )
{
lsx_fail("echo: decay must be less than 1.0!" );
return (SOX_EOF);
}
if ( echo->samples[i] > echo->maxsamples )
echo->maxsamples = echo->samples[i];
}
echo->delay_buf = lsx_malloc(sizeof (double) * echo->maxsamples);
for ( j = 0; j < echo->maxsamples; ++j )
echo->delay_buf[j] = 0.0;
/* Be nice and check the hint with warning, if... */
sum_in_volume = 1.0;
for ( i = 0; i < echo->num_delays; i++ )
sum_in_volume += echo->decay[i];
if ( sum_in_volume * echo->in_gain > 1.0 / echo->out_gain )
lsx_warn("echo: warning >>> gain-out can cause saturation of output <<<");
echo->counter = 0;
echo->fade_out = echo->maxsamples;
effp->out_signal.length = SOX_UNKNOWN_LEN; /* TODO: calculate actual length */
return (SOX_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int sox_echo_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * echo = (priv_t *) effp->priv;
int j;
double d_in, d_out;
sox_sample_t out;
size_t len = min(*isamp, *osamp);
*isamp = *osamp = len;
while (len--) {
/* Store delays as 24-bit signed longs */
d_in = (double) *ibuf++ / 256;
/* Compute output first */
d_out = d_in * echo->in_gain;
for ( j = 0; j < echo->num_delays; j++ ) {
d_out += echo->delay_buf[
(echo->counter + echo->maxsamples - echo->samples[j]) % echo->maxsamples]
* echo->decay[j];
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echo->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Store input in delay buffer */
echo->delay_buf[echo->counter] = d_in;
/* Adjust the counter */
echo->counter = ( echo->counter + 1 ) % echo->maxsamples;
}
/* processed all samples */
return (SOX_SUCCESS);
}
/*
* Drain out reverb lines.
*/
static int sox_echo_drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * echo = (priv_t *) effp->priv;
double d_in, d_out;
sox_sample_t out;
int j;
size_t done;
done = 0;
/* drain out delay samples */
while ( ( done < *osamp ) && ( done < echo->fade_out ) ) {
d_in = 0;
d_out = 0;
for ( j = 0; j < echo->num_delays; j++ ) {
d_out += echo->delay_buf[
(echo->counter + echo->maxsamples - echo->samples[j]) % echo->maxsamples]
* echo->decay[j];
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echo->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Store input in delay buffer */
echo->delay_buf[echo->counter] = d_in;
/* Adjust the counters */
echo->counter = ( echo->counter + 1 ) % echo->maxsamples;
done++;
echo->fade_out--;
};
/* samples played, it remains */
*osamp = done;
if (echo->fade_out == 0)
return SOX_EOF;
else
return SOX_SUCCESS;
}
static int sox_echo_stop(sox_effect_t * effp)
{
priv_t * echo = (priv_t *) effp->priv;
free(echo->delay_buf);
echo->delay_buf = NULL;
return (SOX_SUCCESS);
}
static sox_effect_handler_t sox_echo_effect = {
"echo",
"gain-in gain-out delay decay [ delay decay ... ]",
SOX_EFF_LENGTH | SOX_EFF_GAIN,
sox_echo_getopts,
sox_echo_start,
sox_echo_flow,
sox_echo_drain,
sox_echo_stop,
NULL, sizeof(priv_t)
};
const sox_effect_handler_t *lsx_echo_effect_fn(void)
{
return &sox_echo_effect;
}

278
app/src/main/cpp/sox/echos.c
Unescape View File

@ -0,0 +1,278 @@
/* libSoX Echo effect August 24, 1998
*
* Copyright (C) 1998 Juergen Mueller And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Juergen Mueller And Sundry Contributors are not responsible for
* the consequences of using this software.
*
*
* Flow diagram scheme for n delays ( 1 <= n <= MAX_ECHOS ):
*
* * gain-in ___
* ibuff --+--------------------------------------------------->| |
* | * decay 1 | |
* | +----------------------------------->| |
* | | * decay 2 | + |
* | | +--------------------->| |
* | | | * decay n | |
* | _________ | _________ | _________ +--->|___|
* | | | | | | | | | | |
* +-->| delay 1 |-+-| delay 2 |-+...-| delay n |--+ | * gain-out
* |_________| |_________| |_________| |
* +----->obuff
* Usage:
* echos gain-in gain-out delay-1 decay-1 [delay-2 decay-2 ... delay-n decay-n]
*
* Where:
* gain-in, decay-1 ... decay-n : 0.0 ... 1.0 volume
* gain-out : 0.0 ... volume
* delay-1 ... delay-n : > 0.0 msec
*
* Note:
* when decay is close to 1.0, the samples can begin clipping and the output
* can saturate!
*
* Hint:
* 1 / out-gain > gain-in ( 1 + decay-1 + ... + decay-n )
*
*/
#include "sox_i.h"
#include <stdlib.h> /* Harmless, and prototypes atof() etc. --dgc */
#define DELAY_BUFSIZ ( 50 * 50U * 1024 )
#define MAX_ECHOS 7 /* 24 bit x ( 1 + MAX_ECHOS ) = */
/* 24 bit x 8 = 32 bit !!! */
/* Private data for SKEL file */
typedef struct {
int counter[MAX_ECHOS];
int num_delays;
double *delay_buf;
float in_gain, out_gain;
float delay[MAX_ECHOS], decay[MAX_ECHOS];
ptrdiff_t samples[MAX_ECHOS], pointer[MAX_ECHOS];
size_t sumsamples;
} priv_t;
/* Private data for SKEL file */
/*
* Process options
*/
static int sox_echos_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * echos = (priv_t *) effp->priv;
int i;
echos->num_delays = 0;
--argc, ++argv;
if ((argc < 4) || (argc % 2))
return lsx_usage(effp);
i = 0;
sscanf(argv[i++], "%f", &echos->in_gain);
sscanf(argv[i++], "%f", &echos->out_gain);
while (i < argc) {
/* Linux bug and it's cleaner. */
sscanf(argv[i++], "%f", &echos->delay[echos->num_delays]);
sscanf(argv[i++], "%f", &echos->decay[echos->num_delays]);
echos->num_delays++;
if ( echos->num_delays > MAX_ECHOS )
{
lsx_fail("echos: to many delays, use less than %i delays",
MAX_ECHOS);
return (SOX_EOF);
}
}
echos->sumsamples = 0;
return (SOX_SUCCESS);
}
/*
* Prepare for processing.
*/
static int sox_echos_start(sox_effect_t * effp)
{
priv_t * echos = (priv_t *) effp->priv;
int i;
float sum_in_volume;
unsigned long j;
if ( echos->in_gain < 0.0 )
{
lsx_fail("echos: gain-in must be positive!");
return (SOX_EOF);
}
if ( echos->in_gain > 1.0 )
{
lsx_fail("echos: gain-in must be less than 1.0!");
return (SOX_EOF);
}
if ( echos->out_gain < 0.0 )
{
lsx_fail("echos: gain-in must be positive!");
return (SOX_EOF);
}
for ( i = 0; i < echos->num_delays; i++ ) {
echos->samples[i] = echos->delay[i] * effp->in_signal.rate / 1000.0;
if ( echos->samples[i] < 1 )
{
lsx_fail("echos: delay must be positive!");
return (SOX_EOF);
}
if ( echos->samples[i] > (ptrdiff_t)DELAY_BUFSIZ )
{
lsx_fail("echos: delay must be less than %g seconds!",
DELAY_BUFSIZ / effp->in_signal.rate );
return (SOX_EOF);
}
if ( echos->decay[i] < 0.0 )
{
lsx_fail("echos: decay must be positive!" );
return (SOX_EOF);
}
if ( echos->decay[i] > 1.0 )
{
lsx_fail("echos: decay must be less than 1.0!" );
return (SOX_EOF);
}
echos->counter[i] = 0;
echos->pointer[i] = echos->sumsamples;
echos->sumsamples += echos->samples[i];
}
echos->delay_buf = lsx_malloc(sizeof (double) * echos->sumsamples);
for ( j = 0; j < echos->sumsamples; ++j )
echos->delay_buf[j] = 0.0;
/* Be nice and check the hint with warning, if... */
sum_in_volume = 1.0;
for ( i = 0; i < echos->num_delays; i++ )
sum_in_volume += echos->decay[i];
if ( sum_in_volume * echos->in_gain > 1.0 / echos->out_gain )
lsx_warn("echos: warning >>> gain-out can cause saturation of output <<<");
effp->out_signal.length = SOX_UNKNOWN_LEN; /* TODO: calculate actual length */
return (SOX_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int sox_echos_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * echos = (priv_t *) effp->priv;
int j;
double d_in, d_out;
sox_sample_t out;
size_t len = min(*isamp, *osamp);
*isamp = *osamp = len;
while (len--) {
/* Store delays as 24-bit signed longs */
d_in = (double) *ibuf++ / 256;
/* Compute output first */
d_out = d_in * echos->in_gain;
for ( j = 0; j < echos->num_delays; j++ ) {
d_out += echos->delay_buf[echos->counter[j] + echos->pointer[j]] * echos->decay[j];
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echos->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Mix decay of delays and input */
for ( j = 0; j < echos->num_delays; j++ ) {
if ( j == 0 )
echos->delay_buf[echos->counter[j] + echos->pointer[j]] = d_in;
else
echos->delay_buf[echos->counter[j] + echos->pointer[j]] =
echos->delay_buf[echos->counter[j-1] + echos->pointer[j-1]] + d_in;
}
/* Adjust the counters */
for ( j = 0; j < echos->num_delays; j++ )
echos->counter[j] =
( echos->counter[j] + 1 ) % echos->samples[j];
}
/* processed all samples */
return (SOX_SUCCESS);
}
/*
* Drain out reverb lines.
*/
static int sox_echos_drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * echos = (priv_t *) effp->priv;
double d_in, d_out;
sox_sample_t out;
int j;
size_t done;
done = 0;
/* drain out delay samples */
while ( ( done < *osamp ) && ( done < echos->sumsamples ) ) {
d_in = 0;
d_out = 0;
for ( j = 0; j < echos->num_delays; j++ ) {
d_out += echos->delay_buf[echos->counter[j] + echos->pointer[j]] * echos->decay[j];
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echos->out_gain;
out = SOX_24BIT_CLIP_COUNT((sox_sample_t) d_out, effp->clips);
*obuf++ = out * 256;
/* Mix decay of delays and input */
for ( j = 0; j < echos->num_delays; j++ ) {
if ( j == 0 )
echos->delay_buf[echos->counter[j] + echos->pointer[j]] = d_in;
else
echos->delay_buf[echos->counter[j] + echos->pointer[j]] =
echos->delay_buf[echos->counter[j-1] + echos->pointer[j-1]];
}
/* Adjust the counters */
for ( j = 0; j < echos->num_delays; j++ )
echos->counter[j] =
( echos->counter[j] + 1 ) % echos->samples[j];
done++;
echos->sumsamples--;
};
/* samples played, it remains */
*osamp = done;
if (echos->sumsamples == 0)
return SOX_EOF;
else
return SOX_SUCCESS;
}
/*
* Clean up echos effect.
*/
static int sox_echos_stop(sox_effect_t * effp)
{
priv_t * echos = (priv_t *) effp->priv;
free(echos->delay_buf);
echos->delay_buf = NULL;
return (SOX_SUCCESS);
}
static sox_effect_handler_t sox_echos_effect = {
"echos",
"gain-in gain-out delay decay [ delay decay ... ]",
SOX_EFF_LENGTH | SOX_EFF_GAIN,
sox_echos_getopts,
sox_echos_start,
sox_echos_flow,
sox_echos_drain,
sox_echos_stop,
NULL, sizeof(priv_t)
};
const sox_effect_handler_t *lsx_echos_effect_fn(void)
{
return &sox_echos_effect;
}

669
app/src/main/cpp/sox/effects.c
Unescape View File

@ -0,0 +1,669 @@
/* SoX Effects chain (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define LSX_EFF_ALIAS
#include "sox_i.h"
#include <assert.h>
#include <string.h>
#ifdef HAVE_STRINGS_H
#include <strings.h>
#endif
#define DEBUG_EFFECTS_CHAIN 0
/* Default effect handler functions for do-nothing situations: */
static int default_function(sox_effect_t * effp UNUSED)
{
return SOX_SUCCESS;
}
/* Pass through samples verbatim */
int lsx_flow_copy(sox_effect_t * effp UNUSED, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
*isamp = *osamp = min(*isamp, *osamp);
memcpy(obuf, ibuf, *isamp * sizeof(*obuf));
return SOX_SUCCESS;
}
/* Inform no more samples to drain */
static int default_drain(sox_effect_t * effp UNUSED, sox_sample_t *obuf UNUSED, size_t *osamp)
{
*osamp = 0;
return SOX_EOF;
}
/* Check that no parameters have been given */
static int default_getopts(sox_effect_t * effp, int argc, char **argv UNUSED)
{
return --argc? lsx_usage(effp) : SOX_SUCCESS;
}
/* Partially initialise the effect structure; signal info will come later */
sox_effect_t * sox_create_effect(sox_effect_handler_t const * eh)
{
sox_effect_t * effp = lsx_calloc(1, sizeof(*effp));
effp->obuf = NULL;
effp->global_info = sox_get_effects_globals();
effp->handler = *eh;
if (!effp->handler.getopts) effp->handler.getopts = default_getopts;
if (!effp->handler.start ) effp->handler.start = default_function;
if (!effp->handler.flow ) effp->handler.flow = lsx_flow_copy;
if (!effp->handler.drain ) effp->handler.drain = default_drain;
if (!effp->handler.stop ) effp->handler.stop = default_function;
if (!effp->handler.kill ) effp->handler.kill = default_function;
effp->priv = lsx_calloc(1, effp->handler.priv_size);
return effp;
} /* sox_create_effect */
int sox_effect_options(sox_effect_t *effp, int argc, char * const argv[])
{
int result;
char * * argv2 = lsx_malloc((argc + 1) * sizeof(*argv2));
argv2[0] = (char *)effp->handler.name;
memcpy(argv2 + 1, argv, argc * sizeof(*argv2));
result = effp->handler.getopts(effp, argc + 1, argv2);
free(argv2);
return result;
} /* sox_effect_options */
/* Effects chain: */
sox_effects_chain_t * sox_create_effects_chain(
sox_encodinginfo_t const * in_enc, sox_encodinginfo_t const * out_enc)
{
sox_effects_chain_t * result = lsx_calloc(1, sizeof(sox_effects_chain_t));
result->global_info = *sox_get_effects_globals();
result->in_enc = in_enc;
result->out_enc = out_enc;
return result;
} /* sox_create_effects_chain */
void sox_delete_effects_chain(sox_effects_chain_t *ecp)
{
if (ecp && ecp->length)
sox_delete_effects(ecp);
free(ecp->effects);
free(ecp);
} /* sox_delete_effects_chain */
/* Effect can call in start() or flow() to set minimum input size to flow() */
int lsx_effect_set_imin(sox_effect_t * effp, size_t imin)
{
if (imin > sox_globals.bufsiz / effp->flows) {
lsx_fail("sox_bufsiz not big enough");
return SOX_EOF;
}
effp->imin = imin;
return SOX_SUCCESS;
}
/* Effects table to be extended in steps of EFF_TABLE_STEP */
#define EFF_TABLE_STEP 8
/* Add an effect to the chain. *in is the input signal for this effect. *out is
* a suggestion as to what the output signal should be, but depending on its
* given options and *in, the effect can choose to do differently. Whatever
* output rate and channels the effect does produce are written back to *in,
* ready for the next effect in the chain.
*/
int sox_add_effect(sox_effects_chain_t * chain, sox_effect_t * effp, sox_signalinfo_t * in, sox_signalinfo_t const * out)
{
int ret, (*start)(sox_effect_t * effp) = effp->handler.start;
size_t f;
sox_effect_t eff0; /* Copy of effect for flow 0 before calling start */
effp->global_info = &chain->global_info;
effp->in_signal = *in;
effp->out_signal = *out;
effp->in_encoding = chain->in_enc;
effp->out_encoding = chain->out_enc;
if (!(effp->handler.flags & SOX_EFF_CHAN))
effp->out_signal.channels = in->channels;
if (!(effp->handler.flags & SOX_EFF_RATE))
effp->out_signal.rate = in->rate;
if (!(effp->handler.flags & SOX_EFF_PREC))
effp->out_signal.precision = (effp->handler.flags & SOX_EFF_MODIFY)?
in->precision : SOX_SAMPLE_PRECISION;
if (!(effp->handler.flags & SOX_EFF_GAIN))
effp->out_signal.mult = in->mult;
effp->flows =
(effp->handler.flags & SOX_EFF_MCHAN)? 1 : effp->in_signal.channels;
effp->clips = 0;
effp->imin = 0;
eff0 = *effp, eff0.priv = lsx_memdup(eff0.priv, eff0.handler.priv_size);
eff0.in_signal.mult = NULL; /* Only used in channel 0 */
ret = start(effp);
if (ret == SOX_EFF_NULL) {
lsx_report("has no effect in this configuration");
free(eff0.priv);
effp->handler.kill(effp);
free(effp->priv);
effp->priv = NULL;
return SOX_SUCCESS;
}
if (ret != SOX_SUCCESS) {
free(eff0.priv);
return SOX_EOF;
}
if (in->mult)
lsx_debug("mult=%g", *in->mult);
if (!(effp->handler.flags & SOX_EFF_LENGTH)) {
effp->out_signal.length = in->length;
if (effp->out_signal.length != SOX_UNKNOWN_LEN) {
if (effp->handler.flags & SOX_EFF_CHAN)
effp->out_signal.length =
effp->out_signal.length / in->channels * effp->out_signal.channels;
if (effp->handler.flags & SOX_EFF_RATE)
effp->out_signal.length =
effp->out_signal.length / in->rate * effp->out_signal.rate + .5;
}
}
*in = effp->out_signal;
if (chain->length == chain->table_size) {
chain->table_size += EFF_TABLE_STEP;
lsx_debug_more("sox_add_effect: extending effects table, "
"new size = %" PRIuPTR, chain->table_size);
lsx_revalloc(chain->effects, chain->table_size);
}
chain->effects[chain->length] =
lsx_calloc(effp->flows, sizeof(chain->effects[chain->length][0]));
chain->effects[chain->length][0] = *effp;
for (f = 1; f < effp->flows; ++f) {
chain->effects[chain->length][f] = eff0;
chain->effects[chain->length][f].flow = f;
chain->effects[chain->length][f].priv = lsx_memdup(eff0.priv, eff0.handler.priv_size);
if (start(&chain->effects[chain->length][f]) != SOX_SUCCESS) {
free(eff0.priv);
return SOX_EOF;
}
}
++chain->length;
free(eff0.priv);
return SOX_SUCCESS;
}
/* An effect's output buffer (effp->obuf) generally has this layout:
* |. . . A1A2A3B1B2B3C1C2C3. . . . . . . . . . . . . . . . . . |
* ^0 ^obeg ^oend ^bufsiz
* (where A1 is the first sample of channel 1, A2 the first sample of
* channel 2, etc.), i.e. the channels are interleaved.
* However, while sox_flow_effects() is running, output buffers are
* adapted to how the following effect expects its input, to avoid
* back-and-forth conversions. If the following effect operates on
* each of several channels separately (flows > 1), the layout is
* changed to this uninterleaved form:
* |. A1B1C1. . . . . . . A2B2C2. . . . . . . A3B3C3. . . . . . |
* ^0 ^obeg ^oend ^bufsiz
* <--- channel 1 ----><--- channel 2 ----><--- channel 3 ---->
* The buffer is logically subdivided into channel buffers of size
* bufsiz/flows each, starting at offsets 0, bufsiz/flows,
* 2*(bufsiz/flows) etc. Within the channel buffers, the data starts
* at position obeg/flows and ends before oend/flows. In case bufsiz
* is not evenly divisible by flows, there will be an unused area at
* the very end of the output buffer.
* The interleave() and deinterleave() functions convert between these
* two representations.
*/
static void interleave(size_t flows, size_t length, sox_sample_t *from,
size_t bufsiz, size_t offset, sox_sample_t *to);
static void deinterleave(size_t flows, size_t length, sox_sample_t *from,
sox_sample_t *to, size_t bufsiz, size_t offset);
static int flow_effect(sox_effects_chain_t * chain, size_t n)
{
sox_effect_t *effp1 = chain->effects[n - 1];
sox_effect_t *effp = chain->effects[n];
int effstatus = SOX_SUCCESS;
size_t f = 0;
size_t idone = effp1->oend - effp1->obeg;
size_t obeg = sox_globals.bufsiz - effp->oend;
sox_bool il_change = (effp->flows == 1) !=
(chain->length == n + 1 || chain->effects[n+1]->flows == 1);
#if DEBUG_EFFECTS_CHAIN
size_t pre_idone = idone;
size_t pre_odone = obeg;
#endif
if (effp->flows == 1) { /* Run effect on all channels at once */
idone -= idone % effp->in_signal.channels;
effstatus = effp->handler.flow(effp, effp1->obuf + effp1->obeg,
il_change ? chain->il_buf : effp->obuf + effp->oend,
&idone, &obeg);
if (obeg % effp->out_signal.channels != 0) {
lsx_fail("multi-channel effect flowed asymmetrically!");
effstatus = SOX_EOF;
}
if (il_change)
deinterleave(chain->effects[n+1]->flows, obeg, chain->il_buf,
effp->obuf, sox_globals.bufsiz, effp->oend);
} else { /* Run effect on each channel individually */
sox_sample_t *obuf = il_change ? chain->il_buf : effp->obuf;
size_t flow_offs = sox_globals.bufsiz/effp->flows;
size_t idone_min = SOX_SIZE_MAX, idone_max = 0;
size_t odone_min = SOX_SIZE_MAX, odone_max = 0;
#ifdef HAVE_OPENMP_3_1
#pragma omp parallel for \
if(sox_globals.use_threads) \
schedule(static) default(none) \
shared(effp,effp1,idone,obeg,obuf,flow_offs,chain,n,effstatus) \
reduction(min:idone_min,odone_min) reduction(max:idone_max,odone_max)
#elif defined HAVE_OPENMP
#pragma omp parallel for \
if(sox_globals.use_threads) \
schedule(static) default(none) \
shared(effp,effp1,idone,obeg,obuf,flow_offs,chain,n,effstatus) \
firstprivate(idone_min,odone_min,idone_max,odone_max) \
lastprivate(idone_min,odone_min,idone_max,odone_max)
#endif
for (f = 0; f < effp->flows; ++f) {
size_t idonec = idone / effp->flows;
size_t odonec = obeg / effp->flows;
int eff_status_c = effp->handler.flow(&chain->effects[n][f],
effp1->obuf + f*flow_offs + effp1->obeg/effp->flows,
obuf + f*flow_offs + effp->oend/effp->flows,
&idonec, &odonec);
idone_min = min(idonec, idone_min); idone_max = max(idonec, idone_max);
odone_min = min(odonec, odone_min); odone_max = max(odonec, odone_max);
if (eff_status_c != SOX_SUCCESS)
effstatus = SOX_EOF;
}
if (idone_min != idone_max || odone_min != odone_max) {
lsx_fail("flowed asymmetrically!");
effstatus = SOX_EOF;
}
idone = effp->flows * idone_max;
obeg = effp->flows * odone_max;
if (il_change)
interleave(effp->flows, obeg, chain->il_buf, sox_globals.bufsiz,
effp->oend, effp->obuf + effp->oend);
}
effp1->obeg += idone;
if (effp1->obeg == effp1->oend)
effp1->obeg = effp1->oend = 0;
else if (effp1->oend - effp1->obeg < effp->imin) { /* Need to refill? */
size_t flow_offs = sox_globals.bufsiz/effp->flows;
for (f = 0; f < effp->flows; ++f)
memcpy(effp1->obuf + f * flow_offs,
effp1->obuf + f * flow_offs + effp1->obeg/effp->flows,
(effp1->oend - effp1->obeg)/effp->flows * sizeof(*effp1->obuf));
effp1->oend -= effp1->obeg;
effp1->obeg = 0;
}
effp->oend += obeg;
#if DEBUG_EFFECTS_CHAIN
lsx_report("\t" "flow: %2" PRIuPTR " (%1" PRIuPTR ") "
"%5" PRIuPTR " %5" PRIuPTR " %5" PRIuPTR " %5" PRIuPTR " "
"%5" PRIuPTR " [%" PRIuPTR "-%" PRIuPTR "]",
n, effp->flows, pre_idone, pre_odone, idone, obeg,
effp1->oend - effp1->obeg, effp1->obeg, effp1->oend);
#endif
return effstatus == SOX_SUCCESS? SOX_SUCCESS : SOX_EOF;
}
/* The same as flow_effect but with no input */
static int drain_effect(sox_effects_chain_t * chain, size_t n)
{
sox_effect_t *effp = chain->effects[n];
int effstatus = SOX_SUCCESS;
size_t f = 0;
size_t obeg = sox_globals.bufsiz - effp->oend;
sox_bool il_change = (effp->flows == 1) !=
(chain->length == n + 1 || chain->effects[n+1]->flows == 1);
#if DEBUG_EFFECTS_CHAIN
size_t pre_odone = obeg;
#endif
if (effp->flows == 1) { /* Run effect on all channels at once */
effstatus = effp->handler.drain(effp,
il_change ? chain->il_buf : effp->obuf + effp->oend,
&obeg);
if (obeg % effp->out_signal.channels != 0) {
lsx_fail("multi-channel effect drained asymmetrically!");
effstatus = SOX_EOF;
}
if (il_change)
deinterleave(chain->effects[n+1]->flows, obeg, chain->il_buf,
effp->obuf, sox_globals.bufsiz, effp->oend);
} else { /* Run effect on each channel individually */
sox_sample_t *obuf = il_change ? chain->il_buf : effp->obuf;
size_t flow_offs = sox_globals.bufsiz/effp->flows;
size_t odone_last = 0; /* Initialised to prevent warning */
for (f = 0; f < effp->flows; ++f) {
size_t odonec = obeg / effp->flows;
int eff_status_c = effp->handler.drain(&chain->effects[n][f],
obuf + f*flow_offs + effp->oend/effp->flows,
&odonec);
if (f && (odonec != odone_last)) {
lsx_fail("drained asymmetrically!");
effstatus = SOX_EOF;
}
odone_last = odonec;
if (eff_status_c != SOX_SUCCESS)
effstatus = SOX_EOF;
}
obeg = effp->flows * odone_last;
if (il_change)
interleave(effp->flows, obeg, chain->il_buf, sox_globals.bufsiz,
effp->oend, effp->obuf + effp->oend);
}
if (!obeg) /* This is the only thing that drain has and flow hasn't */
effstatus = SOX_EOF;
effp->oend += obeg;
#if DEBUG_EFFECTS_CHAIN
lsx_report("\t" "drain: %2" PRIuPTR " (%1" PRIuPTR ") "
"%5" PRIuPTR " %5" PRIuPTR " %5" PRIuPTR " %5" PRIuPTR,
n, effp->flows, (size_t)0, pre_odone, (size_t)0, obeg);
#endif
return effstatus == SOX_SUCCESS? SOX_SUCCESS : SOX_EOF;
}
/* Flow data through the effects chain until an effect or callback gives EOF */
int sox_flow_effects(sox_effects_chain_t * chain, int (* callback)(sox_bool all_done, void * client_data), void * client_data)
{
int flow_status = SOX_SUCCESS;
size_t e, source_e = 0; /* effect indices */
size_t max_flows = 0;
sox_bool draining = sox_true;
for (e = 0; e < chain->length; ++e) {
sox_effect_t *effp = chain->effects[e];
effp->obuf =
lsx_realloc(effp->obuf, sox_globals.bufsiz * sizeof(*effp->obuf));
/* Memory will be freed by sox_delete_effect() later. */
/* Possibly there was already a buffer, if this is a used effect;
it may still contain samples in that case. */
if (effp->oend > sox_globals.bufsiz) {
lsx_warn("buffer size insufficient; buffered samples were dropped");
/* can only happen if bufsize has been reduced since the last run */
effp->obeg = effp->oend = 0;
}
max_flows = max(max_flows, effp->flows);
}
if (max_flows > 1) /* might need interleave buffer */
chain->il_buf = lsx_malloc(sox_globals.bufsiz * sizeof(sox_sample_t));
else
chain->il_buf = NULL;
/* Go through the effects, and if there are samples in one of the
buffers, deinterleave it (if necessary). */
for (e = 0; e + 1 < chain->length; e++) {
sox_effect_t *effp = chain->effects[e];
if (effp->oend > effp->obeg && chain->effects[e+1]->flows > 1) {
sox_sample_t *sw = chain->il_buf; chain->il_buf = effp->obuf; effp->obuf = sw;
deinterleave(chain->effects[e+1]->flows, effp->oend - effp->obeg,
chain->il_buf, effp->obuf, sox_globals.bufsiz, effp->obeg);
}
}
e = chain->length - 1;
while (source_e < chain->length) {
#define have_imin (e > 0 && e < chain->length && chain->effects[e - 1]->oend - chain->effects[e - 1]->obeg >= chain->effects[e]->imin)
size_t osize = chain->effects[e]->oend - chain->effects[e]->obeg;
if (e == source_e && (draining || !have_imin)) {
if (drain_effect(chain, e) == SOX_EOF) {
++source_e;
draining = sox_false;
}
} else if (have_imin && flow_effect(chain, e) == SOX_EOF) {
flow_status = SOX_EOF;
if (e == chain->length - 1)
break;
source_e = e;
draining = sox_true;
}
if (e < chain->length && chain->effects[e]->oend - chain->effects[e]->obeg > osize) /* False for output */
++e;
else if (e == source_e)
draining = sox_true;
else if (e < source_e)
e = source_e;
else
--e;
if (callback && callback(source_e == chain->length, client_data) != SOX_SUCCESS) {
flow_status = SOX_EOF; /* Client has requested to stop the flow. */
break;
}
}
/* If an effect's output buffer still has samples, and if it is
uninterleaved, then re-interleave it. Necessary since it might
be reused, and at that time possibly followed by an MCHAN effect. */
for (e = 0; e + 1 < chain->length; e++) {
sox_effect_t *effp = chain->effects[e];
if (effp->oend > effp->obeg && chain->effects[e+1]->flows > 1) {
sox_sample_t *sw = chain->il_buf; chain->il_buf = effp->obuf; effp->obuf = sw;
interleave(chain->effects[e+1]->flows, effp->oend - effp->obeg,
chain->il_buf, sox_globals.bufsiz, effp->obeg, effp->obuf);
}
}
free(chain->il_buf);
return flow_status;
}
sox_uint64_t sox_effects_clips(sox_effects_chain_t * chain)
{
size_t i, f;
uint64_t clips = 0;
for (i = 1; i < chain->length - 1; ++i)
for (f = 0; f < chain->effects[i][0].flows; ++f)
clips += chain->effects[i][f].clips;
return clips;
}
sox_uint64_t sox_stop_effect(sox_effect_t *effp)
{
size_t f;
uint64_t clips = 0;
for (f = 0; f < effp->flows; ++f) {
effp[f].handler.stop(&effp[f]);
clips += effp[f].clips;
}
return clips;
}
void sox_push_effect_last(sox_effects_chain_t *chain, sox_effect_t *effp)
{
if (chain->length == chain->table_size) {
chain->table_size += EFF_TABLE_STEP;
lsx_debug_more("sox_push_effect_last: extending effects table, "
"new size = %" PRIuPTR, chain->table_size);
lsx_revalloc(chain->effects, chain->table_size);
}
chain->effects[chain->length++] = effp;
} /* sox_push_effect_last */
sox_effect_t *sox_pop_effect_last(sox_effects_chain_t *chain)
{
if (chain->length > 0)
{
sox_effect_t *effp;
chain->length--;
effp = chain->effects[chain->length];
chain->effects[chain->length] = NULL;
return effp;
}
else
return NULL;
} /* sox_pop_effect_last */
/* Free resources related to effect.
* Note: This currently closes down the effect which might
* not be obvious from name.
*/
void sox_delete_effect(sox_effect_t *effp)
{
uint64_t clips;
size_t f;
if ((clips = sox_stop_effect(effp)) != 0)
lsx_warn("%s clipped %" PRIu64 " samples; decrease volume?",
effp->handler.name, clips);
if (effp->obeg != effp->oend)
lsx_debug("output buffer still held %" PRIuPTR " samples; dropped.",
(effp->oend - effp->obeg)/effp->out_signal.channels);
/* May or may not indicate a problem; it is normal if the user aborted
processing, or if an effect like "trim" stopped early. */
effp->handler.kill(effp); /* N.B. only one kill; not one per flow */
for (f = 0; f < effp->flows; ++f)
free(effp[f].priv);
free(effp->obuf);
free(effp);
}
void sox_delete_effect_last(sox_effects_chain_t *chain)
{
if (chain->length > 0)
{
chain->length--;
sox_delete_effect(chain->effects[chain->length]);
chain->effects[chain->length] = NULL;
}
} /* sox_delete_effect_last */
/* Remove all effects from the chain.
* Note: This currently closes down the effect which might
* not be obvious from name.
*/
void sox_delete_effects(sox_effects_chain_t * chain)
{
size_t e;
for (e = 0; e < chain->length; ++e) {
sox_delete_effect(chain->effects[e]);
chain->effects[e] = NULL;
}
chain->length = 0;
}
/*----------------------------- Effects library ------------------------------*/
static sox_effect_fn_t s_sox_effect_fns[] = {
#define EFFECT(f) lsx_##f##_effect_fn,
#include "effects.h"
#undef EFFECT
NULL
};
const sox_effect_fn_t*
sox_get_effect_fns(void)
{
return s_sox_effect_fns;
}
/* Find a named effect in the effects library */
sox_effect_handler_t const * sox_find_effect(char const * name)
{
int e;
sox_effect_fn_t const * fns = sox_get_effect_fns();
for (e = 0; fns[e]; ++e) {
const sox_effect_handler_t *eh = fns[e] ();
if (eh && eh->name && strcasecmp(eh->name, name) == 0)
return eh; /* Found it. */
}
return NULL;
}
/*----------------------------- Helper functions -----------------------------*/
/* interleave() parameters:
* flows: number of samples per wide sample
* length: number of samples to copy
* [pertaining to the (non-interleaved) source buffer:]
* from: start address
* bufsiz: total size
* offset: position at which to start reading
* [pertaining to the (interleaved) destination buffer:]
* to: start address
*/
static void interleave(size_t flows, size_t length, sox_sample_t *from,
size_t bufsiz, size_t offset, sox_sample_t *to)
{
size_t i;
const size_t wide_samples = length/flows;
const size_t flow_offs = bufsiz/flows;
from += offset/flows;
for (i = 0; i < wide_samples; i++) {
sox_sample_t *inner_from = from + i;
sox_sample_t *inner_to = to + i * flows;
size_t f;
for (f = 0; f < flows; f++) {
*inner_to++ = *inner_from;
inner_from += flow_offs;
}
}
}
/* deinterleave() parameters:
* flows: number of samples per wide sample
* length: number of samples to copy
* [pertaining to the (interleaved) source buffer:]
* from: start address
* [pertaining to the (non-interleaved) destination buffer:]
* to: start address
* bufsiz: total size
* offset: position at which to start writing
*/
static void deinterleave(size_t flows, size_t length, sox_sample_t *from,
sox_sample_t *to, size_t bufsiz, size_t offset)
{
const size_t wide_samples = length/flows;
const size_t flow_offs = bufsiz/flows;
size_t f;
to += offset/flows;
for (f = 0; f < flows; f++) {
sox_sample_t *inner_to = to + f*flow_offs;
sox_sample_t *inner_from = from + f;
size_t i = wide_samples;
while (i--) {
*inner_to++ = *inner_from;
inner_from += flows;
}
}
}

90
app/src/main/cpp/sox/effects.h
Unescape View File

@ -0,0 +1,90 @@
/* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* FIXME: generate this list automatically */
EFFECT(allpass)
EFFECT(band)
EFFECT(bandpass)
EFFECT(bandreject)
EFFECT(bass)
EFFECT(bend)
EFFECT(biquad)
EFFECT(chorus)
EFFECT(channels)
EFFECT(compand)
EFFECT(contrast)
EFFECT(dcshift)
EFFECT(deemph)
EFFECT(delay)
EFFECT(dft_filter) /* abstract */
EFFECT(dither)
EFFECT(divide)
EFFECT(downsample)
EFFECT(earwax)
EFFECT(echo)
EFFECT(echos)
EFFECT(equalizer)
EFFECT(fade)
EFFECT(fir)
EFFECT(firfit)
EFFECT(flanger)
EFFECT(gain)
EFFECT(highpass)
EFFECT(hilbert)
EFFECT(input)
#ifdef HAVE_LADSPA_H
EFFECT(ladspa)
#endif
EFFECT(loudness)
EFFECT(lowpass)
EFFECT(mcompand)
EFFECT(noiseprof)
EFFECT(noisered)
EFFECT(norm)
EFFECT(oops)
EFFECT(output)
EFFECT(overdrive)
EFFECT(pad)
EFFECT(phaser)
EFFECT(pitch)
EFFECT(rate)
EFFECT(remix)
EFFECT(repeat)
EFFECT(reverb)
EFFECT(reverse)
EFFECT(riaa)
EFFECT(silence)
EFFECT(sinc)
#ifdef HAVE_PNG
EFFECT(spectrogram)
#endif
EFFECT(speed)
#ifdef HAVE_SPEEXDSP
EFFECT(speexdsp)
#endif
EFFECT(splice)
EFFECT(stat)
EFFECT(stats)
EFFECT(stretch)
EFFECT(swap)
EFFECT(synth)
EFFECT(tempo)
EFFECT(treble)
EFFECT(tremolo)
EFFECT(trim)
EFFECT(upsample)
EFFECT(vad)
EFFECT(vol)

483
app/src/main/cpp/sox/effects_i.c
Unescape View File

@ -0,0 +1,483 @@
/* Implements a libSoX internal interface for implementing effects.
* All public functions & data are prefixed with lsx_ .
*
* Copyright (c) 2005-2012 Chris Bagwell and SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define LSX_EFF_ALIAS
#include "sox_i.h"
#include <string.h>
#include <ctype.h>
int lsx_usage(sox_effect_t * effp)
{
if (effp->handler.usage)
lsx_fail("usage: %s", effp->handler.usage);
else
lsx_fail("this effect takes no parameters");
return SOX_EOF;
}
char * lsx_usage_lines(char * * usage, char const * const * lines, size_t n)
{
if (!*usage) {
size_t i, len;
for (len = i = 0; i < n; len += strlen(lines[i++]) + 1);
*usage = lsx_malloc(len); /* FIXME: this memory will never be freed */
strcpy(*usage, lines[0]);
for (i = 1; i < n; ++i) {
strcat(*usage, "\n");
strcat(*usage, lines[i]);
}
}
return *usage;
}
static lsx_enum_item const s_lsx_wave_enum[] = {
LSX_ENUM_ITEM(SOX_WAVE_,SINE)
LSX_ENUM_ITEM(SOX_WAVE_,TRIANGLE)
{0, 0}};
lsx_enum_item const * lsx_get_wave_enum(void)
{
return s_lsx_wave_enum;
}
void lsx_generate_wave_table(
lsx_wave_t wave_type,
sox_data_t data_type,
void *table,
size_t table_size,
double min,
double max,
double phase)
{
uint32_t t;
uint32_t phase_offset = phase / M_PI / 2 * table_size + 0.5;
for (t = 0; t < table_size; t++)
{
uint32_t point = (t + phase_offset) % table_size;
double d;
switch (wave_type)
{
case SOX_WAVE_SINE:
d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
break;
case SOX_WAVE_TRIANGLE:
d = (double)point * 2 / table_size;
switch (4 * point / table_size)
{
case 0: d = d + 0.5; break;
case 1: case 2: d = 1.5 - d; break;
case 3: d = d - 1.5; break;
}
break;
default: /* Oops! FIXME */
d = 0.0; /* Make sure we have a value */
break;
}
d = d * (max - min) + min;
switch (data_type)
{
case SOX_FLOAT:
{
float *fp = (float *)table;
*fp++ = (float)d;
table = fp;
continue;
}
case SOX_DOUBLE:
{
double *dp = (double *)table;
*dp++ = d;
table = dp;
continue;
}
default: break;
}
d += d < 0? -0.5 : +0.5;
switch (data_type)
{
case SOX_SHORT:
{
short *sp = table;
*sp++ = (short)d;
table = sp;
continue;
}
case SOX_INT:
{
int *ip = table;
*ip++ = (int)d;
table = ip;
continue;
}
default: break;
}
}
}
/*
* lsx_parsesamples
*
* Parse a string for # of samples. The input consists of one or more
* parts, with '+' or '-' between them indicating if the sample count
* should be added to or subtracted from the previous value.
* If a part ends with a 's' then it is interpreted as a
* user-calculated # of samples.
* If a part contains ':' or '.' but no 'e' or if it ends with a 't'
* then it is treated as an amount of time. This is converted into
* seconds and fraction of seconds, then the sample rate is used to
* calculate # of samples.
* Parameter def specifies which interpretation should be the default
* for a bare number like "123". It can either be 't' or 's'.
* Returns NULL on error, pointer to next char to parse otherwise.
*/
static char const * parsesamples(sox_rate_t rate, const char *str0, uint64_t *samples, int def, int combine);
char const * lsx_parsesamples(sox_rate_t rate, const char *str0, uint64_t *samples, int def)
{
*samples = 0;
return parsesamples(rate, str0, samples, def, '+');
}
static char const * parsesamples(sox_rate_t rate, const char *str0, uint64_t *samples, int def, int combine)
{
char * str = (char *)str0;
do {
uint64_t samples_part;
sox_bool found_samples = sox_false, found_time = sox_false;
char const * end;
char const * pos;
sox_bool found_colon, found_dot, found_e;
for (;*str == ' '; ++str);
for (end = str; *end && strchr("0123456789:.ets", *end); ++end);
if (end == str)
return NULL; /* error: empty input */
pos = strchr(str, ':');
found_colon = pos && pos < end;
pos = strchr(str, '.');
found_dot = pos && pos < end;
pos = strchr(str, 'e');
found_e = pos && pos < end;
if (found_colon || (found_dot && !found_e) || *(end-1) == 't')
found_time = sox_true;
else if (*(end-1) == 's')
found_samples = sox_true;
if (found_time || (def == 't' && !found_samples)) {
int i;
if (found_e)
return NULL; /* error: e notation in time */
for (samples_part = 0, i = 0; *str != '.' && i < 3; ++i) {
char * last_str = str;
long part = strtol(str, &str, 10);
if (!i && str == last_str)
return NULL; /* error: empty first component */
samples_part += rate * part;
if (i < 2) {
if (*str != ':')
break;
++str;
samples_part *= 60;
}
}
if (*str == '.') {
char * last_str = str;
double part = strtod(str, &str);
if (str == last_str)
return NULL; /* error: empty fractional part */
samples_part += rate * part + .5;
}
if (*str == 't')
str++;
} else {
char * last_str = str;
double part = strtod(str, &str);
if (str == last_str)
return NULL; /* error: no sample count */
samples_part = part + .5;
if (*str == 's')
str++;
}
if (str != end)
return NULL; /* error: trailing characters */
switch (combine) {
case '+': *samples += samples_part; break;
case '-': *samples = samples_part <= *samples ?
*samples - samples_part : 0;
break;
}
combine = '\0';
if (*str && strchr("+-", *str))
combine = *str++;
} while (combine);
return str;
}
#if 0
#include <assert.h>
#define TEST(st, samp, len) \
str = st; \
next = lsx_parsesamples(10000, str, &samples, 't'); \
assert(samples == samp && next == str + len);
int main(int argc, char * * argv)
{
char const * str, * next;
uint64_t samples;
TEST("0" , 0, 1)
TEST("1" , 10000, 1)
TEST("0s" , 0, 2)
TEST("0s,", 0, 2)
TEST("0s/", 0, 2)
TEST("0s@", 0, 2)
TEST("0t" , 0, 2)
TEST("0t,", 0, 2)
TEST("0t/", 0, 2)
TEST("0t@", 0, 2)
TEST("1s" , 1, 2)
TEST("1s,", 1, 2)
TEST("1s/", 1, 2)
TEST("1s@", 1, 2)
TEST(" 01s" , 1, 4)
TEST("1e6s" , 1000000, 4)
TEST("1t" , 10000, 2)
TEST("1t,", 10000, 2)
TEST("1t/", 10000, 2)
TEST("1t@", 10000, 2)
TEST("1.1t" , 11000, 4)
TEST("1.1t,", 11000, 4)
TEST("1.1t/", 11000, 4)
TEST("1.1t@", 11000, 4)
assert(!lsx_parsesamples(10000, "1e6t", &samples, 't'));
TEST(".0", 0, 2)
TEST("0.0", 0, 3)
TEST("0:0.0", 0, 5)
TEST("0:0:0.0", 0, 7)
TEST(".1", 1000, 2)
TEST(".10", 1000, 3)
TEST("0.1", 1000, 3)
TEST("1.1", 11000, 3)
TEST("1:1.1", 611000, 5)
TEST("1:1:1.1", 36611000, 7)
TEST("1:1", 610000, 3)
TEST("1:01", 610000, 4)
TEST("1:1:1", 36610000, 5)
TEST("1:", 600000, 2)
TEST("1::", 36000000, 3)
TEST("0.444444", 4444, 8)
TEST("0.555555", 5556, 8)
assert(!lsx_parsesamples(10000, "x", &samples, 't'));
TEST("1:23+37", 1200000, 7)
TEST("12t+12s", 120012, 7)
TEST("1e6s-10", 900000, 7)
TEST("10-2:00", 0, 7)
TEST("123-45+12s+2:00-3e3s@foo", 1977012, 20)
TEST("1\0" "2", 10000, 1)
return 0;
}
#endif
/*
* lsx_parseposition
*
* Parse a string for an audio position. Similar to lsx_parsesamples
* above, but an initial '=', '+' or '-' indicates that the specified time
* is relative to the start of audio, last used position or end of audio,
* respectively. Parameter def states which of these is the default.
* Parameters latest and end are the positions to which '+' and '-' relate;
* end may be SOX_UNKNOWN_LEN, in which case "-0" is the only valid
* end-relative input and will result in a position of SOX_UNKNOWN_LEN.
* Other parameters and return value are the same as for lsx_parsesamples.
*
* A test parse that only checks for valid syntax can be done by
* specifying samples = NULL. If this passes, a later reparse of the same
* input will only fail if it is relative to the end ("-"), not "-0", and
* the end position is unknown.
*/
char const * lsx_parseposition(sox_rate_t rate, const char *str0, uint64_t *samples, uint64_t latest, uint64_t end, int def)
{
char *str = (char *)str0;
char anchor, combine;
if (!strchr("+-=", def))
return NULL; /* error: invalid default anchor */
anchor = def;
if (*str && strchr("+-=", *str))
anchor = *str++;
combine = '+';
if (strchr("+-", anchor)) {
combine = anchor;
if (*str && strchr("+-", *str))
combine = *str++;
}
if (!samples) {
/* dummy parse, syntax checking only */
uint64_t dummy = 0;
return parsesamples(0., str, &dummy, 't', '+');
}
switch (anchor) {
case '=': *samples = 0; break;
case '+': *samples = latest; break;
case '-': *samples = end; break;
}
if (anchor == '-' && end == SOX_UNKNOWN_LEN) {
/* "-0" only valid input here */
char const *l;
for (l = str; *l && strchr("0123456789:.ets+-", *l); ++l);
if (l == str+1 && *str == '0') {
/* *samples already set to SOX_UNKNOWN_LEN */
return l;
}
return NULL; /* error: end-relative position, but end unknown */
}
return parsesamples(rate, str, samples, 't', combine);
}
/* a note is given as an int,
* 0 => 440 Hz = A
* >0 => number of half notes 'up',
* <0 => number of half notes down,
* example 12 => A of next octave, 880Hz
*
* calculated by freq = 440Hz * 2**(note/12)
*/
static double calc_note_freq(double note, int key)
{
if (key != INT_MAX) { /* Just intonation. */
static const int n[] = {16, 9, 6, 5, 4, 7}; /* Numerator. */
static const int d[] = {15, 8, 5, 4, 3, 5}; /* Denominator. */
static double j[13]; /* Just semitones */
int i, m = floor(note);
if (!j[1]) for (i = 1; i <= 12; ++i)
j[i] = i <= 6? log((double)n[i - 1] / d[i - 1]) / log(2.) : 1 - j[12 - i];
note -= m;
m -= key = m - ((INT_MAX / 2 - ((INT_MAX / 2) % 12) + m - key) % 12);
return 440 * pow(2., key / 12. + j[m] + (j[m + 1] - j[m]) * note);
}
return 440 * pow(2., note / 12);
}
int lsx_parse_note(char const * text, char * * end_ptr)
{
int result = INT_MAX;
if (*text >= 'A' && *text <= 'G') {
result = (int)(5/3. * (*text++ - 'A') + 9.5) % 12 - 9;
if (*text == 'b') {--result; ++text;}
else if (*text == '#') {++result; ++text;}
if (isdigit((unsigned char)*text))
result += 12 * (*text++ - '4');
}
*end_ptr = (char *)text;
return result;
}
/* Read string 'text' and convert to frequency.
* 'text' can be a positive number which is the frequency in Hz.
* If 'text' starts with a '%' and a following number the corresponding
* note is calculated.
* Return -1 on error.
*/
double lsx_parse_frequency_k(char const * text, char * * end_ptr, int key)
{
double result;
if (*text == '%') {
result = strtod(text + 1, end_ptr);
if (*end_ptr == text + 1)
return -1;
return calc_note_freq(result, key);
}
if (*text >= 'A' && *text <= 'G') {
int result2 = lsx_parse_note(text, end_ptr);
return result2 == INT_MAX? - 1 : calc_note_freq((double)result2, key);
}
result = strtod(text, end_ptr);
if (end_ptr) {
if (*end_ptr == text)
return -1;
if (**end_ptr == 'k') {
result *= 1000;
++*end_ptr;
}
}
return result < 0 ? -1 : result;
}
FILE * lsx_open_input_file(sox_effect_t * effp, char const * filename, sox_bool text_mode)
{
FILE * file;
if (!filename || !strcmp(filename, "-")) {
if (effp->global_info->global_info->stdin_in_use_by) {
lsx_fail("stdin already in use by `%s'", effp->global_info->global_info->stdin_in_use_by);
return NULL;
}
effp->global_info->global_info->stdin_in_use_by = effp->handler.name;
file = stdin;
}
else if (!(file = fopen(filename, text_mode ? "r" : "rb"))) {
lsx_fail("couldn't open file %s: %s", filename, strerror(errno));
return NULL;
}
return file;
}
int lsx_effects_init(void)
{
init_fft_cache();
return SOX_SUCCESS;
}
int lsx_effects_quit(void)
{
clear_fft_cache();
return SOX_SUCCESS;
}

645
app/src/main/cpp/sox/effects_i_dsp.c
Unescape View File

@ -0,0 +1,645 @@
/* libSoX internal DSP functions.
* All public functions & data are prefixed with lsx_ .
*
* Copyright (c) 2008,2012 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifdef NDEBUG /* Enable assert always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox_i.h"
#include <assert.h>
#include <string.h>
/* Concurrent Control with "Readers" and "Writers", P.J. Courtois et al, 1971:*/
#if defined HAVE_OPENMP
typedef struct {
int readcount, writecount; /* initial value = 0 */
omp_lock_t mutex_1, mutex_2, mutex_3, w, r; /* initial value = 1 */
} ccrw2_t; /* Problem #2: `writers-preference' */
#define ccrw2_become_reader(p) do {\
omp_set_lock(&p.mutex_3);\
omp_set_lock(&p.r);\
omp_set_lock(&p.mutex_1);\
if (++p.readcount == 1) omp_set_lock(&p.w);\
omp_unset_lock(&p.mutex_1);\
omp_unset_lock(&p.r);\
omp_unset_lock(&p.mutex_3);\
} while (0)
#define ccrw2_cease_reading(p) do {\
omp_set_lock(&p.mutex_1);\
if (!--p.readcount) omp_unset_lock(&p.w);\
omp_unset_lock(&p.mutex_1);\
} while (0)
#define ccrw2_become_writer(p) do {\
omp_set_lock(&p.mutex_2);\
if (++p.writecount == 1) omp_set_lock(&p.r);\
omp_unset_lock(&p.mutex_2);\
omp_set_lock(&p.w);\
} while (0)
#define ccrw2_cease_writing(p) do {\
omp_unset_lock(&p.w);\
omp_set_lock(&p.mutex_2);\
if (!--p.writecount) omp_unset_lock(&p.r);\
omp_unset_lock(&p.mutex_2);\
} while (0)
#define ccrw2_init(p) do {\
omp_init_lock(&p.mutex_1);\
omp_init_lock(&p.mutex_2);\
omp_init_lock(&p.mutex_3);\
omp_init_lock(&p.w);\
omp_init_lock(&p.r);\
} while (0)
#define ccrw2_clear(p) do {\
omp_destroy_lock(&p.r);\
omp_destroy_lock(&p.w);\
omp_destroy_lock(&p.mutex_3);\
omp_destroy_lock(&p.mutex_2);\
omp_destroy_lock(&p.mutex_1);\
} while (0)
#else
#define ccrw2_become_reader(x) (void)0
#define ccrw2_cease_reading(x) (void)0
#define ccrw2_become_writer(x) (void)0
#define ccrw2_cease_writing(x) (void)0
#define ccrw2_init(x) (void)0
#define ccrw2_clear(x) (void)0
#endif /* HAVE_OPENMP */
/* Numerical Recipes cubic spline: */
void lsx_prepare_spline3(double const * x, double const * y, int n,
double start_1d, double end_1d, double * y_2d)
{
double p, qn, sig, un, * u = lsx_malloc((n - 1) * sizeof(*u));
int i;
if (start_1d == HUGE_VAL)
y_2d[0] = u[0] = 0; /* Start with natural spline or */
else { /* set the start first derivative */
y_2d[0] = -.5;
u[0] = (3 / (x[1] - x[0])) * ((y[1] - y[0]) / (x[1] - x[0]) - start_1d);
}
for (i = 1; i < n - 1; ++i) {
sig = (x[i] - x[i - 1]) / (x[i + 1] - x[i - 1]);
p = sig * y_2d[i - 1] + 2;
y_2d[i] = (sig - 1) / p;
u[i] = (y[i + 1] - y[i]) / (x[i + 1] - x[i]) -
(y[i] - y[i - 1]) / (x[i] - x[i - 1]);
u[i] = (6 * u[i] / (x[i + 1] - x[i - 1]) - sig * u[i - 1]) / p;
}
if (end_1d == HUGE_VAL)
qn = un = 0; /* End with natural spline or */
else { /* set the end first derivative */
qn = .5;
un = 3 / (x[n - 1] - x[n - 2]) * (end_1d - (y[n - 1] - y[n - 2]) / (x[n - 1] - x[n - 2]));
}
y_2d[n - 1] = (un - qn * u[n - 2]) / (qn * y_2d[n - 2] + 1);
for (i = n - 2; i >= 0; --i)
y_2d[i] = y_2d[i] * y_2d[i + 1] + u[i];
free(u);
}
double lsx_spline3(double const * x, double const * y, double const * y_2d,
int n, double x1)
{
int t, i[2] = {0, 0};
double d, a, b;
for (i[1] = n - 1; i[1] - i[0] > 1; t = (i[1] + i[0]) >> 1, i[x[t] > x1] = t);
d = x[i[1]] - x[i[0]];
assert(d != 0);
a = (x[i[1]] - x1) / d;
b = (x1 - x[i[0]]) / d;
return a * y[i[0]] + b * y[i[1]] +
((a * a * a - a) * y_2d[i[0]] + (b * b * b - b) * y_2d[i[1]]) * d * d / 6;
}
double lsx_bessel_I_0(double x)
{
double term = 1, sum = 1, last_sum, x2 = x / 2;
int i = 1;
do {
double y = x2 / i++;
last_sum = sum, sum += term *= y * y;
} while (sum != last_sum);
return sum;
}
int lsx_set_dft_length(int num_taps) /* Set to 4 x nearest power of 2 */
{ /* or half of that if danger of causing too many cache misses. */
int min = sox_globals.log2_dft_min_size;
double d = log((double)num_taps) / log(2.);
return 1 << range_limit((int)(d + 2.77), min, max((int)(d + 1.77), 17));
}
#include "fft4g.h"
static int * lsx_fft_br;
static double * lsx_fft_sc;
static int fft_len = -1;
#if defined HAVE_OPENMP
static ccrw2_t fft_cache_ccrw;
#endif
void init_fft_cache(void)
{
assert(lsx_fft_br == NULL);
assert(lsx_fft_sc == NULL);
assert(fft_len == -1);
ccrw2_init(fft_cache_ccrw);
fft_len = 0;
}
void clear_fft_cache(void)
{
assert(fft_len >= 0);
ccrw2_clear(fft_cache_ccrw);
free(lsx_fft_br);
free(lsx_fft_sc);
lsx_fft_sc = NULL;
lsx_fft_br = NULL;
fft_len = -1;
}
static sox_bool update_fft_cache(int len)
{
assert(lsx_is_power_of_2(len));
assert(fft_len >= 0);
ccrw2_become_reader(fft_cache_ccrw);
if (len > fft_len) {
ccrw2_cease_reading(fft_cache_ccrw);
ccrw2_become_writer(fft_cache_ccrw);
if (len > fft_len) {
int old_n = fft_len;
fft_len = len;
lsx_fft_br = lsx_realloc(lsx_fft_br, dft_br_len(fft_len) * sizeof(*lsx_fft_br));
lsx_fft_sc = lsx_realloc(lsx_fft_sc, dft_sc_len(fft_len) * sizeof(*lsx_fft_sc));
if (!old_n)
lsx_fft_br[0] = 0;
return sox_true;
}
ccrw2_cease_writing(fft_cache_ccrw);
ccrw2_become_reader(fft_cache_ccrw);
}
return sox_false;
}
static void done_with_fft_cache(sox_bool is_writer)
{
if (is_writer)
ccrw2_cease_writing(fft_cache_ccrw);
else ccrw2_cease_reading(fft_cache_ccrw);
}
void lsx_safe_rdft(int len, int type, double * d)
{
sox_bool is_writer = update_fft_cache(len);
lsx_rdft(len, type, d, lsx_fft_br, lsx_fft_sc);
done_with_fft_cache(is_writer);
}
void lsx_safe_cdft(int len, int type, double * d)
{
sox_bool is_writer = update_fft_cache(len);
lsx_cdft(len, type, d, lsx_fft_br, lsx_fft_sc);
done_with_fft_cache(is_writer);
}
void lsx_power_spectrum(int n, double const * in, double * out)
{
int i;
double * work = lsx_memdup(in, n * sizeof(*work));
lsx_safe_rdft(n, 1, work);
out[0] = sqr(work[0]);
for (i = 2; i < n; i += 2)
out[i >> 1] = sqr(work[i]) + sqr(work[i + 1]);
out[i >> 1] = sqr(work[1]);
free(work);
}
void lsx_power_spectrum_f(int n, float const * in, float * out)
{
int i;
double * work = lsx_malloc(n * sizeof(*work));
for (i = 0; i< n; ++i) work[i] = in[i];
lsx_safe_rdft(n, 1, work);
out[0] = sqr(work[0]);
for (i = 2; i < n; i += 2)
out[i >> 1] = sqr(work[i]) + sqr(work[i + 1]);
out[i >> 1] = sqr(work[1]);
free(work);
}
void lsx_apply_hann_f(float h[], const int num_points)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
double x = 2 * M_PI * i / m;
h[i] *= .5 - .5 * cos(x);
}
}
void lsx_apply_hann(double h[], const int num_points)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
double x = 2 * M_PI * i / m;
h[i] *= .5 - .5 * cos(x);
}
}
void lsx_apply_hamming(double h[], const int num_points)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
double x = 2 * M_PI * i / m;
h[i] *= .53836 - .46164 * cos(x);
}
}
void lsx_apply_bartlett(double h[], const int num_points)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
h[i] *= 2. / m * (m / 2. - fabs(i - m / 2.));
}
}
void lsx_apply_blackman(double h[], const int num_points, double alpha /*.16*/)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
double x = 2 * M_PI * i / m;
h[i] *= (1 - alpha) *.5 - .5 * cos(x) + alpha * .5 * cos(2 * x);
}
}
void lsx_apply_blackman_nutall(double h[], const int num_points)
{
int i, m = num_points - 1;
for (i = 0; i < num_points; ++i) {
double x = 2 * M_PI * i / m;
h[i] *= .3635819 - .4891775 * cos(x) + .1365995 * cos(2 * x) - .0106411 * cos(3 * x);
}
}
double lsx_kaiser_beta(double att, double tr_bw)
{
if (att >= 60) {
static const double coefs[][4] = {
{-6.784957e-10,1.02856e-05,0.1087556,-0.8988365+.001},
{-6.897885e-10,1.027433e-05,0.10876,-0.8994658+.002},
{-1.000683e-09,1.030092e-05,0.1087677,-0.9007898+.003},
{-3.654474e-10,1.040631e-05,0.1087085,-0.8977766+.006},
{8.106988e-09,6.983091e-06,0.1091387,-0.9172048+.015},
{9.519571e-09,7.272678e-06,0.1090068,-0.9140768+.025},
{-5.626821e-09,1.342186e-05,0.1083999,-0.9065452+.05},
{-9.965946e-08,5.073548e-05,0.1040967,-0.7672778+.085},
{1.604808e-07,-5.856462e-05,0.1185998,-1.34824+.1},
{-1.511964e-07,6.363034e-05,0.1064627,-0.9876665+.18},
};
double realm = log(tr_bw/.0005)/log(2.);
double const * c0 = coefs[range_limit( (int)realm, 0, (int)array_length(coefs)-1)];
double const * c1 = coefs[range_limit(1+(int)realm, 0, (int)array_length(coefs)-1)];
double b0 = ((c0[0]*att + c0[1])*att + c0[2])*att + c0[3];
double b1 = ((c1[0]*att + c1[1])*att + c1[2])*att + c1[3];
return b0 + (b1 - b0) * (realm - (int)realm);
}
if (att > 50 ) return .1102 * (att - 8.7);
if (att > 20.96) return .58417 * pow(att -20.96, .4) + .07886 * (att - 20.96);
return 0;
}
void lsx_apply_kaiser(double h[], const int num_points, double beta)
{
int i, m = num_points - 1;
for (i = 0; i <= m; ++i) {
double x = 2. * i / m - 1;
h[i] *= lsx_bessel_I_0(beta * sqrt(1 - x * x)) / lsx_bessel_I_0(beta);
}
}
void lsx_apply_dolph(double h[], const int N, double att)
{
double b = cosh(acosh(pow(10., att/20)) / (N-1)), sum, t, c, norm = 0;
int i, j;
for (c = 1 - 1 / (b*b), i = (N-1) / 2; i >= 0; --i) {
for (sum = !i, b = t = j = 1; j <= i && sum != t; b *= (i-j) * (1./j), ++j)
t = sum, sum += (b *= c * (N - i - j) * (1./j));
sum /= (N - 1 - i), sum /= (norm = norm? norm : sum);
h[i] *= sum, h[N - 1 - i] *= sum;
}
}
double * lsx_make_lpf(int num_taps, double Fc, double beta, double rho,
double scale, sox_bool dc_norm)
{
int i, m = num_taps - 1;
double * h = calloc(num_taps, sizeof(*h)), sum = 0;
double mult = scale / lsx_bessel_I_0(beta), mult1 = 1 / (.5 * m + rho);
assert(Fc >= 0 && Fc <= 1);
lsx_debug("make_lpf(n=%i Fc=%.7g β=%g ρ=%g dc-norm=%i scale=%g)", num_taps, Fc, beta, rho, dc_norm, scale);
if (!h)
return NULL;
for (i = 0; i <= m / 2; ++i) {
double z = i - .5 * m, x = z * M_PI, y = z * mult1;
h[i] = x? sin(Fc * x) / x : Fc;
sum += h[i] *= lsx_bessel_I_0(beta * sqrt(1 - y * y)) * mult;
if (m - i != i)
sum += h[m - i] = h[i];
}
for (i = 0; dc_norm && i < num_taps; ++i) h[i] *= scale / sum;
return h;
}
void lsx_kaiser_params(double att, double Fc, double tr_bw, double * beta, int * num_taps)
{
*beta = *beta < 0? lsx_kaiser_beta(att, tr_bw * .5 / Fc): *beta;
att = att < 60? (att - 7.95) / (2.285 * M_PI * 2) :
((.0007528358-1.577737e-05**beta)**beta+.6248022)**beta+.06186902;
*num_taps = !*num_taps? ceil(att/tr_bw + 1) : *num_taps;
}
double * lsx_design_lpf(
double Fp, /* End of pass-band */
double Fs, /* Start of stop-band */
double Fn, /* Nyquist freq; e.g. 0.5, 1, PI */
double att, /* Stop-band attenuation in dB */
int * num_taps, /* 0: value will be estimated */
int k, /* >0: number of phases; <0: num_taps ≡ 1 (mod -k) */
double beta) /* <0: value will be estimated */
{
int n = *num_taps, phases = max(k, 1), modulo = max(-k, 1);
double tr_bw, Fc, rho = phases == 1? .5 : att < 120? .63 : .75;
Fp /= fabs(Fn), Fs /= fabs(Fn); /* Normalise to Fn = 1 */
tr_bw = .5 * (Fs - Fp); /* Transition band-width: 6dB to stop points */
tr_bw /= phases, Fs /= phases;
tr_bw = min(tr_bw, .5 * Fs);
Fc = Fs - tr_bw;
assert(Fc - tr_bw >= 0);
lsx_kaiser_params(att, Fc, tr_bw, &beta, num_taps);
if (!n)
*num_taps = phases > 1? *num_taps / phases * phases + phases - 1 : (*num_taps + modulo - 2) / modulo * modulo + 1;
return Fn < 0? 0 : lsx_make_lpf(
*num_taps, Fc, beta, rho, (double)phases, sox_false);
}
static double safe_log(double x)
{
assert(x >= 0);
if (x)
return log(x);
lsx_debug("log(0)");
return -26;
}
void lsx_fir_to_phase(double * * h, int * len, int * post_len, double phase)
{
double * pi_wraps, * work, phase1 = (phase > 50 ? 100 - phase : phase) / 50;
int i, work_len, begin, end, imp_peak = 0, peak = 0;
double imp_sum = 0, peak_imp_sum = 0;
double prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
work = lsx_calloc((size_t)work_len + 2, sizeof(*work)); /* +2: (UN)PACK */
pi_wraps = lsx_malloc((((size_t)work_len + 2) / 2) * sizeof(*pi_wraps));
memcpy(work, *h, *len * sizeof(*work));
lsx_safe_rdft(work_len, 1, work); /* Cepstral: */
LSX_UNPACK(work, work_len);
for (i = 0; i <= work_len; i += 2) {
double angle = atan2(work[i + 1], work[i]);
double detect = 2 * M_PI;
double delta = angle - prev_angle2;
double adjust = detect * ((delta < -detect * .7) - (delta > detect * .7));
prev_angle2 = angle;
cum_2pi += adjust;
angle += cum_2pi;
detect = M_PI;
delta = angle - prev_angle1;
adjust = detect * ((delta < -detect * .7) - (delta > detect * .7));
prev_angle1 = angle;
cum_1pi += fabs(adjust); /* fabs for when 2pi and 1pi have combined */
pi_wraps[i >> 1] = cum_1pi;
work[i] = safe_log(sqrt(sqr(work[i]) + sqr(work[i + 1])));
work[i + 1] = 0;
}
LSX_PACK(work, work_len);
lsx_safe_rdft(work_len, -1, work);
for (i = 0; i < work_len; ++i) work[i] *= 2. / work_len;
for (i = 1; i < work_len / 2; ++i) { /* Window to reject acausal components */
work[i] *= 2;
work[i + work_len / 2] = 0;
}
lsx_safe_rdft(work_len, 1, work);
for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
work[i + 1] = phase1 * i / work_len * pi_wraps[work_len >> 1] +
(1 - phase1) * (work[i + 1] + pi_wraps[i >> 1]) - pi_wraps[i >> 1];
work[0] = exp(work[0]), work[1] = exp(work[1]);
for (i = 2; i < work_len; i += 2) {
double x = exp(work[i]);
work[i ] = x * cos(work[i + 1]);
work[i + 1] = x * sin(work[i + 1]);
}
lsx_safe_rdft(work_len, -1, work);
for (i = 0; i < work_len; ++i) work[i] *= 2. / work_len;
/* Find peak pos. */
for (i = 0; i <= (int)(pi_wraps[work_len >> 1] / M_PI + .5); ++i) {
imp_sum += work[i];
if (fabs(imp_sum) > fabs(peak_imp_sum)) {
peak_imp_sum = imp_sum;
peak = i;
}
if (work[i] > work[imp_peak]) /* For debug check only */
imp_peak = i;
}
while (peak && fabs(work[peak-1]) > fabs(work[peak]) && work[peak-1] * work[peak] > 0)
--peak;
if (!phase1)
begin = 0;
else if (phase1 == 1)
begin = peak - *len / 2;
else {
begin = (.997 - (2 - phase1) * .22) * *len + .5;
end = (.997 + (0 - phase1) * .22) * *len + .5;
begin = peak - (begin & ~3);
end = peak + 1 + ((end + 3) & ~3);
*len = end - begin;
*h = lsx_realloc(*h, *len * sizeof(**h));
}
for (i = 0; i < *len; ++i) (*h)[i] =
work[(begin + (phase > 50 ? *len - 1 - i : i) + work_len) & (work_len - 1)];
*post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
lsx_debug("nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)",
pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
work[imp_peak], *len, *post_len, 100 - 100. * *post_len / (*len - 1));
free(pi_wraps), free(work);
}
void lsx_plot_fir(double * h, int num_points, sox_rate_t rate, sox_plot_t type, char const * title, double y1, double y2)
{
int i, N = lsx_set_dft_length(num_points);
if (type == sox_plot_gnuplot) {
double * h1 = lsx_calloc(N, sizeof(*h1));
double * H = lsx_malloc((N / 2 + 1) * sizeof(*H));
memcpy(h1, h, sizeof(*h1) * num_points);
lsx_power_spectrum(N, h1, H);
printf(
"# gnuplot file\n"
"set title '%s'\n"
"set xlabel 'Frequency (Hz)'\n"
"set ylabel 'Amplitude Response (dB)'\n"
"set grid xtics ytics\n"
"set key off\n"
"plot '-' with lines\n"
, title);
for (i = 0; i <= N/2; ++i)
printf("%g %g\n", i * rate / N, 10 * log10(H[i]));
printf(
"e\n"
"pause -1 'Hit return to continue'\n");
free(H);
free(h1);
}
else if (type == sox_plot_octave) {
printf("%% GNU Octave file (may also work with MATLAB(R) )\nb=[");
for (i = 0; i < num_points; ++i)
printf("%24.16e\n", h[i]);
printf("];\n"
"[h,w]=freqz(b,1,%i);\n"
"plot(%g*w/pi,20*log10(h))\n"
"title('%s')\n"
"xlabel('Frequency (Hz)')\n"
"ylabel('Amplitude Response (dB)')\n"
"grid on\n"
"axis([0 %g %g %g])\n"
"disp('Hit return to continue')\n"
"pause\n"
, N, rate * .5, title, rate * .5, y1, y2);
}
else if (type == sox_plot_data) {
printf("# %s\n"
"# FIR filter\n"
"# rate: %g\n"
"# name: b\n"
"# type: matrix\n"
"# rows: %i\n"
"# columns: 1\n", title, rate, num_points);
for (i = 0; i < num_points; ++i)
printf("%24.16e\n", h[i]);
}
}
#if HAVE_FENV_H
#include <fenv.h>
#if defined FE_INVALID
#if HAVE_LRINT && LONG_MAX == 2147483647
#define lrint32 lrint
#elif defined __GNUC__ && defined __x86_64__
#define lrint32 lrint32
static __inline sox_int32_t lrint32(double input) {
sox_int32_t result;
__asm__ __volatile__("fistpl %0": "=m"(result): "t"(input): "st");
return result;
}
#endif
#endif
#endif
#if defined lrint32
#define _ dest[i] = lrint32(src[i]), ++i,
#pragma STDC FENV_ACCESS ON
static void rint_clip(sox_sample_t * const dest, double const * const src,
size_t i, size_t const n, sox_uint64_t * const clips)
{
for (; i < n; ++i) {
dest[i] = lrint32(src[i]);
if (fetestexcept(FE_INVALID)) {
feclearexcept(FE_INVALID);
dest[i] = src[i] > 0? SOX_SAMPLE_MAX : SOX_SAMPLE_MIN;
++*clips;
}
}
}
void lsx_save_samples(sox_sample_t * const dest, double const * const src,
size_t const n, sox_uint64_t * const clips)
{
size_t i;
feclearexcept(FE_INVALID);
for (i = 0; i < (n & ~7);) {
_ _ _ _ _ _ _ _ 0;
if (fetestexcept(FE_INVALID)) {
feclearexcept(FE_INVALID);
rint_clip(dest, src, i - 8, i, clips);
}
}
rint_clip(dest, src, i, n, clips);
}
void lsx_load_samples(double * const dest, sox_sample_t const * const src,
size_t const n)
{
size_t i;
for (i = 0; i < n; ++i)
dest[i] = src[i];
}
#pragma STDC FENV_ACCESS OFF
#undef _
#else
void lsx_save_samples(sox_sample_t * const dest, double const * const src,
size_t const n, sox_uint64_t * const clips)
{
SOX_SAMPLE_LOCALS;
size_t i;
for (i = 0; i < n; ++i)
dest[i] = SOX_FLOAT_64BIT_TO_SAMPLE(src[i], *clips);
}
void lsx_load_samples(double * const dest, sox_sample_t const * const src,
size_t const n)
{
size_t i;
for (i = 0; i < n; ++i)
dest[i] = SOX_SAMPLE_TO_FLOAT_64BIT(src[i],);
}
#endif

97
app/src/main/cpp/sox/example0.c
Unescape View File

@ -0,0 +1,97 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2007-8 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include <stdlib.h>
#include <stdio.h>
#include <assert.h>
/*
* Reads input file, applies vol & flanger effects, stores in output file.
* E.g. example1 monkey.au monkey.aiff
*/
int main(int argc, char * argv[])
{
static sox_format_t * in, * out; /* input and output files */
sox_effects_chain_t * chain;
sox_effect_t * e;
char * args[10];
assert(argc == 3);
/* All libSoX applications must start by initialising the SoX library */
assert(sox_init() == SOX_SUCCESS);
/* Open the input file (with default parameters) */
assert((in = sox_open_read(argv[1], NULL, NULL, NULL)));
/* Open the output file; we must specify the output signal characteristics.
* Since we are using only simple effects, they are the same as the input
* file characteristics */
assert((out = sox_open_write(argv[2], &in->signal, NULL, NULL, NULL, NULL)));
/* Create an effects chain; some effects need to know about the input
* or output file encoding so we provide that information here */
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
/* The first effect in the effect chain must be something that can source
* samples; in this case, we use the built-in handler that inputs
* data from an audio file */
e = sox_create_effect(sox_find_effect("input"));
args[0] = (char *)in, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
/* This becomes the first `effect' in the chain */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Create the `vol' effect, and initialise it with the desired parameters: */
e = sox_create_effect(sox_find_effect("vol"));
args[0] = "3dB", assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
/* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Create the `flanger' effect, and initialise it with default parameters: */
e = sox_create_effect(sox_find_effect("flanger"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
/* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* The last effect in the effect chain must be something that only consumes
* samples; in this case, we use the built-in handler that outputs
* data to an audio file */
e = sox_create_effect(sox_find_effect("output"));
args[0] = (char *)out, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Flow samples through the effects processing chain until EOF is reached */
sox_flow_effects(chain, NULL, NULL);
/* All done; tidy up: */
sox_delete_effects_chain(chain);
sox_close(out);
sox_close(in);
sox_quit();
return 0;
}

167
app/src/main/cpp/sox/example1.c
Unescape View File

@ -0,0 +1,167 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2007-8 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include <stdlib.h>
#include <stdio.h>
#include <assert.h>
static sox_format_t * in, * out; /* input and output files */
/* The function that will be called to input samples into the effects chain.
* In this example, we get samples to process from a SoX-openned audio file.
* In a different application, they might be generated or come from a different
* part of the application. */
static int input_drain(
sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
(void)effp; /* This parameter is not needed in this example */
/* ensure that *osamp is a multiple of the number of channels. */
*osamp -= *osamp % effp->out_signal.channels;
/* Read up to *osamp samples into obuf; store the actual number read
* back to *osamp */
*osamp = sox_read(in, obuf, *osamp);
/* sox_read may return a number that is less than was requested; only if
* 0 samples is returned does it indicate that end-of-file has been reached
* or an error has occurred */
if (!*osamp && in->sox_errno)
fprintf(stderr, "%s: %s\n", in->filename, in->sox_errstr);
return *osamp? SOX_SUCCESS : SOX_EOF;
}
/* The function that will be called to output samples from the effects chain.
* In this example, we store the samples in a SoX-opened audio file.
* In a different application, they might perhaps be analysed in some way,
* or displayed as a wave-form */
static int output_flow(sox_effect_t *effp LSX_UNUSED, sox_sample_t const * ibuf,
sox_sample_t * obuf LSX_UNUSED, size_t * isamp, size_t * osamp)
{
/* Write out *isamp samples */
size_t len = sox_write(out, ibuf, *isamp);
/* len is the number of samples that were actually written out; if this is
* different to *isamp, then something has gone wrong--most often, it's
* out of disc space */
if (len != *isamp) {
fprintf(stderr, "%s: %s\n", out->filename, out->sox_errstr);
return SOX_EOF;
}
/* Outputting is the last `effect' in the effect chain so always passes
* 0 samples on to the next effect (as there isn't one!) */
*osamp = 0;
(void)effp; /* This parameter is not needed in this example */
return SOX_SUCCESS; /* All samples output successfully */
}
/* A `stub' effect handler to handle inputting samples to the effects
* chain; the only function needed for this example is `drain' */
static sox_effect_handler_t const * input_handler(void)
{
static sox_effect_handler_t handler = {
"input", NULL, SOX_EFF_MCHAN, NULL, NULL, NULL, input_drain, NULL, NULL, 0
};
return &handler;
}
/* A `stub' effect handler to handle outputting samples from the effects
* chain; the only function needed for this example is `flow' */
static sox_effect_handler_t const * output_handler(void)
{
static sox_effect_handler_t handler = {
"output", NULL, SOX_EFF_MCHAN, NULL, NULL, output_flow, NULL, NULL, NULL, 0
};
return &handler;
}
/*
* Reads input file, applies vol & flanger effects, stores in output file.
* E.g. example1 monkey.au monkey.aiff
*/
int main(int argc, char * argv[])
{
sox_effects_chain_t * chain;
sox_effect_t * e;
char * vol[] = {"3dB"};
assert(argc == 3);
/* All libSoX applications must start by initialising the SoX library */
assert(sox_init() == SOX_SUCCESS);
/* Open the input file (with default parameters) */
assert((in = sox_open_read(argv[1], NULL, NULL, NULL)));
/* Open the output file; we must specify the output signal characteristics.
* Since we are using only simple effects, they are the same as the input
* file characteristics */
assert((out = sox_open_write(argv[2], &in->signal, NULL, NULL, NULL, NULL)));
/* Create an effects chain; some effects need to know about the input
* or output file encoding so we provide that information here */
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
/* The first effect in the effect chain must be something that can source
* samples; in this case, we have defined an input handler that inputs
* data from an audio file */
e = sox_create_effect(input_handler());
/* This becomes the first `effect' in the chain */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Create the `vol' effect, and initialise it with the desired parameters: */
e = sox_create_effect(sox_find_effect("vol"));
assert(sox_effect_options(e, 1, vol) == SOX_SUCCESS);
/* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Create the `flanger' effect, and initialise it with default parameters: */
e = sox_create_effect(sox_find_effect("flanger"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
/* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* The last effect in the effect chain must be something that only consumes
* samples; in this case, we have defined an output handler that outputs
* data to an audio file */
e = sox_create_effect(output_handler());
assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);
free(e);
/* Flow samples through the effects processing chain until EOF is reached */
sox_flow_effects(chain, NULL, NULL);
/* All done; tidy up: */
sox_delete_effects_chain(chain);
sox_close(out);
sox_close(in);
sox_quit();
return 0;
}

118
app/src/main/cpp/sox/example2.c
Unescape View File

@ -0,0 +1,118 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2008 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include "util.h"
#include <stdio.h>
#include <math.h>
#include <assert.h>
/*
* Reads input file and displays a few seconds of wave-form, starting from
* a given time through the audio. E.g. example2 song2.au 30.75 1
*/
int main(int argc, char * argv[])
{
sox_format_t * in;
sox_sample_t * buf;
size_t blocks, block_size;
/* Period of audio over which we will measure its volume in order to
* display the wave-form: */
static const double block_period = 0.025; /* seconds */
double start_secs = 0, period = 2;
char dummy;
uint64_t seek;
/* All libSoX applications must start by initialising the SoX library */
assert(sox_init() == SOX_SUCCESS);
assert(argc > 1);
++argv, --argc; /* Move to 1st parameter */
/* Open the input file (with default parameters) */
assert((in = sox_open_read(*argv, NULL, NULL, NULL)));
++argv, --argc; /* Move past this parameter */
if (argc) { /* If given, read the start time: */
assert(sscanf(*argv, "%lf%c", &start_secs, &dummy) == 1);
++argv, --argc; /* Move past this parameter */
}
if (argc) { /* If given, read the period of time to display: */
assert(sscanf(*argv, "%lf%c", &period, &dummy) == 1);
++argv, --argc; /* Move past this parameter */
}
/* Calculate the start position in number of samples: */
seek = start_secs * in->signal.rate * in->signal.channels + .5;
/* Make sure that this is at a `wide sample' boundary: */
seek -= seek % in->signal.channels;
/* Move the file pointer to the desired starting position */
assert(sox_seek(in, seek, SOX_SEEK_SET) == SOX_SUCCESS);
/* Convert block size (in seconds) to a number of samples: */
block_size = block_period * in->signal.rate * in->signal.channels + .5;
/* Make sure that this is at a `wide sample' boundary: */
block_size -= block_size % in->signal.channels;
/* Allocate a block of memory to store the block of audio samples: */
assert((buf = malloc(sizeof(sox_sample_t) * block_size)));
/* This example program requires that the audio has precisely 2 channels: */
assert(in->signal.channels == 2);
/* Read and process blocks of audio for the selected period or until EOF: */
for (blocks = 0; sox_read(in, buf, block_size) == block_size && blocks * block_period < period; ++blocks) {
double left = 0, right = 0;
size_t i;
static const char line[] = "===================================";
int l, r;
for (i = 0; i < block_size; ++i) {
SOX_SAMPLE_LOCALS;
/* convert the sample from SoX's internal format to a `double' for
* processing in this application: */
double sample = SOX_SAMPLE_TO_FLOAT_64BIT(buf[i],);
/* The samples for each channel are interleaved; in this example
* we allow only stereo audio, so the left channel audio can be found in
* even-numbered samples, and the right channel audio in odd-numbered
* samples: */
if (i & 1)
right = max(right, fabs(sample)); /* Find the peak volume in the block */
else
left = max(left, fabs(sample)); /* Find the peak volume in the block */
}
/* Build up the wave form by displaying the left & right channel
* volume as a line length: */
l = (1 - left) * 35 + .5;
r = (1 - right) * 35 + .5;
printf("%8.3f%36s|%s\n", start_secs + blocks * block_period, line + l, line + r);
}
/* All done; tidy up: */
free(buf);
sox_close(in);
sox_quit();
return 0;
}

113
app/src/main/cpp/sox/example3.c
Unescape View File

@ -0,0 +1,113 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2007-9 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include "util.h"
#include <stdio.h>
#include <assert.h>
/*
* Example of a custom output message handler.
*/
static void output_message(unsigned level, const char *filename, const char *fmt, va_list ap)
{
char const * const str[] = {"FAIL", "WARN", "INFO", "DBUG"};
if (sox_globals.verbosity >= level) {
char base_name[128];
sox_basename(base_name, sizeof(base_name), filename);
fprintf(stderr, "%s %s: ", str[min(level - 1, 3)], base_name);
vfprintf(stderr, fmt, ap);
fprintf(stderr, "\n");
}
}
/*
* On an alsa capable system, plays an audio file starting 10 seconds in.
* Copes with sample-rate and channel change if necessary since its
* common for audio drivers to support a subset of rates and channel
* counts.
* E.g. example3 song2.ogg
*
* Can easily be changed to work with other audio device drivers supported
* by libSoX; e.g. "oss", "ao", "coreaudio", etc.
* See the soxformat(7) manual page.
*/
int main(int argc, char * argv[])
{
static sox_format_t * in, * out; /* input and output files */
sox_effects_chain_t * chain;
sox_effect_t * e;
sox_signalinfo_t interm_signal;
char * args[10];
assert(argc == 2);
sox_globals.output_message_handler = output_message;
sox_globals.verbosity = 1;
assert(sox_init() == SOX_SUCCESS);
assert((in = sox_open_read(argv[1], NULL, NULL, NULL)));
/* Change "alsa" in this line to use an alternative audio device driver: */
assert((out= sox_open_write("default", &in->signal, NULL, "alsa", NULL, NULL)));
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
interm_signal = in->signal; /* NB: deep copy */
e = sox_create_effect(sox_find_effect("input"));
args[0] = (char *)in, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &in->signal) == SOX_SUCCESS);
free(e);
e = sox_create_effect(sox_find_effect("trim"));
args[0] = "10", assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &in->signal) == SOX_SUCCESS);
free(e);
if (in->signal.rate != out->signal.rate) {
e = sox_create_effect(sox_find_effect("rate"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
}
if (in->signal.channels != out->signal.channels) {
e = sox_create_effect(sox_find_effect("channels"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
}
e = sox_create_effect(sox_find_effect("output"));
args[0] = (char *)out, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
sox_flow_effects(chain, NULL, NULL);
sox_delete_effects_chain(chain);
sox_close(out);
sox_close(in);
sox_quit();
return 0;
}

98
app/src/main/cpp/sox/example4.c
Unescape View File

@ -0,0 +1,98 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2009 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "sox.h"
#include <stdio.h>
/* Concatenate audio files. Note that the files must have the same number
* of channels and the same sample rate.
*
* Usage: example4 input-1 input-2 [... input-n] output
*/
#define check(x) do if (!(x)) { \
fprintf(stderr, "check failed: %s\n", #x); goto error; } while (0)
int main(int argc, char * argv[])
{
sox_format_t * output = NULL;
int i;
check(argc >= 1 + 2 + 1); /* Need at least 2 input files + 1 output file. */
check(sox_init() == SOX_SUCCESS);
/* Defer openning the output file as we want to set its characteristics
* based on those of the input files. */
for (i = 1; i < argc - 1; ++i) { /* For each input file... */
sox_format_t * input;
static sox_signalinfo_t signal; /* static quashes `uninitialised' warning.*/
/* The (maximum) number of samples that we shall read/write at a time;
* chosen as a rough match to typical operating system I/O buffer size: */
#define MAX_SAMPLES (size_t)2048
sox_sample_t samples[MAX_SAMPLES]; /* Temporary store whilst copying. */
size_t number_read;
/* Open this input file: */
check(input = sox_open_read(argv[i], NULL, NULL, NULL));
if (i == 1) { /* If this is the first input file... */
/* Open the output file using the same signal and encoding character-
* istics as the first input file. Note that here, input->signal.length
* will not be equal to the output file length so we are relying on
* libSoX to set the output length correctly (i.e. non-seekable output
* is not catered for); an alternative would be to first calculate the
* output length by summing the lengths of the input files and modifying
* the second parameter to sox_open_write accordingly. */
check(output = sox_open_write(argv[argc - 1],
&input->signal, &input->encoding, NULL, NULL, NULL));
/* Also, we'll store the signal characteristics of the first file
* so that we can check that these match those of the other inputs: */
signal = input->signal;
}
else { /* Second or subsequent input file... */
/* Check that this input file's signal matches that of the first file: */
check(input->signal.channels == signal.channels);
check(input->signal.rate == signal.rate);
}
/* Copy all of the audio from this input file to the output file: */
while ((number_read = sox_read(input, samples, MAX_SAMPLES)))
check(sox_write(output, samples, number_read) == number_read);
check(sox_close(input) == SOX_SUCCESS); /* Finished with this input file.*/
}
check(sox_close(output) == SOX_SUCCESS); /* Finished with the output file. */
output = NULL;
check(sox_quit() == SOX_SUCCESS);
return 0;
error: /* Truncate output file on error: */
if (output) {
FILE * f;
sox_close(output);
if ((f = fopen(argv[argc - 1], "w"))) fclose(f);
}
return 1;
}

83
app/src/main/cpp/sox/example5.c
Unescape View File

@ -0,0 +1,83 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2009 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include "util.h"
#include <stdio.h>
#include <assert.h>
/* Example of reading and writing audio files stored in memory buffers
* rather than actual files.
*
* Usage: example5 input output
*/
/* Uncomment following line for fixed instead of malloc'd buffer: */
/*#define FIXED_BUFFER */
#if defined FIXED_BUFFER
#define buffer_size 123456
static char buffer[buffer_size];
#endif
int main(int argc, char * argv[])
{
static sox_format_t * in, * out; /* input and output files */
#define MAX_SAMPLES (size_t)2048
sox_sample_t samples[MAX_SAMPLES]; /* Temporary store whilst copying. */
#if !defined FIXED_BUFFER
char * buffer;
size_t buffer_size;
#endif
size_t number_read;
assert(argc == 3);
/* All libSoX applications must start by initialising the SoX library */
assert(sox_init() == SOX_SUCCESS);
/* Open the input file (with default parameters) */
assert((in = sox_open_read(argv[1], NULL, NULL, NULL)));
#if defined FIXED_BUFFER
assert((out = sox_open_mem_write(buffer, buffer_size, &in->signal, NULL, "sox", NULL)));
#else
assert((out = sox_open_memstream_write(&buffer, &buffer_size, &in->signal, NULL, "sox", NULL)));
#endif
while ((number_read = sox_read(in, samples, MAX_SAMPLES)))
assert(sox_write(out, samples, number_read) == number_read);
sox_close(out);
sox_close(in);
assert((in = sox_open_mem_read(buffer, buffer_size, NULL, NULL, NULL)));
assert((out = sox_open_write(argv[2], &in->signal, NULL, NULL, NULL, NULL)));
while ((number_read = sox_read(in, samples, MAX_SAMPLES)))
assert(sox_write(out, samples, number_read) == number_read);
sox_close(out);
sox_close(in);
#if !defined FIXED_BUFFER
free(buffer);
#endif
sox_quit();
return 0;
}

128
app/src/main/cpp/sox/example6.c
Unescape View File

@ -0,0 +1,128 @@
/* Simple example of using SoX libraries
*
* Copyright (c) 2007-14 robs@users.sourceforge.net
*
* This program 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.
*
* This program 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifdef NDEBUG /* N.B. assert used with active statements so enable always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox.h"
#include "util.h"
#include <stdio.h>
#include <assert.h>
/*
* Shows how to explicitly specify output file signal and encoding attributes.
*
* The example converts a given input file of any type to mono mu-law at 8kHz
* sampling-rate (providing that this is supported by the given output file
* type).
*
* For example:
*
* sox -r 16k -n input.wav synth 8 sin 0:8k sin 8k:0 gain -1
* ./example6 input.wav output.wav
* soxi input.wav output.wav
*
* gives:
*
* Input File : 'input.wav'
* Channels : 2
* Sample Rate : 16000
* Precision : 32-bit
* Duration : 00:00:08.00 = 128000 samples ~ 600 CDDA sectors
* File Size : 1.02M
* Bit Rate : 1.02M
* Sample Encoding: 32-bit Signed Integer PCM
*
* Input File : 'output.wav'
* Channels : 1
* Sample Rate : 8000
* Precision : 14-bit
* Duration : 00:00:08.00 = 64000 samples ~ 600 CDDA sectors
* File Size : 64.1k
* Bit Rate : 64.1k
* Sample Encoding: 8-bit u-law
*/
int main(int argc, char * argv[])
{
static sox_format_t * in, * out; /* input and output files */
sox_effects_chain_t * chain;
sox_effect_t * e;
char * args[10];
sox_signalinfo_t interm_signal; /* @ intermediate points in the chain. */
sox_encodinginfo_t out_encoding = {
SOX_ENCODING_ULAW,
8,
0,
sox_option_default,
sox_option_default,
sox_option_default,
sox_false
};
sox_signalinfo_t out_signal = {
8000,
1,
0,
0,
NULL
};
assert(argc == 3);
assert(sox_init() == SOX_SUCCESS);
assert((in = sox_open_read(argv[1], NULL, NULL, NULL)));
assert((out = sox_open_write(argv[2], &out_signal, &out_encoding, NULL, NULL, NULL)));
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
interm_signal = in->signal; /* NB: deep copy */
e = sox_create_effect(sox_find_effect("input"));
args[0] = (char *)in, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &in->signal) == SOX_SUCCESS);
free(e);
if (in->signal.rate != out->signal.rate) {
e = sox_create_effect(sox_find_effect("rate"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
}
if (in->signal.channels != out->signal.channels) {
e = sox_create_effect(sox_find_effect("channels"));
assert(sox_effect_options(e, 0, NULL) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
}
e = sox_create_effect(sox_find_effect("output"));
args[0] = (char *)out, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
assert(sox_add_effect(chain, e, &interm_signal, &out->signal) == SOX_SUCCESS);
free(e);
sox_flow_effects(chain, NULL, NULL);
sox_delete_effects_chain(chain);
sox_close(out);
sox_close(in);
sox_quit();
return 0;
}

21
app/src/main/cpp/sox/f4-fmt.c
Unescape View File

@ -0,0 +1,21 @@
/* libSoX file formats: raw (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "raw.h"
RAW_FORMAT1(f4, "f32", 32, 0, FLOAT)

21
app/src/main/cpp/sox/f8-fmt.c
Unescape View File

@ -0,0 +1,21 @@
/* libSoX file formats: raw (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "raw.h"
RAW_FORMAT1(f8, "f64", 64, 0, FLOAT)

399
app/src/main/cpp/sox/fade.c
Unescape View File

@ -0,0 +1,399 @@
/* Ari Moisio <armoi@sci.fi> Aug 29 2000, based on skeleton effect
* Written by Chris Bagwell (cbagwell@sprynet.com) - March 16, 1999
*
* Copyright 1999 Chris Bagwell And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Chris Bagwell And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
#include "sox_i.h"
/* Fade curves */
#define FADE_QUARTER 'q' /* Quarter of sine wave, 0 to pi/2 */
#define FADE_HALF 'h' /* Half of sine wave, pi/2 to 1.5 * pi
* scaled so that -1 means no output
* and 1 means 0 db attenuation. */
#define FADE_LOG 'l' /* Logarithmic curve. Fades -100 db
* in given time. */
#define FADE_TRI 't' /* Linear slope. */
#define FADE_PAR 'p' /* Inverted parabola. */
#include <string.h>
/* Private data for fade file */
typedef struct { /* These are measured as samples */
uint64_t in_start, in_stop, out_start, out_stop, samplesdone;
char *in_stop_str, *out_start_str, *out_stop_str;
char in_fadetype, out_fadetype;
char do_out;
int endpadwarned;
} priv_t;
/* prototypes */
static double fade_gain(uint64_t index, uint64_t range, int fadetype);
/*
* Process options
*
* Don't do initialization now.
* The 'info' fields are not yet filled in.
*/
static int sox_fade_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * fade = (priv_t *) effp->priv;
char t_char[2];
int t_argno;
uint64_t samples;
const char *n;
--argc, ++argv;
if (argc < 1 || argc > 4)
return lsx_usage(effp);
/* because sample rate is unavailable at this point we store the
* string off for later computations.
*/
if (sscanf(argv[0], "%1[qhltp]", t_char))
{
fade->in_fadetype = *t_char;
fade->out_fadetype = *t_char;
argv++;
argc--;
}
else
{
/* No type given. */
fade->in_fadetype = 'l';
fade->out_fadetype = 'l';
}
fade->in_stop_str = lsx_strdup(argv[0]);
/* Do a dummy parse to see if it will fail */
n = lsx_parsesamples(0., fade->in_stop_str, &samples, 't');
if (!n || *n)
return lsx_usage(effp);
fade->in_stop = samples;
fade->out_start_str = fade->out_stop_str = 0;
for (t_argno = 1; t_argno < argc && t_argno < 3; t_argno++)
{
/* See if there is fade-in/fade-out times/curves specified. */
if(t_argno == 1)
{
fade->out_stop_str = lsx_strdup(argv[t_argno]);
/* Do a dummy parse to see if it will fail */
n = lsx_parseposition(0., fade->out_stop_str, NULL, (uint64_t)0, (uint64_t)0, '=');
if (!n || *n)
return lsx_usage(effp);
fade->out_stop = samples;
}
else
{
fade->out_start_str = lsx_strdup(argv[t_argno]);
/* Do a dummy parse to see if it will fail */
n = lsx_parsesamples(0., fade->out_start_str, &samples, 't');
if (!n || *n)
return lsx_usage(effp);
fade->out_start = samples;
}
} /* End for(t_argno) */
return(SOX_SUCCESS);
}
/*
* Prepare processing.
* Do all initializations.
*/
static int sox_fade_start(sox_effect_t * effp)
{
priv_t * fade = (priv_t *) effp->priv;
sox_bool truncate = sox_false;
uint64_t samples;
uint64_t in_length = effp->in_signal.length != SOX_UNKNOWN_LEN ?
effp->in_signal.length / effp->in_signal.channels : SOX_UNKNOWN_LEN;
/* converting time values to samples */
fade->in_start = 0;
if (lsx_parsesamples(effp->in_signal.rate, fade->in_stop_str,
&samples, 't') == NULL)
return lsx_usage(effp);
fade->in_stop = samples;
fade->do_out = 0;
/* See if user specified a stop time */
if (fade->out_stop_str)
{
fade->do_out = 1;
if (!lsx_parseposition(effp->in_signal.rate, fade->out_stop_str,
&samples, (uint64_t)0, in_length, '=') ||
samples == SOX_UNKNOWN_LEN) {
lsx_fail("audio length is unknown");
return SOX_EOF;
}
fade->out_stop = samples;
if (!(truncate = !!fade->out_stop)) {
fade->out_stop = effp->in_signal.length != SOX_UNKNOWN_LEN ?
effp->in_signal.length / effp->in_signal.channels :
0;
if (!fade->out_stop) {
lsx_fail("cannot fade out: audio length is neither known nor given");
return SOX_EOF;
}
}
/* See if user wants to fade out. */
if (fade->out_start_str)
{
if (lsx_parsesamples(effp->in_signal.rate, fade->out_start_str,
&samples, 't') == NULL)
return lsx_usage(effp);
/* Fade time is relative to stop time. */
fade->out_start = fade->out_stop - samples;
}
else
/* If user doesn't specify fade out length then
* use same length as input side. This is stored
* in in_stop.
*/
fade->out_start = fade->out_stop - fade->in_stop;
}
else
/* If not specified then user wants to process all
* of file. Use a value of zero to indicate this.
*/
fade->out_stop = 0;
if (fade->out_start) { /* Sanity check */
if (fade->in_stop > fade->out_start)
--fade->in_stop; /* 1 sample grace for rounding error. */
if (fade->in_stop > fade->out_start) {
lsx_fail("fade-out overlaps fade-in");
return SOX_EOF;
}
}
fade->samplesdone = fade->in_start;
fade->endpadwarned = 0;
lsx_debug("in_start = %" PRIu64 " in_stop = %" PRIu64 " "
"out_start = %" PRIu64 " out_stop = %" PRIu64,
fade->in_start, fade->in_stop, fade->out_start, fade->out_stop);
if (fade->in_start == fade->in_stop && !truncate &&
fade->out_start == fade->out_stop)
return SOX_EFF_NULL;
effp->out_signal.length = truncate ?
fade->out_stop * effp->in_signal.channels : effp->in_signal.length;
return SOX_SUCCESS;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int sox_fade_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * fade = (priv_t *) effp->priv;
/* len is total samples, chcnt counts channels */
int len = 0, t_output = 1, more_output = 1;
sox_sample_t t_ibuf;
size_t chcnt = 0;
len = ((*isamp > *osamp) ? *osamp : *isamp);
*osamp = 0;
*isamp = 0;
for(; len && more_output; len--)
{
t_ibuf = *ibuf;
if ((fade->samplesdone >= fade->in_start) &&
(!fade->do_out || fade->samplesdone < fade->out_stop))
{ /* something to generate output */
if (fade->samplesdone < fade->in_stop)
{ /* fade-in phase, increase gain */
*obuf = t_ibuf *
fade_gain(fade->samplesdone - fade->in_start,
fade->in_stop - fade->in_start,
fade->in_fadetype);
} /* endif fade-in */
else if (!fade->do_out || fade->samplesdone < fade->out_start)
{ /* steady gain phase */
*obuf = t_ibuf;
} /* endif steady phase */
else
{ /* fade-out phase, decrease gain */
*obuf = t_ibuf *
fade_gain(fade->out_stop - fade->samplesdone,
fade->out_stop - fade->out_start,
fade->out_fadetype);
} /* endif fade-out */
if (!(!fade->do_out || fade->samplesdone < fade->out_stop))
more_output = 0;
t_output = 1;
}
else
{ /* No output generated */
t_output = 0;
} /* endif something to output */
*isamp += 1;
ibuf++;
if (t_output)
{ /* Output generated, update pointers and counters */
obuf++;
*osamp += 1;
} /* endif t_output */
/* Process next channel */
chcnt++;
if (chcnt >= effp->in_signal.channels)
{ /* all channels of this sample processed */
chcnt = 0;
fade->samplesdone += 1;
} /* endif all channels */
} /* endfor */
/* If not more samples will be returned, let application know
* this.
*/
if (fade->do_out && fade->samplesdone >= fade->out_stop)
return SOX_EOF;
else
return SOX_SUCCESS;
}
/*
* Drain out remaining samples if the effect generates any.
*/
static int sox_fade_drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * fade = (priv_t *) effp->priv;
int len;
size_t t_chan = 0;
len = *osamp;
len -= len % effp->in_signal.channels;
*osamp = 0;
if (fade->do_out && fade->samplesdone < fade->out_stop &&
!(fade->endpadwarned))
{ /* Warning about padding silence into end of sample */
lsx_warn("End time past end of audio. Padding with silence");
fade->endpadwarned = 1;
} /* endif endpadwarned */
for (;len && (fade->do_out &&
fade->samplesdone < fade->out_stop); len--)
{
*obuf = 0;
obuf++;
*osamp += 1;
t_chan++;
if (t_chan >= effp->in_signal.channels)
{
fade->samplesdone += 1;
t_chan = 0;
} /* endif channels */
} /* endfor */
if (fade->do_out && fade->samplesdone >= fade->out_stop)
return SOX_EOF;
else
return SOX_SUCCESS;
}
/*
* Do anything required when you stop reading samples.
* (free allocated memory, etc.)
*/
static int lsx_kill(sox_effect_t * effp)
{
priv_t * fade = (priv_t *) effp->priv;
free(fade->in_stop_str);
free(fade->out_start_str);
free(fade->out_stop_str);
return (SOX_SUCCESS);
}
/* Function returns gain value 0.0 - 1.0 according index / range ratio
* and -1.0 if type is invalid
* todo: to optimize performance calculate gain every now and then and interpolate */
static double fade_gain(uint64_t index, uint64_t range, int type)
{
double retval = 0.0, findex = 0.0;
/* TODO: does it really have to be contrained to [0.0, 1.0]? */
findex = max(0.0, min(1.0, 1.0 * index / range));
switch (type) {
case FADE_TRI : /* triangle */
retval = findex;
break;
case FADE_QUARTER : /* quarter of sinewave */
retval = sin(findex * M_PI / 2);
break;
case FADE_HALF : /* half of sinewave... eh cosine wave */
retval = (1 - cos(findex * M_PI )) / 2 ;
break;
case FADE_LOG : /* logarithmic */
/* 5 means 100 db attenuation. */
/* TODO: should this be adopted with bit depth */
retval = pow(0.1, (1 - findex) * 5);
break;
case FADE_PAR : /* inverted parabola */
retval = (1 - (1 - findex) * (1 - findex));
break;
/* TODO: more fade curves? */
default : /* Error indicating wrong fade curve */
retval = -1.0;
break;
}
return retval;
}
static sox_effect_handler_t sox_fade_effect = {
"fade",
"[ type ] fade-in-length [ stop-position [ fade-out-length ] ]\n"
" Time is in hh:mm:ss.frac format.\n"
" Fade type one of q, h, t, l or p.",
SOX_EFF_MCHAN | SOX_EFF_LENGTH,
sox_fade_getopts,
sox_fade_start,
sox_fade_flow,
sox_fade_drain,
NULL,
lsx_kill, sizeof(priv_t)
};
const sox_effect_handler_t *lsx_fade_effect_fn(void)
{
return &sox_fade_effect;
}

31
app/src/main/cpp/sox/fap.c
Unescape View File

@ -0,0 +1,31 @@
/* libSoX file format: FAP Copyright (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
LSX_FORMAT_HANDLER(fap)
{
static char const * const names[] = {"fap", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_SIGN2, 24, 16, 8,0,0};
static sox_format_handler_t handler;
handler = *lsx_sndfile_format_fn();
handler.description =
"Ensoniq PARIS digital audio editing system (little endian)";
handler.names = names;
handler.write_formats = write_encodings;
return &handler;
}

1362
app/src/main/cpp/sox/fft4g.c
View File

File diff suppressed because it is too large Load Diff

37
app/src/main/cpp/sox/fft4g.h
Unescape View File

@ -0,0 +1,37 @@
/* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define FFT4G_MAX_SIZE 262144
void lsx_cdft(int, int, double *, int *, double *);
void lsx_rdft(int, int, double *, int *, double *);
void lsx_ddct(int, int, double *, int *, double *);
void lsx_ddst(int, int, double *, int *, double *);
void lsx_dfct(int, double *, double *, int *, double *);
void lsx_dfst(int, double *, double *, int *, double *);
void lsx_cdft_f(int, int, float *, int *, float *);
void lsx_rdft_f(int, int, float *, int *, float *);
void lsx_ddct_f(int, int, float *, int *, float *);
void lsx_ddst_f(int, int, float *, int *, float *);
void lsx_dfct_f(int, float *, float *, int *, float *);
void lsx_dfst_f(int, float *, float *, int *, float *);
#define dft_br_len(l) (2 + (1 << (int)(log(l / 2 + .5) / log(2.)) / 2))
#define dft_sc_len(l) (l / 2)
/* Over-allocate h by 2 to use these macros */
#define LSX_PACK(h, n) h[1] = h[n]
#define LSX_UNPACK(h, n) h[n] = h[1], h[n + 1] = h[1] = 0;

119
app/src/main/cpp/sox/fifo.h
Unescape View File

@ -0,0 +1,119 @@
/* Addressable FIFO buffer Copyright (c) 2007 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef fifo_included
#define fifo_included
#include <string.h>
#ifndef FIFO_SIZE_T
#define FIFO_SIZE_T size_t
#endif
typedef struct {
char * data;
size_t allocation; /* Number of bytes allocated for data. */
size_t item_size; /* Size of each item in data */
size_t begin; /* Offset of the first byte to read. */
size_t end; /* 1 + Offset of the last byte byte to read. */
} fifo_t;
#define FIFO_MIN 0x4000
UNUSED static void fifo_clear(fifo_t * f)
{
f->end = f->begin = 0;
}
UNUSED static void * fifo_reserve(fifo_t * f, FIFO_SIZE_T n)
{
n *= f->item_size;
if (f->begin == f->end)
fifo_clear(f);
while (1) {
if (f->end + n <= f->allocation) {
void *p = f->data + f->end;
f->end += n;
return p;
}
if (f->begin > FIFO_MIN) {
memmove(f->data, f->data + f->begin, f->end - f->begin);
f->end -= f->begin;
f->begin = 0;
continue;
}
f->allocation += n;
f->data = lsx_realloc(f->data, f->allocation);
}
}
UNUSED static void * fifo_write(fifo_t * f, FIFO_SIZE_T n, void const * data)
{
void * s = fifo_reserve(f, n);
if (data)
memcpy(s, data, n * f->item_size);
return s;
}
UNUSED static void fifo_trim_to(fifo_t * f, FIFO_SIZE_T n)
{
n *= f->item_size;
f->end = f->begin + n;
}
UNUSED static void fifo_trim_by(fifo_t * f, FIFO_SIZE_T n)
{
n *= f->item_size;
f->end -= n;
}
UNUSED static FIFO_SIZE_T fifo_occupancy(fifo_t * f)
{
return (f->end - f->begin) / f->item_size;
}
UNUSED static void * fifo_read(fifo_t * f, FIFO_SIZE_T n, void * data)
{
char * ret = f->data + f->begin;
n *= f->item_size;
if (n > (FIFO_SIZE_T)(f->end - f->begin))
return NULL;
if (data)
memcpy(data, ret, (size_t)n);
f->begin += n;
return ret;
}
#define fifo_read_ptr(f) fifo_read(f, (FIFO_SIZE_T)0, NULL)
UNUSED static void fifo_delete(fifo_t * f)
{
free(f->data);
}
UNUSED static void fifo_create(fifo_t * f, FIFO_SIZE_T item_size)
{
f->item_size = item_size;
f->allocation = FIFO_MIN;
f->data = lsx_malloc(f->allocation);
fifo_clear(f);
}
#endif

105
app/src/main/cpp/sox/fir.c
Unescape View File

@ -0,0 +1,105 @@
/* Effect: fir filter from coefs Copyright (c) 2009 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "dft_filter.h"
typedef struct {
dft_filter_priv_t base;
char const * filename;
double * h;
int n;
} priv_t;
static int create(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
dft_filter_priv_t * b = &p->base;
double d;
char c;
b->filter_ptr = &b->filter;
--argc, ++argv;
if (!argc)
p->filename = "-"; /* default to stdin */
else if (argc == 1)
p->filename = argv[0], --argc;
else for (; argc && sscanf(*argv, "%lf%c", &d, &c) == 1; --argc, ++argv) {
p->n++;
p->h = lsx_realloc(p->h, p->n * sizeof(*p->h));
p->h[p->n - 1] = d;
}
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
dft_filter_t * f = p->base.filter_ptr;
double d;
char c;
int i;
if (!f->num_taps) {
if (!p->n && p->filename) {
FILE * file = lsx_open_input_file(effp, p->filename, sox_true);
if (!file)
return SOX_EOF;
while ((i = fscanf(file, " #%*[^\n]%c", &c)) >= 0) {
if (i >= 1) continue; /* found and skipped a comment */
if ((i = fscanf(file, "%lf", &d)) > 0) {
/* found a coefficient value */
p->n++;
p->h = lsx_realloc(p->h, p->n * sizeof(*p->h));
p->h[p->n - 1] = d;
} else break; /* either EOF, or something went wrong
(read or syntax error) */
}
if (!feof(file)) {
lsx_fail("error reading coefficient file");
if (file != stdin) fclose(file);
return SOX_EOF;
}
if (file != stdin) fclose(file);
}
lsx_report("%i coefficients", p->n);
if (!p->n)
return SOX_EFF_NULL;
if (effp->global_info->plot != sox_plot_off) {
char title[100];
sprintf(title, "SoX effect: fir (%d coefficients)", p->n);
lsx_plot_fir(p->h, p->n, effp->in_signal.rate,
effp->global_info->plot, title, -30., 30.);
free(p->h);
return SOX_EOF;
}
lsx_set_dft_filter(f, p->h, p->n, p->n >> 1);
}
return lsx_dft_filter_effect_fn()->start(effp);
}
sox_effect_handler_t const * lsx_fir_effect_fn(void)
{
static sox_effect_handler_t handler;
handler = *lsx_dft_filter_effect_fn();
handler.name = "fir";
handler.usage = "[coef-file|coefs]";
handler.getopts = create;
handler.start = start;
handler.priv_size = sizeof(priv_t);
return &handler;
}

145
app/src/main/cpp/sox/firfit.c
Unescape View File

@ -0,0 +1,145 @@
/* Effect: firfit filter Copyright (c) 2009 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* This is W.I.P. hence marked SOX_EFF_ALPHA for now.
* Need to add other interpolation types e.g. linear, bspline, window types,
* and filter length, maybe phase response type too.
*/
#include "sox_i.h"
#include "dft_filter.h"
typedef struct {
dft_filter_priv_t base;
char const * filename;
struct {double f, gain;} * knots;
int num_knots, n;
} priv_t;
static int create(sox_effect_t * effp, int argc, char **argv)
{
priv_t * p = (priv_t *)effp->priv;
dft_filter_priv_t * b = &p->base;
b->filter_ptr = &b->filter;
--argc, ++argv;
if (argc == 1)
p->filename = argv[0], --argc;
p->n = 2047;
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static double * make_filter(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
double * log_freqs, * gains, * d, * work, * h;
sox_rate_t rate = effp->in_signal.rate;
int i, work_len;
lsx_valloc(log_freqs , p->num_knots);
lsx_valloc(gains, p->num_knots);
lsx_valloc(d , p->num_knots);
for (i = 0; i < p->num_knots; ++i) {
log_freqs[i] = log(max(p->knots[i].f, 1));
gains[i] = p->knots[i].gain;
}
lsx_prepare_spline3(log_freqs, gains, p->num_knots, HUGE_VAL, HUGE_VAL, d);
for (work_len = 8192; work_len < rate / 2; work_len <<= 1);
work = lsx_calloc(work_len + 2, sizeof(*work));
lsx_valloc(h, p->n);
for (i = 0; i <= work_len; i += 2) {
double f = rate * 0.5 * i / work_len;
double spl1 = f < max(p->knots[0].f, 1)? gains[0] :
f > p->knots[p->num_knots - 1].f? gains[p->num_knots - 1] :
lsx_spline3(log_freqs, gains, d, p->num_knots, log(f));
work[i] = dB_to_linear(spl1);
}
LSX_PACK(work, work_len);
lsx_safe_rdft(work_len, -1, work);
for (i = 0; i < p->n; ++i)
h[i] = work[(work_len - p->n / 2 + i) % work_len] * 2. / work_len;
lsx_apply_blackman_nutall(h, p->n);
free(work);
return h;
}
static sox_bool read_knots(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
FILE * file = lsx_open_input_file(effp, p->filename, sox_true);
sox_bool result = sox_false;
int num_converted = 1;
char c;
if (file) {
lsx_valloc(p->knots, 1);
while (fscanf(file, " #%*[^\n]%c", &c) >= 0) {
num_converted = fscanf(file, "%lf %lf",
&p->knots[p->num_knots].f, &p->knots[p->num_knots].gain);
if (num_converted == 2) {
if (p->num_knots && p->knots[p->num_knots].f <= p->knots[p->num_knots - 1].f) {
lsx_fail("knot frequencies must be strictly increasing");
break;
}
lsx_revalloc(p->knots, ++p->num_knots + 1);
} else if (num_converted != 0)
break;
}
lsx_report("%i knots", p->num_knots);
if (feof(file) && num_converted != 1)
result = sox_true;
else lsx_fail("error reading knot file `%s', line number %u", p->filename, 1 + p->num_knots);
if (file != stdin)
fclose(file);
}
return result;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
dft_filter_t * f = p->base.filter_ptr;
if (!f->num_taps) {
double * h;
if (!p->num_knots && !read_knots(effp))
return SOX_EOF;
h = make_filter(effp);
if (effp->global_info->plot != sox_plot_off) {
lsx_plot_fir(h, p->n, effp->in_signal.rate,
effp->global_info->plot, "SoX effect: firfit", -30., +30.);
return SOX_EOF;
}
lsx_set_dft_filter(f, h, p->n, p->n >> 1);
}
return lsx_dft_filter_effect_fn()->start(effp);
}
sox_effect_handler_t const * lsx_firfit_effect_fn(void)
{
static sox_effect_handler_t handler;
handler = *lsx_dft_filter_effect_fn();
handler.name = "firfit";
handler.usage = "[knots-file]";
handler.flags |= SOX_EFF_ALPHA;
handler.getopts = create;
handler.start = start;
handler.priv_size = sizeof(priv_t);
return &handler;
}

606
app/src/main/cpp/sox/flac.c
Unescape View File

@ -0,0 +1,606 @@
/* libSoX file format: FLAC (c) 2006-7 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <string.h>
/* Next line for systems that don't define off_t when you #include
stdio.h; apparently OS/2 has this bug */
#include <sys/types.h>
#include <FLAC/all.h>
#define MAX_COMPRESSION 8
typedef struct {
/* Info: */
unsigned bits_per_sample;
unsigned channels;
unsigned sample_rate;
uint64_t total_samples;
/* Decode buffer: */
sox_sample_t *req_buffer; /* this may be on the stack */
size_t number_of_requested_samples;
sox_sample_t *leftover_buf; /* heap */
unsigned number_of_leftover_samples;
FLAC__StreamDecoder * decoder;
FLAC__bool eof;
sox_bool seek_pending;
uint64_t seek_offset;
/* Encode buffer: */
FLAC__int32 * decoded_samples;
unsigned number_of_samples;
FLAC__StreamEncoder * encoder;
FLAC__StreamMetadata * metadata[2];
unsigned num_metadata;
} priv_t;
static FLAC__StreamDecoderReadStatus decoder_read_callback(FLAC__StreamDecoder const* decoder UNUSED, FLAC__byte buffer[], size_t* bytes, void* ft_data)
{
sox_format_t* ft = (sox_format_t*)ft_data;
if(*bytes > 0) {
*bytes = lsx_readbuf(ft, buffer, *bytes);
if(lsx_error(ft))
return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
else if(*bytes == 0)
return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
else
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}
else
return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
}
static FLAC__StreamDecoderSeekStatus decoder_seek_callback(FLAC__StreamDecoder const* decoder UNUSED, FLAC__uint64 absolute_byte_offset, void* ft_data)
{
sox_format_t* ft = (sox_format_t*)ft_data;
if(lsx_seeki(ft, (off_t)absolute_byte_offset, SEEK_SET) < 0)
return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR;
else
return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
}
static FLAC__StreamDecoderTellStatus decoder_tell_callback(FLAC__StreamDecoder const* decoder UNUSED, FLAC__uint64* absolute_byte_offset, void* ft_data)
{
sox_format_t* ft = (sox_format_t*)ft_data;
off_t pos;
if((pos = lsx_tell(ft)) < 0)
return FLAC__STREAM_DECODER_TELL_STATUS_ERROR;
else {
*absolute_byte_offset = (FLAC__uint64)pos;
return FLAC__STREAM_DECODER_TELL_STATUS_OK;
}
}
static FLAC__StreamDecoderLengthStatus decoder_length_callback(FLAC__StreamDecoder const* decoder UNUSED, FLAC__uint64* stream_length, void* ft_data)
{
sox_format_t* ft = (sox_format_t*)ft_data;
*stream_length = lsx_filelength(ft);
return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
}
static FLAC__bool decoder_eof_callback(FLAC__StreamDecoder const* decoder UNUSED, void* ft_data)
{
sox_format_t* ft = (sox_format_t*)ft_data;
return lsx_eof(ft) ? 1 : 0;
}
static void decoder_metadata_callback(FLAC__StreamDecoder const * const flac, FLAC__StreamMetadata const * const metadata, void * const client_data)
{
sox_format_t * ft = (sox_format_t *) client_data;
priv_t * p = (priv_t *)ft->priv;
(void) flac;
if (metadata->type == FLAC__METADATA_TYPE_STREAMINFO) {
p->bits_per_sample = metadata->data.stream_info.bits_per_sample;
p->channels = metadata->data.stream_info.channels;
p->sample_rate = metadata->data.stream_info.sample_rate;
p->total_samples = metadata->data.stream_info.total_samples;
}
else if (metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
const FLAC__StreamMetadata_VorbisComment *vc = &metadata->data.vorbis_comment;
size_t i;
if (vc->num_comments == 0)
return;
if (ft->oob.comments != NULL) {
lsx_warn("multiple Vorbis comment block ignored");
return;
}
for (i = 0; i < vc->num_comments; ++i)
if (vc->comments[i].entry)
sox_append_comment(&ft->oob.comments, (char const *) vc->comments[i].entry);
}
}
static void decoder_error_callback(FLAC__StreamDecoder const * const flac, FLAC__StreamDecoderErrorStatus const status, void * const client_data)
{
sox_format_t * ft = (sox_format_t *) client_data;
(void) flac;
lsx_fail_errno(ft, SOX_EINVAL, "%s", FLAC__StreamDecoderErrorStatusString[status]);
}
static FLAC__StreamDecoderWriteStatus decoder_write_callback(FLAC__StreamDecoder const * const flac, FLAC__Frame const * const frame, FLAC__int32 const * const buffer[], void * const client_data)
{
sox_format_t * ft = (sox_format_t *) client_data;
priv_t * p = (priv_t *)ft->priv;
sox_sample_t * dst = p->req_buffer;
unsigned channel;
unsigned nsamples = frame->header.blocksize;
unsigned sample = 0;
size_t actual = nsamples * p->channels;
(void) flac;
if (frame->header.bits_per_sample != p->bits_per_sample || frame->header.channels != p->channels || frame->header.sample_rate != p->sample_rate) {
lsx_fail_errno(ft, SOX_EINVAL, "FLAC ERROR: parameters differ between frame and header");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
if (dst == NULL) {
lsx_warn("FLAC ERROR: entered write callback without a buffer (SoX bug)");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
/* FLAC may give us too much data, prepare the leftover buffer */
if (actual > p->number_of_requested_samples) {
size_t to_stash = actual - p->number_of_requested_samples;
p->leftover_buf = lsx_malloc(to_stash * sizeof(sox_sample_t));
p->number_of_leftover_samples = to_stash;
nsamples = p->number_of_requested_samples / p->channels;
p->req_buffer += p->number_of_requested_samples;
p->number_of_requested_samples = 0;
} else {
p->req_buffer += actual;
p->number_of_requested_samples -= actual;
}
leftover_copy:
for (; sample < nsamples; sample++) {
for (channel = 0; channel < p->channels; channel++) {
FLAC__int32 d = buffer[channel][sample];
switch (p->bits_per_sample) {
case 8: *dst++ = SOX_SIGNED_8BIT_TO_SAMPLE(d,); break;
case 16: *dst++ = SOX_SIGNED_16BIT_TO_SAMPLE(d,); break;
case 24: *dst++ = SOX_SIGNED_24BIT_TO_SAMPLE(d,); break;
case 32: *dst++ = SOX_SIGNED_32BIT_TO_SAMPLE(d,); break;
}
}
}
/* copy into the leftover buffer if we've prepared it */
if (sample < frame->header.blocksize) {
nsamples = frame->header.blocksize;
dst = p->leftover_buf;
goto leftover_copy;
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static int start_read(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
lsx_debug("API version %u", FLAC_API_VERSION_CURRENT);
p->decoder = FLAC__stream_decoder_new();
if (p->decoder == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the decoder instance");
return SOX_EOF;
}
FLAC__stream_decoder_set_md5_checking(p->decoder, sox_true);
FLAC__stream_decoder_set_metadata_respond_all(p->decoder);
if (FLAC__stream_decoder_init_stream(
p->decoder,
decoder_read_callback,
ft->seekable ? decoder_seek_callback : NULL,
ft->seekable ? decoder_tell_callback : NULL,
ft->seekable ? decoder_length_callback : NULL,
ft->seekable ? decoder_eof_callback : NULL,
decoder_write_callback,
decoder_metadata_callback,
decoder_error_callback,
ft) != FLAC__STREAM_DECODER_INIT_STATUS_OK){
lsx_fail_errno(ft, SOX_EHDR, "FLAC ERROR initialising decoder");
return SOX_EOF;
}
if (!FLAC__stream_decoder_process_until_end_of_metadata(p->decoder)) {
lsx_fail_errno(ft, SOX_EHDR, "FLAC ERROR whilst decoding metadata");
return SOX_EOF;
}
if (FLAC__stream_decoder_get_state(p->decoder) > FLAC__STREAM_DECODER_END_OF_STREAM) {
lsx_fail_errno(ft, SOX_EHDR, "FLAC ERROR during metadata decoding");
return SOX_EOF;
}
ft->encoding.encoding = SOX_ENCODING_FLAC;
ft->signal.rate = p->sample_rate;
ft->encoding.bits_per_sample = p->bits_per_sample;
ft->signal.channels = p->channels;
ft->signal.length = p->total_samples * p->channels;
return SOX_SUCCESS;
}
static size_t read_samples(sox_format_t * const ft, sox_sample_t * sampleBuffer, size_t const requested)
{
priv_t * p = (priv_t *)ft->priv;
size_t prev_requested;
if (p->seek_pending) {
p->seek_pending = sox_false;
/* discard leftover decoded data */
free(p->leftover_buf);
p->leftover_buf = NULL;
p->number_of_leftover_samples = 0;
p->req_buffer = sampleBuffer;
p->number_of_requested_samples = requested;
/* calls decoder_write_callback */
if (!FLAC__stream_decoder_seek_absolute(p->decoder, (FLAC__uint64)(p->seek_offset / ft->signal.channels))) {
p->req_buffer = NULL;
return 0;
}
} else if (p->number_of_leftover_samples > 0) {
/* small request, no need to decode more samples since we have leftovers */
if (requested < p->number_of_leftover_samples) {
size_t req_bytes = requested * sizeof(sox_sample_t);
memcpy(sampleBuffer, p->leftover_buf, req_bytes);
p->number_of_leftover_samples -= requested;
memmove(p->leftover_buf, (char *)p->leftover_buf + req_bytes,
(size_t)p->number_of_leftover_samples * sizeof(sox_sample_t));
return requested;
}
/* first, give them all of our leftover data: */
memcpy(sampleBuffer, p->leftover_buf,
p->number_of_leftover_samples * sizeof(sox_sample_t));
p->req_buffer = sampleBuffer + p->number_of_leftover_samples;
p->number_of_requested_samples = requested - p->number_of_leftover_samples;
free(p->leftover_buf);
p->leftover_buf = NULL;
p->number_of_leftover_samples = 0;
/* continue invoking decoder below */
} else {
p->req_buffer = sampleBuffer;
p->number_of_requested_samples = requested;
}
/* invoke the decoder, calls decoder_write_callback */
while ((prev_requested = p->number_of_requested_samples) && !p->eof) {
if (!FLAC__stream_decoder_process_single(p->decoder))
break; /* error, but maybe got earlier in the loop, though */
/* number_of_requested_samples decrements as the decoder progresses */
if (p->number_of_requested_samples == prev_requested)
p->eof = sox_true;
}
p->req_buffer = NULL;
return requested - p->number_of_requested_samples;
}
static int stop_read(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
if (!FLAC__stream_decoder_finish(p->decoder) && p->eof)
lsx_warn("decoder MD5 checksum mismatch.");
FLAC__stream_decoder_delete(p->decoder);
free(p->leftover_buf);
p->leftover_buf = NULL;
p->number_of_leftover_samples = 0;
return SOX_SUCCESS;
}
static FLAC__StreamEncoderWriteStatus flac_stream_encoder_write_callback(FLAC__StreamEncoder const * const flac, const FLAC__byte buffer[], size_t const bytes, unsigned const samples, unsigned const current_frame, void * const client_data)
{
sox_format_t * const ft = (sox_format_t *) client_data;
(void) flac, (void) samples, (void) current_frame;
return lsx_writebuf(ft, buffer, bytes) == bytes ? FLAC__STREAM_ENCODER_WRITE_STATUS_OK : FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
}
static void flac_stream_encoder_metadata_callback(FLAC__StreamEncoder const * encoder, FLAC__StreamMetadata const * metadata, void * client_data)
{
(void) encoder, (void) metadata, (void) client_data;
}
static FLAC__StreamEncoderSeekStatus flac_stream_encoder_seek_callback(FLAC__StreamEncoder const * encoder, FLAC__uint64 absolute_byte_offset, void * client_data)
{
sox_format_t * const ft = (sox_format_t *) client_data;
(void) encoder;
if (!ft->seekable)
return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
else if (lsx_seeki(ft, (off_t)absolute_byte_offset, SEEK_SET) != SOX_SUCCESS)
return FLAC__STREAM_ENCODER_SEEK_STATUS_ERROR;
else
return FLAC__STREAM_ENCODER_SEEK_STATUS_OK;
}
static FLAC__StreamEncoderTellStatus flac_stream_encoder_tell_callback(FLAC__StreamEncoder const * encoder, FLAC__uint64 * absolute_byte_offset, void * client_data)
{
sox_format_t * const ft = (sox_format_t *) client_data;
off_t pos;
(void) encoder;
if (!ft->seekable)
return FLAC__STREAM_ENCODER_TELL_STATUS_UNSUPPORTED;
else if ((pos = lsx_tell(ft)) < 0)
return FLAC__STREAM_ENCODER_TELL_STATUS_ERROR;
else {
*absolute_byte_offset = (FLAC__uint64)pos;
return FLAC__STREAM_ENCODER_TELL_STATUS_OK;
}
}
static int start_write(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
FLAC__StreamEncoderInitStatus status;
unsigned compression_level = MAX_COMPRESSION; /* Default to "best" */
if (ft->encoding.compression != HUGE_VAL) {
compression_level = ft->encoding.compression;
if (compression_level != ft->encoding.compression ||
compression_level > MAX_COMPRESSION) {
lsx_fail_errno(ft, SOX_EINVAL,
"FLAC compression level must be a whole number from 0 to %i",
MAX_COMPRESSION);
return SOX_EOF;
}
}
p->encoder = FLAC__stream_encoder_new();
if (p->encoder == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder instance");
return SOX_EOF;
}
p->bits_per_sample = ft->encoding.bits_per_sample;
ft->signal.precision = ft->encoding.bits_per_sample;
lsx_report("encoding at %i bits per sample", p->bits_per_sample);
FLAC__stream_encoder_set_channels(p->encoder, ft->signal.channels);
FLAC__stream_encoder_set_bits_per_sample(p->encoder, p->bits_per_sample);
FLAC__stream_encoder_set_sample_rate(p->encoder, (unsigned)(ft->signal.rate + .5));
{ /* Check if rate is streamable: */
static const unsigned streamable_rates[] =
{8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000};
size_t i;
sox_bool streamable = sox_false;
for (i = 0; !streamable && i < array_length(streamable_rates); ++i)
streamable = (streamable_rates[i] == ft->signal.rate);
if (!streamable) {
lsx_report("non-standard rate; output may not be streamable");
FLAC__stream_encoder_set_streamable_subset(p->encoder, sox_false);
}
}
#if FLAC_API_VERSION_CURRENT >= 10
FLAC__stream_encoder_set_compression_level(p->encoder, compression_level);
#else
{
static struct {
unsigned blocksize;
FLAC__bool do_exhaustive_model_search;
FLAC__bool do_mid_side_stereo;
FLAC__bool loose_mid_side_stereo;
unsigned max_lpc_order;
unsigned max_residual_partition_order;
unsigned min_residual_partition_order;
} const options[MAX_COMPRESSION + 1] = {
{1152, sox_false, sox_false, sox_false, 0, 2, 2},
{1152, sox_false, sox_true, sox_true, 0, 2, 2},
{1152, sox_false, sox_true, sox_false, 0, 3, 0},
{4608, sox_false, sox_false, sox_false, 6, 3, 3},
{4608, sox_false, sox_true, sox_true, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 4, 0},
{4608, sox_true, sox_true, sox_false, 8, 6, 0},
{4608, sox_true, sox_true, sox_false, 12, 6, 0},
};
#define SET_OPTION(x) do {\
lsx_report(#x" = %i", options[compression_level].x); \
FLAC__stream_encoder_set_##x(p->encoder, options[compression_level].x);\
} while (0)
SET_OPTION(blocksize);
SET_OPTION(do_exhaustive_model_search);
SET_OPTION(max_lpc_order);
SET_OPTION(max_residual_partition_order);
SET_OPTION(min_residual_partition_order);
if (ft->signal.channels == 2) {
SET_OPTION(do_mid_side_stereo);
SET_OPTION(loose_mid_side_stereo);
}
#undef SET_OPTION
}
#endif
if (ft->signal.length != 0) {
FLAC__stream_encoder_set_total_samples_estimate(p->encoder, (FLAC__uint64)(ft->signal.length / ft->signal.channels));
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_SEEKTABLE);
if (p->metadata[p->num_metadata] == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table template");
return SOX_EOF;
}
{
if (!FLAC__metadata_object_seektable_template_append_spaced_points_by_samples(p->metadata[p->num_metadata], (unsigned)(10 * ft->signal.rate + .5), (FLAC__uint64)(ft->signal.length/ft->signal.channels))) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table points");
return SOX_EOF;
}
}
p->metadata[p->num_metadata]->is_last = sox_false; /* the encoder will set this for us */
++p->num_metadata;
}
if (ft->oob.comments) { /* Make the comment structure */
FLAC__StreamMetadata_VorbisComment_Entry entry;
int i;
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT);
for (i = 0; ft->oob.comments[i]; ++i) {
static const char prepend[] = "Comment=";
char * text = lsx_calloc(strlen(prepend) + strlen(ft->oob.comments[i]) + 1, sizeof(*text));
/* Prepend `Comment=' if no field-name already in the comment */
if (!strchr(ft->oob.comments[i], '='))
strcpy(text, prepend);
entry.entry = (FLAC__byte *) strcat(text, ft->oob.comments[i]);
entry.length = strlen(text);
FLAC__metadata_object_vorbiscomment_append_comment(p->metadata[p->num_metadata], entry, /*copy= */ sox_true);
free(text);
}
++p->num_metadata;
}
if (p->num_metadata)
FLAC__stream_encoder_set_metadata(p->encoder, p->metadata, p->num_metadata);
status = FLAC__stream_encoder_init_stream(p->encoder, flac_stream_encoder_write_callback,
flac_stream_encoder_seek_callback, flac_stream_encoder_tell_callback, flac_stream_encoder_metadata_callback, ft);
if (status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
lsx_fail_errno(ft, SOX_EINVAL, "%s", FLAC__StreamEncoderInitStatusString[status]);
return SOX_EOF;
}
return SOX_SUCCESS;
}
static size_t write_samples(sox_format_t * const ft, sox_sample_t const * const sampleBuffer, size_t const len)
{
priv_t * p = (priv_t *)ft->priv;
unsigned i;
/* allocate or grow buffer */
if (p->number_of_samples < len) {
p->number_of_samples = len;
free(p->decoded_samples);
p->decoded_samples = lsx_malloc(p->number_of_samples * sizeof(FLAC__int32));
}
for (i = 0; i < len; ++i) {
SOX_SAMPLE_LOCALS;
long pcm = SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i], ft->clips);
p->decoded_samples[i] = pcm >> (32 - p->bits_per_sample);
switch (p->bits_per_sample) {
case 8: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_8BIT(sampleBuffer[i], ft->clips);
break;
case 16: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_16BIT(sampleBuffer[i], ft->clips);
break;
case 24: p->decoded_samples[i] = /* sign extension: */
SOX_SAMPLE_TO_SIGNED_24BIT(sampleBuffer[i],ft->clips) << 8;
p->decoded_samples[i] >>= 8;
break;
case 32: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i],ft->clips);
break;
}
}
FLAC__stream_encoder_process_interleaved(p->encoder, p->decoded_samples, (unsigned) len / ft->signal.channels);
return FLAC__stream_encoder_get_state(p->encoder) == FLAC__STREAM_ENCODER_OK ? len : 0;
}
static int stop_write(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
FLAC__StreamEncoderState state = FLAC__stream_encoder_get_state(p->encoder);
unsigned i;
FLAC__stream_encoder_finish(p->encoder);
FLAC__stream_encoder_delete(p->encoder);
for (i = 0; i < p->num_metadata; ++i)
FLAC__metadata_object_delete(p->metadata[i]);
free(p->decoded_samples);
if (state != FLAC__STREAM_ENCODER_OK) {
lsx_fail_errno(ft, SOX_EINVAL, "FLAC ERROR: failed to encode to end of stream");
return SOX_EOF;
}
return SOX_SUCCESS;
}
static int seek(sox_format_t * ft, uint64_t offset)
{
priv_t * p = (priv_t *)ft->priv;
p->seek_offset = offset;
p->seek_pending = sox_true;
return ft->mode == 'r' ? SOX_SUCCESS : SOX_EOF;
}
LSX_FORMAT_HANDLER(flac)
{
static char const * const names[] = {"flac", NULL};
static unsigned const encodings[] = {SOX_ENCODING_FLAC, 8, 16, 24, 0, 0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Free Lossless Audio CODEC compressed audio", names, 0,
start_read, read_samples, stop_read,
start_write, write_samples, stop_write,
seek, encodings, NULL, sizeof(priv_t)
};
return &handler;
}

275
app/src/main/cpp/sox/flanger.c
Unescape View File

@ -0,0 +1,275 @@
/* libSoX effect: Stereo Flanger (c) 2006 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* TODO: Slide in the delay at the start? */
#include "sox_i.h"
#include <string.h>
typedef enum {INTERP_LINEAR, INTERP_QUADRATIC} interp_t;
#define MAX_CHANNELS 4
typedef struct {
/* Parameters */
double delay_min;
double delay_depth;
double feedback_gain;
double delay_gain;
double speed;
lsx_wave_t wave_shape;
double channel_phase;
interp_t interpolation;
/* Delay buffers */
double * delay_bufs[MAX_CHANNELS];
size_t delay_buf_length;
size_t delay_buf_pos;
double delay_last[MAX_CHANNELS];
/* Low Frequency Oscillator */
float * lfo;
size_t lfo_length;
size_t lfo_pos;
/* Balancing */
double in_gain;
} priv_t;
static lsx_enum_item const interp_enum[] = {
LSX_ENUM_ITEM(INTERP_,LINEAR)
LSX_ENUM_ITEM(INTERP_,QUADRATIC)
{0, 0}};
static int getopts(sox_effect_t * effp, int argc, char *argv[])
{
priv_t * p = (priv_t *) effp->priv;
--argc, ++argv;
/* Set non-zero defaults: */
p->delay_depth = 2;
p->delay_gain = 71;
p->speed = 0.5;
p->channel_phase= 25;
do { /* break-able block */
NUMERIC_PARAMETER(delay_min , 0 , 30 )
NUMERIC_PARAMETER(delay_depth , 0 , 10 )
NUMERIC_PARAMETER(feedback_gain,-95 , 95 )
NUMERIC_PARAMETER(delay_gain , 0 , 100)
NUMERIC_PARAMETER(speed , 0.1, 10 )
TEXTUAL_PARAMETER(wave_shape, lsx_get_wave_enum())
NUMERIC_PARAMETER(channel_phase, 0 , 100)
TEXTUAL_PARAMETER(interpolation, interp_enum)
} while (0);
if (argc != 0)
return lsx_usage(effp);
lsx_report("parameters:\n"
"delay = %gms\n"
"depth = %gms\n"
"regen = %g%%\n"
"width = %g%%\n"
"speed = %gHz\n"
"shape = %s\n"
"phase = %g%%\n"
"interp= %s",
p->delay_min,
p->delay_depth,
p->feedback_gain,
p->delay_gain,
p->speed,
lsx_get_wave_enum()[p->wave_shape].text,
p->channel_phase,
interp_enum[p->interpolation].text);
/* Scale to unity: */
p->feedback_gain /= 100;
p->delay_gain /= 100;
p->channel_phase /= 100;
p->delay_min /= 1000;
p->delay_depth /= 1000;
return SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * f = (priv_t *) effp->priv;
int c, channels = effp->in_signal.channels;
if (channels > MAX_CHANNELS) {
lsx_fail("Can not operate with more than %i channels", MAX_CHANNELS);
return SOX_EOF;
}
/* Balance output: */
f->in_gain = 1 / (1 + f->delay_gain);
f->delay_gain /= 1 + f->delay_gain;
/* Balance feedback loop: */
f->delay_gain *= 1 - fabs(f->feedback_gain);
lsx_debug("in_gain=%g feedback_gain=%g delay_gain=%g\n",
f->in_gain, f->feedback_gain, f->delay_gain);
/* Create the delay buffers, one for each channel: */
f->delay_buf_length =
(f->delay_min + f->delay_depth) * effp->in_signal.rate + 0.5;
++f->delay_buf_length; /* Need 0 to n, i.e. n + 1. */
++f->delay_buf_length; /* Quadratic interpolator needs one more. */
for (c = 0; c < channels; ++c)
f->delay_bufs[c] = lsx_calloc(f->delay_buf_length, sizeof(*f->delay_bufs[0]));
/* Create the LFO lookup table: */
f->lfo_length = effp->in_signal.rate / f->speed;
f->lfo = lsx_calloc(f->lfo_length, sizeof(*f->lfo));
lsx_generate_wave_table(
f->wave_shape,
SOX_FLOAT,
f->lfo,
f->lfo_length,
floor(f->delay_min * effp->in_signal.rate + .5),
f->delay_buf_length - 2.,
3 * M_PI_2); /* Start the sweep at minimum delay (for mono at least) */
lsx_debug("delay_buf_length=%" PRIuPTR " lfo_length=%" PRIuPTR "\n",
f->delay_buf_length, f->lfo_length);
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, sox_sample_t const * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * f = (priv_t *) effp->priv;
int c, channels = effp->in_signal.channels;
size_t len = (*isamp > *osamp ? *osamp : *isamp) / channels;
*isamp = *osamp = len * channels;
while (len--) {
f->delay_buf_pos =
(f->delay_buf_pos + f->delay_buf_length - 1) % f->delay_buf_length;
for (c = 0; c < channels; ++c) {
double delayed_0, delayed_1;
double delayed;
double in, out;
size_t channel_phase = c * f->lfo_length * f->channel_phase + .5;
double delay = f->lfo[(f->lfo_pos + channel_phase) % f->lfo_length];
double frac_delay = modf(delay, &delay);
size_t int_delay = (size_t)delay;
in = *ibuf++;
f->delay_bufs[c][f->delay_buf_pos] = in + f->delay_last[c] * f->feedback_gain;
delayed_0 = f->delay_bufs[c]
[(f->delay_buf_pos + int_delay++) % f->delay_buf_length];
delayed_1 = f->delay_bufs[c]
[(f->delay_buf_pos + int_delay++) % f->delay_buf_length];
if (f->interpolation == INTERP_LINEAR)
delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
else /* if (f->interpolation == INTERP_QUADRATIC) */
{
double a, b;
double delayed_2 = f->delay_bufs[c]
[(f->delay_buf_pos + int_delay++) % f->delay_buf_length];
delayed_2 -= delayed_0;
delayed_1 -= delayed_0;
a = delayed_2 *.5 - delayed_1;
b = delayed_1 * 2 - delayed_2 *.5;
delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
}
f->delay_last[c] = delayed;
out = in * f->in_gain + delayed * f->delay_gain;
*obuf++ = SOX_ROUND_CLIP_COUNT(out, effp->clips);
}
f->lfo_pos = (f->lfo_pos + 1) % f->lfo_length;
}
return SOX_SUCCESS;
}
static int stop(sox_effect_t * effp)
{
priv_t * f = (priv_t *) effp->priv;
int c, channels = effp->in_signal.channels;
for (c = 0; c < channels; ++c)
free(f->delay_bufs[c]);
free(f->lfo);
memset(f, 0, sizeof(*f));
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_flanger_effect_fn(void)
{
static sox_effect_handler_t handler = {
"flanger", NULL, SOX_EFF_MCHAN,
getopts, start, flow, NULL, stop, NULL, sizeof(priv_t)};
static char const * lines[] = {
"[delay depth regen width speed shape phase interp]",
" .",
" /|regen",
" / |",
" +--( |------------+",
" | \\ | | .",
" _V_ \\| _______ | |\\ width ___",
" | | ' | | | | \\ | |",
" +-->| + |---->| DELAY |--+-->| )----->| |",
" | |___| |_______| | / | |",
" | delay : depth |/ | |",
" In | : interp ' | | Out",
" --->+ __:__ | + |--->",
" | | |speed | |",
" | | ~ |shape | |",
" | |_____|phase | |",
" +------------------------------------->| |",
" |___|",
" RANGE DEFAULT DESCRIPTION",
"delay 0 30 0 base delay in milliseconds",
"depth 0 10 2 added swept delay in milliseconds",
"regen -95 +95 0 percentage regeneration (delayed signal feedback)",
"width 0 100 71 percentage of delayed signal mixed with original",
"speed 0.1 10 0.5 sweeps per second (Hz) ",
"shape -- sin swept wave shape: sine|triangle",
"phase 0 100 25 swept wave percentage phase-shift for multi-channel",
" (e.g. stereo) flange; 0 = 100 = same phase on each channel",
"interp -- lin delay-line interpolation: linear|quadratic"
};
static char * usage;
handler.usage = lsx_usage_lines(&usage, lines, array_length(lines));
return &handler;
}

1351
app/src/main/cpp/sox/formats.c
View File

File diff suppressed because it is too large Load Diff

130
app/src/main/cpp/sox/formats.h
Unescape View File

@ -0,0 +1,130 @@
/* libSoX static formats list (c) 2006-9 Chris Bagwell and SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*-------------------------- Static format handlers --------------------------*/
FORMAT(aifc)
FORMAT(aiff)
FORMAT(al)
FORMAT(au)
FORMAT(avr)
FORMAT(cdr)
FORMAT(cvsd)
FORMAT(cvu)
FORMAT(dat)
FORMAT(dvms)
FORMAT(f4)
FORMAT(f8)
FORMAT(gsrt)
FORMAT(hcom)
FORMAT(htk)
FORMAT(ima)
FORMAT(la)
FORMAT(lu)
FORMAT(maud)
FORMAT(nul)
FORMAT(prc)
FORMAT(raw)
FORMAT(s1)
FORMAT(s2)
FORMAT(s3)
FORMAT(s4)
FORMAT(sf)
FORMAT(sln)
FORMAT(smp)
FORMAT(sounder)
FORMAT(soundtool)
FORMAT(sox)
FORMAT(sphere)
FORMAT(svx)
FORMAT(txw)
FORMAT(u1)
FORMAT(u2)
FORMAT(u3)
FORMAT(u4)
FORMAT(ul)
FORMAT(voc)
FORMAT(vox)
FORMAT(wav)
FORMAT(wve)
FORMAT(xa)
/*--------------------- Plugin or static format handlers ---------------------*/
#if defined HAVE_ALSA && (defined STATIC_ALSA || !defined HAVE_LIBLTDL)
FORMAT(alsa)
#endif
#if defined HAVE_AMRNB && (defined STATIC_AMRNB || !defined HAVE_LIBLTDL)
FORMAT(amr_nb)
#endif
#if defined HAVE_AMRWB && (defined STATIC_AMRWB || !defined HAVE_LIBLTDL)
FORMAT(amr_wb)
#endif
#if defined HAVE_AO && (defined STATIC_AO || !defined HAVE_LIBLTDL)
FORMAT(ao)
#endif
#if defined HAVE_COREAUDIO && (defined STATIC_COREAUDIO || !defined HAVE_LIBLTDL)
FORMAT(coreaudio)
#endif
#if defined HAVE_FLAC && (defined STATIC_FLAC || !defined HAVE_LIBLTDL)
FORMAT(flac)
#endif
#if defined HAVE_GSM && (defined STATIC_GSM || !defined HAVE_LIBLTDL)
FORMAT(gsm)
#endif
#if defined HAVE_LPC10 && (defined STATIC_LPC10 || !defined HAVE_LIBLTDL)
FORMAT(lpc10)
#endif
#if defined HAVE_MP3 && (defined STATIC_MP3 || !defined HAVE_LIBLTDL)
FORMAT(mp3)
#endif
#if defined HAVE_OPUS && (defined STATIC_OPUS || !defined HAVE_LIBLTDL)
FORMAT(opus)
#endif
#if defined HAVE_OSS && (defined STATIC_OSS || !defined HAVE_LIBLTDL)
FORMAT(oss)
#endif
#if defined HAVE_PULSEAUDIO && (defined STATIC_PULSEAUDIO || !defined HAVE_LIBLTDL)
FORMAT(pulseaudio)
#endif
#if defined HAVE_WAVEAUDIO && (defined STATIC_WAVEAUDIO || !defined HAVE_LIBLTDL)
FORMAT(waveaudio)
#endif
#if defined HAVE_SNDIO && (defined STATIC_SNDIO || !defined HAVE_LIBLTDL)
FORMAT(sndio)
#endif
#if defined HAVE_SNDFILE && (defined STATIC_SNDFILE || !defined HAVE_LIBLTDL)
FORMAT(sndfile)
FORMAT(caf)
FORMAT(fap)
FORMAT(mat4)
FORMAT(mat5)
FORMAT(paf)
FORMAT(pvf)
FORMAT(sd2)
FORMAT(w64)
FORMAT(xi)
#endif
#if defined HAVE_SUNAUDIO && (defined STATIC_SUNAUDIO || !defined HAVE_LIBLTDL)
FORMAT(sunau)
#endif
#if defined HAVE_OGGVORBIS && (defined STATIC_OGGVORBIS || !defined HAVE_LIBLTDL)
FORMAT(vorbis)
#endif
#if defined HAVE_WAVPACK && (defined STATIC_WAVPACK || !defined HAVE_LIBLTDL)
FORMAT(wavpack)
#endif

545
app/src/main/cpp/sox/formats_i.c
Unescape View File

@ -0,0 +1,545 @@
/* Implements a libSoX internal interface for use in implementing file formats.
* All public functions & data are prefixed with lsx_ .
*
* (c) 2005-8 Chris Bagwell and SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <string.h>
#include <sys/stat.h>
#include <stdarg.h>
void lsx_fail_errno(sox_format_t * ft, int sox_errno, const char *fmt, ...)
{
va_list args;
ft->sox_errno = sox_errno;
va_start(args, fmt);
#ifdef HAVE_VSNPRINTF
vsnprintf(ft->sox_errstr, sizeof(ft->sox_errstr), fmt, args);
#else
vsprintf(ft->sox_errstr, fmt, args);
#endif
va_end(args);
ft->sox_errstr[255] = '\0';
}
void lsx_set_signal_defaults(sox_format_t * ft)
{
if (!ft->signal.rate ) ft->signal.rate = SOX_DEFAULT_RATE;
if (!ft->signal.precision) ft->signal.precision = SOX_DEFAULT_PRECISION;
if (!ft->signal.channels ) ft->signal.channels = SOX_DEFAULT_CHANNELS;
if (!ft->encoding.bits_per_sample)
ft->encoding.bits_per_sample = ft->signal.precision;
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN)
ft->encoding.encoding = SOX_ENCODING_SIGN2;
}
int lsx_check_read_params(sox_format_t * ft, unsigned channels,
sox_rate_t rate, sox_encoding_t encoding, unsigned bits_per_sample,
uint64_t num_samples, sox_bool check_length)
{
ft->signal.length = ft->signal.length == SOX_IGNORE_LENGTH? SOX_UNSPEC : num_samples;
if (ft->seekable)
ft->data_start = lsx_tell(ft);
if (channels && ft->signal.channels && ft->signal.channels != channels)
lsx_warn("`%s': overriding number of channels", ft->filename);
else ft->signal.channels = channels;
if (rate && ft->signal.rate && ft->signal.rate != rate)
lsx_warn("`%s': overriding sample rate", ft->filename);
else ft->signal.rate = rate;
if (encoding && ft->encoding.encoding && ft->encoding.encoding != encoding)
lsx_warn("`%s': overriding encoding type", ft->filename);
else ft->encoding.encoding = encoding;
if (bits_per_sample && ft->encoding.bits_per_sample && ft->encoding.bits_per_sample != bits_per_sample)
lsx_warn("`%s': overriding encoding size", ft->filename);
ft->encoding.bits_per_sample = bits_per_sample;
if (check_length && ft->encoding.bits_per_sample && lsx_filelength(ft)) {
uint64_t calculated_length = div_bits(lsx_filelength(ft) - ft->data_start, ft->encoding.bits_per_sample);
if (!ft->signal.length)
ft->signal.length = calculated_length;
else if (num_samples != calculated_length)
lsx_warn("`%s': file header gives the total number of samples as %" PRIu64 " but file length indicates the number is in fact %" PRIu64, ft->filename, num_samples, calculated_length);
}
if (sox_precision(ft->encoding.encoding, ft->encoding.bits_per_sample))
return SOX_SUCCESS;
lsx_fail_errno(ft, EINVAL, "invalid format for this file type");
return SOX_EOF;
}
/* Read in a buffer of data of length len bytes.
* Returns number of bytes read.
*/
size_t lsx_readbuf(sox_format_t * ft, void *buf, size_t len)
{
size_t ret = fread(buf, (size_t) 1, len, (FILE*)ft->fp);
if (ret != len && ferror((FILE*)ft->fp))
lsx_fail_errno(ft, errno, "lsx_readbuf");
ft->tell_off += ret;
return ret;
}
/* Skip input without seeking. */
int lsx_skipbytes(sox_format_t * ft, size_t n)
{
unsigned char trash;
while (n--)
if (lsx_readb(ft, &trash) == SOX_EOF)
return (SOX_EOF);
return (SOX_SUCCESS);
}
/* Pad output. */
int lsx_padbytes(sox_format_t * ft, size_t n)
{
while (n--)
if (lsx_writeb(ft, '\0') == SOX_EOF)
return (SOX_EOF);
return (SOX_SUCCESS);
}
/* Write a buffer of data of length bytes.
* Returns number of bytes written.
*/
size_t lsx_writebuf(sox_format_t * ft, void const * buf, size_t len)
{
size_t ret = fwrite(buf, (size_t) 1, len, (FILE*)ft->fp);
if (ret != len) {
lsx_fail_errno(ft, errno, "error writing output file");
clearerr((FILE*)ft->fp); /* Allows us to seek back to write header */
}
ft->tell_off += ret;
return ret;
}
sox_uint64_t lsx_filelength(sox_format_t * ft)
{
struct stat st;
int ret = ft->fp ? fstat(fileno((FILE*)ft->fp), &st) : 0;
return (!ret && (st.st_mode & S_IFREG))? (uint64_t)st.st_size : 0;
}
int lsx_flush(sox_format_t * ft)
{
return fflush((FILE*)ft->fp);
}
off_t lsx_tell(sox_format_t * ft)
{
return ft->seekable? (off_t)ftello((FILE*)ft->fp) : (off_t)ft->tell_off;
}
int lsx_eof(sox_format_t * ft)
{
return feof((FILE*)ft->fp);
}
int lsx_error(sox_format_t * ft)
{
return ferror((FILE*)ft->fp);
}
void lsx_rewind(sox_format_t * ft)
{
rewind((FILE*)ft->fp);
ft->tell_off = 0;
}
void lsx_clearerr(sox_format_t * ft)
{
clearerr((FILE*)ft->fp);
ft->sox_errno = 0;
}
int lsx_unreadb(sox_format_t * ft, unsigned b)
{
return ungetc((int)b, ft->fp);
}
/* Implements traditional fseek() behavior. Meant to abstract out
* file operations so that they could one day also work on memory
* buffers.
*
* N.B. Can only seek forwards on non-seekable streams!
*/
int lsx_seeki(sox_format_t * ft, off_t offset, int whence)
{
if (ft->seekable == 0) {
/* If a stream peel off chars else EPERM */
if (whence == SEEK_CUR) {
while (offset > 0 && !feof((FILE*)ft->fp)) {
getc((FILE*)ft->fp);
offset--;
++ft->tell_off;
}
if (offset)
lsx_fail_errno(ft,SOX_EOF, "offset past EOF");
else
ft->sox_errno = SOX_SUCCESS;
} else
lsx_fail_errno(ft,SOX_EPERM, "file not seekable");
} else {
if (fseeko((FILE*)ft->fp, offset, whence) == -1)
lsx_fail_errno(ft,errno, "%s", strerror(errno));
else
ft->sox_errno = SOX_SUCCESS;
}
return ft->sox_errno;
}
int lsx_offset_seek(sox_format_t * ft, off_t byte_offset, off_t to_sample)
{
double wide_sample = to_sample - (to_sample % ft->signal.channels);
double to_d = wide_sample * ft->encoding.bits_per_sample / 8;
off_t to = to_d;
return (to != to_d)? SOX_EOF : lsx_seeki(ft, (byte_offset + to), SEEK_SET);
}
/* Read and write known datatypes in "machine format". Swap if indicated.
* They all return SOX_EOF on error and SOX_SUCCESS on success.
*/
/* Read n-char string (and possibly null-terminating).
* Stop reading and null-terminate string if either a 0 or \n is reached.
*/
int lsx_reads(sox_format_t * ft, char *c, size_t len)
{
char *sc;
char in;
sc = c;
do
{
if (lsx_readbuf(ft, &in, (size_t)1) != 1)
{
*sc = 0;
return (SOX_EOF);
}
if (in == 0 || in == '\n')
break;
*sc = in;
sc++;
} while (sc - c < (ptrdiff_t)len);
*sc = 0;
return(SOX_SUCCESS);
}
/* Write null-terminated string (without \0). */
int lsx_writes(sox_format_t * ft, char const * c)
{
if (lsx_writebuf(ft, c, strlen(c)) != strlen(c))
return(SOX_EOF);
return(SOX_SUCCESS);
}
/* return swapped 32-bit float */
static void lsx_swapf(float * f)
{
union {
uint32_t dw;
float f;
} u;
u.f= *f;
u.dw= (u.dw>>24) | ((u.dw>>8)&0xff00) | ((u.dw<<8)&0xff0000) | (u.dw<<24);
*f = u.f;
}
static void swap(void * data, size_t len)
{
uint8_t * bytes = (uint8_t *)data;
size_t i;
for (i = 0; i < len / 2; ++i) {
char tmp = bytes[i];
bytes[i] = bytes[len - 1 - i];
bytes[len - 1 - i] = tmp;
}
}
static double lsx_swapdf(double data)
{
swap(&data, sizeof(data));
return data;
}
static uint64_t lsx_swapqw(uint64_t data)
{
swap(&data, sizeof(data));
return data;
}
/* Lookup table to reverse the bit order of a byte. ie MSB become LSB */
static uint8_t const cswap[256] = {
0x00, 0x80, 0x40, 0xC0, 0x20, 0xA0, 0x60, 0xE0, 0x10, 0x90, 0x50, 0xD0,
0x30, 0xB0, 0x70, 0xF0, 0x08, 0x88, 0x48, 0xC8, 0x28, 0xA8, 0x68, 0xE8,
0x18, 0x98, 0x58, 0xD8, 0x38, 0xB8, 0x78, 0xF8, 0x04, 0x84, 0x44, 0xC4,
0x24, 0xA4, 0x64, 0xE4, 0x14, 0x94, 0x54, 0xD4, 0x34, 0xB4, 0x74, 0xF4,
0x0C, 0x8C, 0x4C, 0xCC, 0x2C, 0xAC, 0x6C, 0xEC, 0x1C, 0x9C, 0x5C, 0xDC,
0x3C, 0xBC, 0x7C, 0xFC, 0x02, 0x82, 0x42, 0xC2, 0x22, 0xA2, 0x62, 0xE2,
0x12, 0x92, 0x52, 0xD2, 0x32, 0xB2, 0x72, 0xF2, 0x0A, 0x8A, 0x4A, 0xCA,
0x2A, 0xAA, 0x6A, 0xEA, 0x1A, 0x9A, 0x5A, 0xDA, 0x3A, 0xBA, 0x7A, 0xFA,
0x06, 0x86, 0x46, 0xC6, 0x26, 0xA6, 0x66, 0xE6, 0x16, 0x96, 0x56, 0xD6,
0x36, 0xB6, 0x76, 0xF6, 0x0E, 0x8E, 0x4E, 0xCE, 0x2E, 0xAE, 0x6E, 0xEE,
0x1E, 0x9E, 0x5E, 0xDE, 0x3E, 0xBE, 0x7E, 0xFE, 0x01, 0x81, 0x41, 0xC1,
0x21, 0xA1, 0x61, 0xE1, 0x11, 0x91, 0x51, 0xD1, 0x31, 0xB1, 0x71, 0xF1,
0x09, 0x89, 0x49, 0xC9, 0x29, 0xA9, 0x69, 0xE9, 0x19, 0x99, 0x59, 0xD9,
0x39, 0xB9, 0x79, 0xF9, 0x05, 0x85, 0x45, 0xC5, 0x25, 0xA5, 0x65, 0xE5,
0x15, 0x95, 0x55, 0xD5, 0x35, 0xB5, 0x75, 0xF5, 0x0D, 0x8D, 0x4D, 0xCD,
0x2D, 0xAD, 0x6D, 0xED, 0x1D, 0x9D, 0x5D, 0xDD, 0x3D, 0xBD, 0x7D, 0xFD,
0x03, 0x83, 0x43, 0xC3, 0x23, 0xA3, 0x63, 0xE3, 0x13, 0x93, 0x53, 0xD3,
0x33, 0xB3, 0x73, 0xF3, 0x0B, 0x8B, 0x4B, 0xCB, 0x2B, 0xAB, 0x6B, 0xEB,
0x1B, 0x9B, 0x5B, 0xDB, 0x3B, 0xBB, 0x7B, 0xFB, 0x07, 0x87, 0x47, 0xC7,
0x27, 0xA7, 0x67, 0xE7, 0x17, 0x97, 0x57, 0xD7, 0x37, 0xB7, 0x77, 0xF7,
0x0F, 0x8F, 0x4F, 0xCF, 0x2F, 0xAF, 0x6F, 0xEF, 0x1F, 0x9F, 0x5F, 0xDF,
0x3F, 0xBF, 0x7F, 0xFF
};
/* Utilities to byte-swap values, use libc optimized macros if possible */
#define TWIDDLE_BYTE(ub, type) \
do { \
if (ft->encoding.reverse_bits) \
ub = cswap[ub]; \
if (ft->encoding.reverse_nibbles) \
ub = ((ub & 15) << 4) | (ub >> 4); \
} while (0);
#define TWIDDLE_WORD(uw, type) \
if (ft->encoding.reverse_bytes) \
uw = lsx_swap ## type(uw);
#define TWIDDLE_FLOAT(f, type) \
if (ft->encoding.reverse_bytes) \
lsx_swapf(&f);
/* N.B. This macro doesn't work for unaligned types (e.g. 3-byte
types). */
#define READ_FUNC(type, size, ctype, twiddle) \
size_t lsx_read_ ## type ## _buf( \
sox_format_t * ft, ctype *buf, size_t len) \
{ \
size_t n, nread; \
nread = lsx_readbuf(ft, buf, len * size) / size; \
for (n = 0; n < nread; n++) \
twiddle(buf[n], type); \
return nread; \
}
/* Unpack a 3-byte value from a uint8_t * */
#define sox_unpack3(p) (ft->encoding.reverse_bytes == MACHINE_IS_BIGENDIAN? \
((p)[0] | ((p)[1] << 8) | ((p)[2] << 16)) : \
((p)[2] | ((p)[1] << 8) | ((p)[0] << 16)))
/* This (slower) macro works for unaligned types (e.g. 3-byte types)
that need to be unpacked. */
#define READ_FUNC_UNPACK(type, size, ctype, twiddle) \
size_t lsx_read_ ## type ## _buf( \
sox_format_t * ft, ctype *buf, size_t len) \
{ \
size_t n, nread; \
uint8_t *data = lsx_malloc(size * len); \
nread = lsx_readbuf(ft, data, len * size) / size; \
for (n = 0; n < nread; n++) \
buf[n] = sox_unpack ## size(data + n * size); \
free(data); \
return n; \
}
READ_FUNC(b, 1, uint8_t, TWIDDLE_BYTE)
READ_FUNC(w, 2, uint16_t, TWIDDLE_WORD)
READ_FUNC_UNPACK(3, 3, sox_uint24_t, TWIDDLE_WORD)
READ_FUNC(dw, 4, uint32_t, TWIDDLE_WORD)
READ_FUNC(qw, 8, uint64_t, TWIDDLE_WORD)
READ_FUNC(f, sizeof(float), float, TWIDDLE_FLOAT)
READ_FUNC(df, sizeof(double), double, TWIDDLE_WORD)
#define READ1_FUNC(type, ctype) \
int lsx_read ## type(sox_format_t * ft, ctype * datum) { \
if (lsx_read_ ## type ## _buf(ft, datum, (size_t)1) == 1) \
return SOX_SUCCESS; \
if (!lsx_error(ft)) \
lsx_fail_errno(ft, errno, premature_eof); \
return SOX_EOF; \
}
static char const premature_eof[] = "premature EOF";
READ1_FUNC(b, uint8_t)
READ1_FUNC(w, uint16_t)
READ1_FUNC(3, sox_uint24_t)
READ1_FUNC(dw, uint32_t)
READ1_FUNC(qw, uint64_t)
READ1_FUNC(f, float)
READ1_FUNC(df, double)
int lsx_readchars(sox_format_t * ft, char * chars, size_t len)
{
size_t ret = lsx_readbuf(ft, chars, len);
if (ret == len)
return SOX_SUCCESS;
if (!lsx_error(ft))
lsx_fail_errno(ft, errno, premature_eof);
return SOX_EOF;
}
int lsx_read_fields(sox_format_t *ft, uint32_t *len, const char *spec, ...)
{
int err = SOX_SUCCESS;
va_list ap;
#define do_read(type, f, n) do { \
size_t nr; \
if (*len < n * sizeof(type)) { \
err = SOX_EOF; \
goto end; \
} \
nr = lsx_read_##f##_buf(ft, va_arg(ap, type *), r); \
if (nr != n) \
err = SOX_EOF; \
*len -= nr * sizeof(type); \
} while (0)
va_start(ap, spec);
while (*spec) {
unsigned long r = 1;
char c = *spec;
if (c >= '0' && c <= '9') {
char *next;
r = strtoul(spec, &next, 10);
spec = next;
c = *spec;
} else if (c == '*') {
r = va_arg(ap, int);
c = *++spec;
}
switch (c) {
case 'b': do_read(uint8_t, b, r); break;
case 'h': do_read(uint16_t, w, r); break;
case 'i': do_read(uint32_t, dw, r); break;
case 'q': do_read(uint64_t, qw, r); break;
case 'x': err = lsx_skipbytes(ft, r); break;
default:
lsx_fail("lsx_read_fields: invalid format character '%c'", c);
err = SOX_EOF;
break;
}
if (err)
break;
spec++;
}
end:
va_end(ap);
#undef do_read
return err;
}
/* N.B. This macro doesn't work for unaligned types (e.g. 3-byte
types). */
#define WRITE_FUNC(type, size, ctype, twiddle) \
size_t lsx_write_ ## type ## _buf( \
sox_format_t * ft, ctype *buf, size_t len) \
{ \
size_t n, nwritten; \
for (n = 0; n < len; n++) \
twiddle(buf[n], type); \
nwritten = lsx_writebuf(ft, buf, len * size); \
return nwritten / size; \
}
/* Pack a 3-byte value to a uint8_t * */
#define sox_pack3(p, v) do {if (ft->encoding.reverse_bytes == MACHINE_IS_BIGENDIAN)\
{(p)[0] = v & 0xff; (p)[1] = (v >> 8) & 0xff; (p)[2] = (v >> 16) & 0xff;} else \
{(p)[2] = v & 0xff; (p)[1] = (v >> 8) & 0xff; (p)[0] = (v >> 16) & 0xff;} \
} while (0)
/* This (slower) macro works for unaligned types (e.g. 3-byte types)
that need to be packed. */
#define WRITE_FUNC_PACK(type, size, ctype, twiddle) \
size_t lsx_write_ ## type ## _buf( \
sox_format_t * ft, ctype *buf, size_t len) \
{ \
size_t n, nwritten; \
uint8_t *data = lsx_malloc(size * len); \
for (n = 0; n < len; n++) \
sox_pack ## size(data + n * size, buf[n]); \
nwritten = lsx_writebuf(ft, data, len * size); \
free(data); \
return nwritten / size; \
}
WRITE_FUNC(b, 1, uint8_t, TWIDDLE_BYTE)
WRITE_FUNC(w, 2, uint16_t, TWIDDLE_WORD)
WRITE_FUNC_PACK(3, 3, sox_uint24_t, TWIDDLE_WORD)
WRITE_FUNC(dw, 4, uint32_t, TWIDDLE_WORD)
WRITE_FUNC(qw, 8, uint64_t, TWIDDLE_WORD)
WRITE_FUNC(f, sizeof(float), float, TWIDDLE_FLOAT)
WRITE_FUNC(df, sizeof(double), double, TWIDDLE_WORD)
#define WRITE1U_FUNC(type, ctype) \
int lsx_write ## type(sox_format_t * ft, unsigned d) \
{ ctype datum = (ctype)d; \
return lsx_write_ ## type ## _buf(ft, &datum, (size_t)1) == 1 ? SOX_SUCCESS : SOX_EOF; \
}
#define WRITE1S_FUNC(type, ctype) \
int lsx_writes ## type(sox_format_t * ft, signed d) \
{ ctype datum = (ctype)d; \
return lsx_write_ ## type ## _buf(ft, &datum, (size_t)1) == 1 ? SOX_SUCCESS : SOX_EOF; \
}
#define WRITE1_FUNC(type, ctype) \
int lsx_write ## type(sox_format_t * ft, ctype datum) \
{ \
return lsx_write_ ## type ## _buf(ft, &datum, (size_t)1) == 1 ? SOX_SUCCESS : SOX_EOF; \
}
WRITE1U_FUNC(b, uint8_t)
WRITE1U_FUNC(w, uint16_t)
WRITE1U_FUNC(3, sox_uint24_t)
WRITE1U_FUNC(dw, uint32_t)
WRITE1_FUNC(qw, uint64_t)
WRITE1S_FUNC(b, uint8_t)
WRITE1S_FUNC(w, uint16_t)
WRITE1_FUNC(df, double)
int lsx_writef(sox_format_t * ft, double datum)
{
float f = datum;
return lsx_write_f_buf(ft, &f, (size_t) 1) == 1 ? SOX_SUCCESS : SOX_EOF;
}

2421
app/src/main/cpp/sox/g711.c
View File

File diff suppressed because it is too large Load Diff

21
app/src/main/cpp/sox/g711.h
Unescape View File

@ -0,0 +1,21 @@
/* libSoX G711.h - include for G711 u-law and a-law conversion routines
*
* Copyright (C) 2001 Chris Bagwell
*
* Permission to use, copy, modify, and distribute this software and its
* documentation for any purpose and without fee is hereby granted, provided
* that the above copyright notice appear in all copies and that both that
* copyright notice and this permission notice appear in supporting
* documentation. This software is provided "as is" without express or
* implied warranty.
*/
extern const uint8_t lsx_13linear2alaw[0x2000];
extern const int16_t lsx_alaw2linear16[256];
#define sox_13linear2alaw(sw) (lsx_13linear2alaw[((sw) + 0x1000)])
#define sox_alaw2linear16(uc) (lsx_alaw2linear16[uc])
extern const uint8_t lsx_14linear2ulaw[0x4000];
extern const int16_t lsx_ulaw2linear16[256];
#define sox_14linear2ulaw(sw) (lsx_14linear2ulaw[((sw) + 0x2000)])
#define sox_ulaw2linear16(uc) (lsx_ulaw2linear16[uc])

167
app/src/main/cpp/sox/g721.c
Unescape View File

@ -0,0 +1,167 @@
/* This source code is a product of Sun Microsystems, Inc. and is provided
* for unrestricted use. Users may copy or modify this source code without
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on
* the part of Sun Microsystems, Inc. to assist in its use, correction,
* modification or enhancement.
*
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
* OR ANY PART THEREOF.
*
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
* or profits or other special, indirect and consequential damages, even if
* Sun has been advised of the possibility of such damages.
*
* Sun Microsystems, Inc.
* 2550 Garcia Avenue
* Mountain View, California 94043
*/
/*
* g721.c
*
* Description:
*
* g721_encoder(), g721_decoder()
*
* These routines comprise an implementation of the CCITT G.721 ADPCM
* coding algorithm. Essentially, this implementation is identical to
* the bit level description except for a few deviations which
* take advantage of work station attributes, such as hardware 2's
* complement arithmetic and large memory. Specifically, certain time
* consuming operations such as multiplications are replaced
* with lookup tables and software 2's complement operations are
* replaced with hardware 2's complement.
*
* The deviation from the bit level specification (lookup tables)
* preserves the bit level performance specifications.
*
* As outlined in the G.721 Recommendation, the algorithm is broken
* down into modules. Each section of code below is preceded by
* the name of the module which it is implementing.
*
*/
#include "sox_i.h"
#include "g72x.h"
#include "g711.h"
static const short qtab_721[7] = {-124, 80, 178, 246, 300, 349, 400};
/*
* Maps G.721 code word to reconstructed scale factor normalized log
* magnitude values.
*/
static const short _dqlntab[16] = {-2048, 4, 135, 213, 273, 323, 373, 425,
425, 373, 323, 273, 213, 135, 4, -2048};
/* Maps G.721 code word to log of scale factor multiplier. */
static const short _witab[16] = {-12, 18, 41, 64, 112, 198, 355, 1122,
1122, 355, 198, 112, 64, 41, 18, -12};
/*
* Maps G.721 code words to a set of values whose long and short
* term averages are computed and then compared to give an indication
* how stationary (steady state) the signal is.
*/
static const short _fitab[16] = {0, 0, 0, 0x200, 0x200, 0x200, 0x600, 0xE00,
0xE00, 0x600, 0x200, 0x200, 0x200, 0, 0, 0};
/*
* g721_encoder()
*
* Encodes the input vale of linear PCM, A-law or u-law data sl and returns
* the resulting code. -1 is returned for unknown input coding value.
*/
int g721_encoder(int sl, int in_coding, struct g72x_state *state_ptr)
{
short sezi, se, sez; /* ACCUM */
short d; /* SUBTA */
short sr; /* ADDB */
short y; /* MIX */
short dqsez; /* ADDC */
short dq, i;
switch (in_coding) { /* linearize input sample to 14-bit PCM */
case AUDIO_ENCODING_ALAW:
sl = sox_alaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_ULAW:
sl = sox_ulaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_LINEAR:
sl >>= 2; /* 14-bit dynamic range */
break;
default:
return (-1);
}
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
se = (sezi + predictor_pole(state_ptr)) >> 1; /* estimated signal */
d = sl - se; /* estimation difference */
/* quantize the prediction difference */
y = step_size(state_ptr); /* quantizer step size */
i = quantize(d, y, qtab_721, 7); /* i = ADPCM code */
dq = reconstruct(i & 8, _dqlntab[i], y); /* quantized est diff */
sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconst. signal */
dqsez = sr + sez - se; /* pole prediction diff. */
update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);
return (i);
}
/*
* g721_decoder()
*
* Description:
*
* Decodes a 4-bit code of G.721 encoded data of i and
* returns the resulting linear PCM, A-law or u-law value.
* return -1 for unknown out_coding value.
*/
int g721_decoder(int i, int out_coding, struct g72x_state *state_ptr)
{
short sezi, sei, sez, se; /* ACCUM */
short y; /* MIX */
short sr; /* ADDB */
short dq;
short dqsez;
i &= 0x0f; /* mask to get proper bits */
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
sei = sezi + predictor_pole(state_ptr);
se = sei >> 1; /* se = estimated signal */
y = step_size(state_ptr); /* dynamic quantizer step size */
dq = reconstruct(i & 0x08, _dqlntab[i], y); /* quantized diff. */
sr = (dq < 0) ? (se - (dq & 0x3FFF)) : se + dq; /* reconst. signal */
dqsez = sr - se + sez; /* pole prediction diff. */
update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);
switch (out_coding) {
case AUDIO_ENCODING_ALAW:
return (tandem_adjust_alaw(sr, se, y, i, 8, qtab_721));
case AUDIO_ENCODING_ULAW:
return (tandem_adjust_ulaw(sr, se, y, i, 8, qtab_721));
case AUDIO_ENCODING_LINEAR:
return (sr << 2); /* sr was 14-bit dynamic range */
default:
return (-1);
}
}

151
app/src/main/cpp/sox/g723_24.c
Unescape View File

@ -0,0 +1,151 @@
/* This source code is a product of Sun Microsystems, Inc. and is provided
* for unrestricted use. Users may copy or modify this source code without
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on
* the part of Sun Microsystems, Inc. to assist in its use, correction,
* modification or enhancement.
*
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
* OR ANY PART THEREOF.
*
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
* or profits or other special, indirect and consequential damages, even if
* Sun has been advised of the possibility of such damages.
*
* Sun Microsystems, Inc.
* 2550 Garcia Avenue
* Mountain View, California 94043
*/
/*
* g723_24.c
*
* Description:
*
* g723_24_encoder(), g723_24_decoder()
*
* These routines comprise an implementation of the CCITT G.723 24 Kbps
* ADPCM coding algorithm. Essentially, this implementation is identical to
* the bit level description except for a few deviations which take advantage
* of workstation attributes, such as hardware 2's complement arithmetic.
*
*/
#include "sox_i.h"
#include "g711.h"
#include "g72x.h"
/*
* Maps G.723_24 code word to reconstructed scale factor normalized log
* magnitude values.
*/
static const short _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};
/* Maps G.723_24 code word to log of scale factor multiplier. */
static const short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
/*
* Maps G.723_24 code words to a set of values whose long and short
* term averages are computed and then compared to give an indication
* how stationary (steady state) the signal is.
*/
static const short _fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};
static const short qtab_723_24[3] = {8, 218, 331};
/*
* g723_24_encoder()
*
* Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code.
* Returns -1 if invalid input coding value.
*/
int g723_24_encoder(int sl, int in_coding, struct g72x_state *state_ptr)
{
short sei, sezi, se, sez; /* ACCUM */
short d; /* SUBTA */
short y; /* MIX */
short sr; /* ADDB */
short dqsez; /* ADDC */
short dq, i;
switch (in_coding) { /* linearize input sample to 14-bit PCM */
case AUDIO_ENCODING_ALAW:
sl = sox_alaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_ULAW:
sl = sox_ulaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_LINEAR:
sl >>= 2; /* sl of 14-bit dynamic range */
break;
default:
return (-1);
}
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
sei = sezi + predictor_pole(state_ptr);
se = sei >> 1; /* se = estimated signal */
d = sl - se; /* d = estimation diff. */
/* quantize prediction difference d */
y = step_size(state_ptr); /* quantizer step size */
i = quantize(d, y, qtab_723_24, 3); /* i = ADPCM code */
dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */
sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */
dqsez = sr + sez - se; /* pole prediction diff. */
update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
return (i);
}
/*
* g723_24_decoder()
*
* Decodes a 3-bit CCITT G.723_24 ADPCM code and returns
* the resulting 16-bit linear PCM, A-law or u-law sample value.
* -1 is returned if the output coding is unknown.
*/
int g723_24_decoder(int i, int out_coding, struct g72x_state *state_ptr)
{
short sezi, sei, sez, se; /* ACCUM */
short y; /* MIX */
short sr; /* ADDB */
short dq;
short dqsez;
i &= 0x07; /* mask to get proper bits */
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
sei = sezi + predictor_pole(state_ptr);
se = sei >> 1; /* se = estimated signal */
y = step_size(state_ptr); /* adaptive quantizer step size */
dq = reconstruct(i & 0x04, _dqlntab[i], y); /* unquantize pred diff */
sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */
dqsez = sr - se + sez; /* pole prediction diff. */
update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
switch (out_coding) {
case AUDIO_ENCODING_ALAW:
return (tandem_adjust_alaw(sr, se, y, i, 4, qtab_723_24));
case AUDIO_ENCODING_ULAW:
return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24));
case AUDIO_ENCODING_LINEAR:
return (sr << 2); /* sr was of 14-bit dynamic range */
default:
return (-1);
}
}

171
app/src/main/cpp/sox/g723_40.c
Unescape View File

@ -0,0 +1,171 @@
/* This source code is a product of Sun Microsystems, Inc. and is provided
* for unrestricted use. Users may copy or modify this source code without
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on
* the part of Sun Microsystems, Inc. to assist in its use, correction,
* modification or enhancement.
*
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
* OR ANY PART THEREOF.
*
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
* or profits or other special, indirect and consequential damages, even if
* Sun has been advised of the possibility of such damages.
*
* Sun Microsystems, Inc.
* 2550 Garcia Avenue
* Mountain View, California 94043
*/
/*
* g723_40.c
*
* Description:
*
* g723_40_encoder(), g723_40_decoder()
*
* These routines comprise an implementation of the CCITT G.723 40Kbps
* ADPCM coding algorithm. Essentially, this implementation is identical to
* the bit level description except for a few deviations which
* take advantage of workstation attributes, such as hardware 2's
* complement arithmetic.
*
* The deviation from the bit level specification (lookup tables),
* preserves the bit level performance specifications.
*
* As outlined in the G.723 Recommendation, the algorithm is broken
* down into modules. Each section of code below is preceded by
* the name of the module which it is implementing.
*
*/
#include "sox_i.h"
#include "g711.h"
#include "g72x.h"
/*
* Maps G.723_40 code word to ructeconstructed scale factor normalized log
* magnitude values.
*/
static const short _dqlntab[32] = {-2048, -66, 28, 104, 169, 224, 274, 318,
358, 395, 429, 459, 488, 514, 539, 566,
566, 539, 514, 488, 459, 429, 395, 358,
318, 274, 224, 169, 104, 28, -66, -2048};
/* Maps G.723_40 code word to log of scale factor multiplier. */
static const short _witab[32] = {448, 448, 768, 1248, 1280, 1312, 1856, 3200,
4512, 5728, 7008, 8960, 11456, 14080, 16928, 22272,
22272, 16928, 14080, 11456, 8960, 7008, 5728, 4512,
3200, 1856, 1312, 1280, 1248, 768, 448, 448};
/*
* Maps G.723_40 code words to a set of values whose long and short
* term averages are computed and then compared to give an indication
* how stationary (steady state) the signal is.
*/
static const short _fitab[32] = {0, 0, 0, 0, 0, 0x200, 0x200, 0x200,
0x200, 0x200, 0x400, 0x600, 0x800, 0xA00, 0xC00, 0xC00,
0xC00, 0xC00, 0xA00, 0x800, 0x600, 0x400, 0x200, 0x200,
0x200, 0x200, 0x200, 0, 0, 0, 0, 0};
static const short qtab_723_40[15] = {-122, -16, 68, 139, 198, 250, 298, 339,
378, 413, 445, 475, 502, 528, 553};
/*
* g723_40_encoder()
*
* Encodes a 16-bit linear PCM, A-law or u-law input sample and retuens
* the resulting 5-bit CCITT G.723 40Kbps code.
* Returns -1 if the input coding value is invalid.
*/
int g723_40_encoder(int sl, int in_coding, struct g72x_state *state_ptr)
{
short sei, sezi, se, sez; /* ACCUM */
short d; /* SUBTA */
short y; /* MIX */
short sr; /* ADDB */
short dqsez; /* ADDC */
short dq, i;
switch (in_coding) { /* linearize input sample to 14-bit PCM */
case AUDIO_ENCODING_ALAW:
sl = sox_alaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_ULAW:
sl = sox_ulaw2linear16(sl) >> 2;
break;
case AUDIO_ENCODING_LINEAR:
sl >>= 2; /* sl of 14-bit dynamic range */
break;
default:
return (-1);
}
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
sei = sezi + predictor_pole(state_ptr);
se = sei >> 1; /* se = estimated signal */
d = sl - se; /* d = estimation difference */
/* quantize prediction difference */
y = step_size(state_ptr); /* adaptive quantizer step size */
i = quantize(d, y, qtab_723_40, 15); /* i = ADPCM code */
dq = reconstruct(i & 0x10, _dqlntab[i], y); /* quantized diff */
sr = (dq < 0) ? se - (dq & 0x7FFF) : se + dq; /* reconstructed signal */
dqsez = sr + sez - se; /* dqsez = pole prediction diff. */
update(5, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
return (i);
}
/*
* g723_40_decoder()
*
* Decodes a 5-bit CCITT G.723 40Kbps code and returns
* the resulting 16-bit linear PCM, A-law or u-law sample value.
* -1 is returned if the output coding is unknown.
*/
int g723_40_decoder(int i, int out_coding, struct g72x_state *state_ptr)
{
short sezi, sei, sez, se; /* ACCUM */
short y; /* MIX */
short sr; /* ADDB */
short dq;
short dqsez;
i &= 0x1f; /* mask to get proper bits */
sezi = predictor_zero(state_ptr);
sez = sezi >> 1;
sei = sezi + predictor_pole(state_ptr);
se = sei >> 1; /* se = estimated signal */
y = step_size(state_ptr); /* adaptive quantizer step size */
dq = reconstruct(i & 0x10, _dqlntab[i], y); /* estimation diff. */
sr = (dq < 0) ? (se - (dq & 0x7FFF)) : (se + dq); /* reconst. signal */
dqsez = sr - se + sez; /* pole prediction diff. */
update(5, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
switch (out_coding) {
case AUDIO_ENCODING_ALAW:
return (tandem_adjust_alaw(sr, se, y, i, 0x10, qtab_723_40));
case AUDIO_ENCODING_ULAW:
return (tandem_adjust_ulaw(sr, se, y, i, 0x10, qtab_723_40));
case AUDIO_ENCODING_LINEAR:
return (sr << 2); /* sr was of 14-bit dynamic range */
default:
return (-1);
}
}

575
app/src/main/cpp/sox/g72x.c
Unescape View File

@ -0,0 +1,575 @@
/* Common routines for G.721 and G.723 conversions.
*
* (c) SoX Contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*
* This code is based on code from Sun, which came with the following
* copyright notice:
* -----------------------------------------------------------------------
* This source code is a product of Sun Microsystems, Inc. and is provided
* for unrestricted use. Users may copy or modify this source code without
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on
* the part of Sun Microsystems, Inc. to assist in its use, correction,
* modification or enhancement.
*
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
* OR ANY PART THEREOF.
*
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
* or profits or other special, indirect and consequential damages, even if
* Sun has been advised of the possibility of such damages.
*
* Sun Microsystems, Inc.
* 2550 Garcia Avenue
* Mountain View, California 94043
* -----------------------------------------------------------------------
*/
#include "sox_i.h"
#include "g711.h"
#include "g72x.h"
static const char LogTable256[] =
{
0, 0, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3,
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7
};
static inline int log2plus1(int val)
{
/* From http://graphics.stanford.edu/~seander/bithacks.html#IntegerLogLookup */
unsigned int v = (unsigned int)val; /* 32-bit word to find the log of */
unsigned r; /* r will be lg(v) */
register unsigned int t, tt; /* temporaries */
if ((tt = v >> 16))
{
r = (t = tt >> 8) ? 24 + LogTable256[t] : 16 + LogTable256[tt];
}
else
{
r = (t = v >> 8) ? 8 + LogTable256[t] : LogTable256[v];
}
return r + 1;
}
/*
* quan()
*
* quantizes the input val against the table of size short integers.
* It returns i if table[i - 1] <= val < table[i].
*
* Using linear search for simple coding.
*/
static int quan(int val, short const *table, int size)
{
int i;
for (i = 0; i < size; i++)
if (val < *table++)
break;
return (i);
}
/*
* fmult()
*
* returns the integer product of the 14-bit integer "an" and
* "floating point" representation (4-bit exponent, 6-bit mantessa) "srn".
*/
static int fmult(int an, int srn)
{
short anmag, anexp, anmant;
short wanexp, wanmant;
short retval;
anmag = (an > 0) ? an : ((-an) & 0x1FFF);
anexp = log2plus1(anmag) - 6;
anmant = (anmag == 0) ? 32 :
(anexp >= 0) ? anmag >> anexp : anmag << -anexp;
wanexp = anexp + ((srn >> 6) & 0xF) - 13;
wanmant = (anmant * (srn & 077) + 0x30) >> 4;
retval = (wanexp >= 0) ? ((wanmant << wanexp) & 0x7FFF) :
(wanmant >> -wanexp);
return (((an ^ srn) < 0) ? -retval : retval);
}
/*
* g72x_init_state()
*
* This routine initializes and/or resets the g72x_state structure
* pointed to by 'state_ptr'.
* All the initial state values are specified in the CCITT G.721 document.
*/
void g72x_init_state(struct g72x_state *state_ptr)
{
int cnta;
state_ptr->yl = 34816;
state_ptr->yu = 544;
state_ptr->dms = 0;
state_ptr->dml = 0;
state_ptr->ap = 0;
for (cnta = 0; cnta < 2; cnta++) {
state_ptr->a[cnta] = 0;
state_ptr->pk[cnta] = 0;
state_ptr->sr[cnta] = 32;
}
for (cnta = 0; cnta < 6; cnta++) {
state_ptr->b[cnta] = 0;
state_ptr->dq[cnta] = 32;
}
state_ptr->td = 0;
}
/*
* predictor_zero()
*
* computes the estimated signal from 6-zero predictor.
*
*/
int predictor_zero(struct g72x_state *state_ptr)
{
int i;
int sezi;
sezi = fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
for (i = 1; i < 6; i++) /* ACCUM */
sezi += fmult(state_ptr->b[i] >> 2, state_ptr->dq[i]);
return (sezi);
}
/*
* predictor_pole()
*
* computes the estimated signal from 2-pole predictor.
*
*/
int predictor_pole(struct g72x_state *state_ptr)
{
return (fmult(state_ptr->a[1] >> 2, state_ptr->sr[1]) +
fmult(state_ptr->a[0] >> 2, state_ptr->sr[0]));
}
/*
* step_size()
*
* computes the quantization step size of the adaptive quantizer.
*
*/
int step_size(struct g72x_state *state_ptr)
{
int y;
int dif;
int al;
if (state_ptr->ap >= 256)
return (state_ptr->yu);
else {
y = state_ptr->yl >> 6;
dif = state_ptr->yu - y;
al = state_ptr->ap >> 2;
if (dif > 0)
y += (dif * al) >> 6;
else if (dif < 0)
y += (dif * al + 0x3F) >> 6;
return (y);
}
}
/*
* quantize()
*
* Given a raw sample, 'd', of the difference signal and a
* quantization step size scale factor, 'y', this routine returns the
* ADPCM codeword to which that sample gets quantized. The step
* size scale factor division operation is done in the log base 2 domain
* as a subtraction.
*/
int quantize(int d, int y, short const *table, int size)
{
short dqm; /* Magnitude of 'd' */
short exp; /* Integer part of base 2 log of 'd' */
short mant; /* Fractional part of base 2 log */
short dl; /* Log of magnitude of 'd' */
short dln; /* Step size scale factor normalized log */
int i;
/*
* LOG
*
* Compute base 2 log of 'd', and store in 'dl'.
*/
dqm = abs(d);
exp = log2plus1(dqm >> 1);
mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */
dl = (exp << 7) + mant;
/*
* SUBTB
*
* "Divide" by step size multiplier.
*/
dln = dl - (y >> 2);
/*
* QUAN
*
* Obtain codword i for 'd'.
*/
i = quan(dln, table, size);
if (d < 0) /* take 1's complement of i */
return ((size << 1) + 1 - i);
else if (i == 0) /* take 1's complement of 0 */
return ((size << 1) + 1); /* new in 1988 */
else
return (i);
}
/*
* reconstruct()
*
* Returns reconstructed difference signal 'dq' obtained from
* codeword 'i' and quantization step size scale factor 'y'.
* Multiplication is performed in log base 2 domain as addition.
*/
int reconstruct(int sign, int dqln, int y)
{
short dql; /* Log of 'dq' magnitude */
short dex; /* Integer part of log */
short dqt;
short dq; /* Reconstructed difference signal sample */
dql = dqln + (y >> 2); /* ADDA */
if (dql < 0) {
return ((sign) ? -0x8000 : 0);
} else { /* ANTILOG */
dex = (dql >> 7) & 15;
dqt = 128 + (dql & 127);
dq = (dqt << 7) >> (14 - dex);
return ((sign) ? (dq - 0x8000) : dq);
}
}
/*
* update()
*
* updates the state variables for each output code
*/
void update(int code_size, int y, int wi, int fi, int dq, int sr,
int dqsez, struct g72x_state *state_ptr)
{
int cnt;
short mag, exp; /* Adaptive predictor, FLOAT A */
short a2p=0; /* LIMC */
short a1ul; /* UPA1 */
short pks1; /* UPA2 */
short fa1;
char tr; /* tone/transition detector */
short ylint, thr2, dqthr;
short ylfrac, thr1;
short pk0;
pk0 = (dqsez < 0) ? 1 : 0; /* needed in updating predictor poles */
mag = dq & 0x7FFF; /* prediction difference magnitude */
/* TRANS */
ylint = state_ptr->yl >> 15; /* exponent part of yl */
ylfrac = (state_ptr->yl >> 10) & 0x1F; /* fractional part of yl */
thr1 = (32 + ylfrac) << ylint; /* threshold */
thr2 = (ylint > 9) ? 31 << 10 : thr1; /* limit thr2 to 31 << 10 */
dqthr = (thr2 + (thr2 >> 1)) >> 1; /* dqthr = 0.75 * thr2 */
if (state_ptr->td == 0) /* signal supposed voice */
tr = 0;
else if (mag <= dqthr) /* supposed data, but small mag */
tr = 0; /* treated as voice */
else /* signal is data (modem) */
tr = 1;
/*
* Quantizer scale factor adaptation.
*/
/* FUNCTW & FILTD & DELAY */
/* update non-steady state step size multiplier */
state_ptr->yu = y + ((wi - y) >> 5);
/* LIMB */
if (state_ptr->yu < 544) /* 544 <= yu <= 5120 */
state_ptr->yu = 544;
else if (state_ptr->yu > 5120)
state_ptr->yu = 5120;
/* FILTE & DELAY */
/* update steady state step size multiplier */
state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
/*
* Adaptive predictor coefficients.
*/
if (tr == 1) { /* reset a's and b's for modem signal */
state_ptr->a[0] = 0;
state_ptr->a[1] = 0;
state_ptr->b[0] = 0;
state_ptr->b[1] = 0;
state_ptr->b[2] = 0;
state_ptr->b[3] = 0;
state_ptr->b[4] = 0;
state_ptr->b[5] = 0;
} else { /* update a's and b's */
pks1 = pk0 ^ state_ptr->pk[0]; /* UPA2 */
/* update predictor pole a[1] */
a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
if (dqsez != 0) {
fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
if (fa1 < -8191) /* a2p = function of fa1 */
a2p -= 0x100;
else if (fa1 > 8191)
a2p += 0xFF;
else
a2p += fa1 >> 5;
if (pk0 ^ state_ptr->pk[1])
{
/* LIMC */
if (a2p <= -12160)
a2p = -12288;
else if (a2p >= 12416)
a2p = 12288;
else
a2p -= 0x80;
}
else if (a2p <= -12416)
a2p = -12288;
else if (a2p >= 12160)
a2p = 12288;
else
a2p += 0x80;
}
/* Possible bug: a2p not initialized if dqsez == 0) */
/* TRIGB & DELAY */
state_ptr->a[1] = a2p;
/* UPA1 */
/* update predictor pole a[0] */
state_ptr->a[0] -= state_ptr->a[0] >> 8;
if (dqsez != 0)
{
if (pks1 == 0)
state_ptr->a[0] += 192;
else
state_ptr->a[0] -= 192;
}
/* LIMD */
a1ul = 15360 - a2p;
if (state_ptr->a[0] < -a1ul)
state_ptr->a[0] = -a1ul;
else if (state_ptr->a[0] > a1ul)
state_ptr->a[0] = a1ul;
/* UPB : update predictor zeros b[6] */
for (cnt = 0; cnt < 6; cnt++) {
if (code_size == 5) /* for 40Kbps G.723 */
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 9;
else /* for G.721 and 24Kbps G.723 */
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
if (dq & 0x7FFF) { /* XOR */
if ((dq ^ state_ptr->dq[cnt]) >= 0)
state_ptr->b[cnt] += 128;
else
state_ptr->b[cnt] -= 128;
}
}
}
for (cnt = 5; cnt > 0; cnt--)
state_ptr->dq[cnt] = state_ptr->dq[cnt-1];
/* FLOAT A : convert dq[0] to 4-bit exp, 6-bit mantissa f.p. */
if (mag == 0) {
state_ptr->dq[0] = (dq >= 0) ? 0x20 : (short)(unsigned short)0xFC20;
} else {
exp = log2plus1(mag);
state_ptr->dq[0] = (dq >= 0) ?
(exp << 6) + ((mag << 6) >> exp) :
(exp << 6) + ((mag << 6) >> exp) - 0x400;
}
state_ptr->sr[1] = state_ptr->sr[0];
/* FLOAT B : convert sr to 4-bit exp., 6-bit mantissa f.p. */
if (sr == 0) {
state_ptr->sr[0] = 0x20;
} else if (sr > 0) {
exp = log2plus1(sr);
state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
} else if (sr > -32768) {
mag = -sr;
exp = log2plus1(mag);
state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400;
} else
state_ptr->sr[0] = (short)(unsigned short)0xFC20;
/* DELAY A */
state_ptr->pk[1] = state_ptr->pk[0];
state_ptr->pk[0] = pk0;
/* TONE */
if (tr == 1) /* this sample has been treated as data */
state_ptr->td = 0; /* next one will be treated as voice */
else if (a2p < -11776) /* small sample-to-sample correlation */
state_ptr->td = 1; /* signal may be data */
else /* signal is voice */
state_ptr->td = 0;
/*
* Adaptation speed control.
*/
state_ptr->dms += (fi - state_ptr->dms) >> 5; /* FILTA */
state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7); /* FILTB */
if (tr == 1)
state_ptr->ap = 256;
else if (y < 1536) /* SUBTC */
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else if (state_ptr->td == 1)
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else if (abs((state_ptr->dms << 2) - state_ptr->dml) >=
(state_ptr->dml >> 3))
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else
state_ptr->ap += (-state_ptr->ap) >> 4;
}
/*
* tandem_adjust(sr, se, y, i, sign)
*
* At the end of ADPCM decoding, it simulates an encoder which may be receiving
* the output of this decoder as a tandem process. If the output of the
* simulated encoder differs from the input to this decoder, the decoder output
* is adjusted by one level of A-law or u-law codes.
*
* Input:
* sr decoder output linear PCM sample,
* se predictor estimate sample,
* y quantizer step size,
* i decoder input code,
* sign sign bit of code i
*
* Return:
* adjusted A-law or u-law compressed sample.
*/
int tandem_adjust_alaw(int sr, int se, int y, int i, int sign, short const *qtab)
{
unsigned char sp; /* A-law compressed 8-bit code */
short dx; /* prediction error */
char id; /* quantized prediction error */
int sd; /* adjusted A-law decoded sample value */
int im; /* biased magnitude of i */
int imx; /* biased magnitude of id */
if (sr <= -32768)
sr = -1;
sp = sox_13linear2alaw(((sr >> 1) << 3));/* short to A-law compression */
dx = (sox_alaw2linear16(sp) >> 2) - se; /* 16-bit prediction error */
id = quantize(dx, y, qtab, sign - 1);
if (id == i) { /* no adjustment on sp */
return (sp);
} else { /* sp adjustment needed */
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
im = i ^ sign; /* 2's complement to biased unsigned */
imx = id ^ sign;
if (imx > im) { /* sp adjusted to next lower value */
if (sp & 0x80) {
sd = (sp == 0xD5) ? 0x55 :
((sp ^ 0x55) - 1) ^ 0x55;
} else {
sd = (sp == 0x2A) ? 0x2A :
((sp ^ 0x55) + 1) ^ 0x55;
}
} else { /* sp adjusted to next higher value */
if (sp & 0x80)
sd = (sp == 0xAA) ? 0xAA :
((sp ^ 0x55) + 1) ^ 0x55;
else
sd = (sp == 0x55) ? 0xD5 :
((sp ^ 0x55) - 1) ^ 0x55;
}
return (sd);
}
}
int tandem_adjust_ulaw(int sr, int se, int y, int i, int sign, short const *qtab)
{
unsigned char sp; /* u-law compressed 8-bit code */
short dx; /* prediction error */
char id; /* quantized prediction error */
int sd; /* adjusted u-law decoded sample value */
int im; /* biased magnitude of i */
int imx; /* biased magnitude of id */
if (sr <= -32768)
sr = 0;
sp = sox_14linear2ulaw((sr << 2));/* short to u-law compression */
dx = (sox_ulaw2linear16(sp) >> 2) - se; /* 16-bit prediction error */
id = quantize(dx, y, qtab, sign - 1);
if (id == i) {
return (sp);
} else {
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
im = i ^ sign; /* 2's complement to biased unsigned */
imx = id ^ sign;
if (imx > im) { /* sp adjusted to next lower value */
if (sp & 0x80)
sd = (sp == 0xFF) ? 0x7E : sp + 1;
else
sd = (sp == 0) ? 0 : sp - 1;
} else { /* sp adjusted to next higher value */
if (sp & 0x80)
sd = (sp == 0x80) ? 0x80 : sp - 1;
else
sd = (sp == 0x7F) ? 0xFE : sp + 1;
}
return (sd);
}
}

157
app/src/main/cpp/sox/g72x.h
Unescape View File

@ -0,0 +1,157 @@
/* This source code is a product of Sun Microsystems, Inc. and is provided
* for unrestricted use. Users may copy or modify this source code without
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on
* the part of Sun Microsystems, Inc. to assist in its use, correction,
* modification or enhancement.
*
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
* OR ANY PART THEREOF.
*
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
* or profits or other special, indirect and consequential damages, even if
* Sun has been advised of the possibility of such damages.
*
* Sun Microsystems, Inc.
* 2550 Garcia Avenue
* Mountain View, California 94043
*/
/*
* g72x.h
*
* Header file for CCITT conversion routines.
*
*/
#ifndef _G72X_H
#define _G72X_H
/* aliases */
#define g721_decoder lsx_g721_decoder
#define g721_encoder lsx_g721_encoder
#define g723_24_decoder lsx_g723_24_decoder
#define g723_24_encoder lsx_g723_24_encoder
#define g723_40_decoder lsx_g723_40_decoder
#define g723_40_encoder lsx_g723_40_encoder
#define g72x_init_state lsx_g72x_init_state
#define predictor_pole lsx_g72x_predictor_pole
#define predictor_zero lsx_g72x_predictor_zero
#define quantize lsx_g72x_quantize
#define reconstruct lsx_g72x_reconstruct
#define step_size lsx_g72x_step_size
#define tandem_adjust_alaw lsx_g72x_tandem_adjust_alaw
#define tandem_adjust_ulaw lsx_g72x_tandem_adjust_ulaw
#define update lsx_g72x_update
#define AUDIO_ENCODING_ULAW (1) /* ISDN u-law */
#define AUDIO_ENCODING_ALAW (2) /* ISDN A-law */
#define AUDIO_ENCODING_LINEAR (3) /* PCM 2's-complement (0-center) */
/*
* The following is the definition of the state structure
* used by the G.721/G.723 encoder and decoder to preserve their internal
* state between successive calls. The meanings of the majority
* of the state structure fields are explained in detail in the
* CCITT Recommendation G.721. The field names are essentially indentical
* to variable names in the bit level description of the coding algorithm
* included in this Recommendation.
*/
struct g72x_state {
long yl; /* Locked or steady state step size multiplier. */
short yu; /* Unlocked or non-steady state step size multiplier. */
short dms; /* Short term energy estimate. */
short dml; /* Long term energy estimate. */
short ap; /* Linear weighting coefficient of 'yl' and 'yu'. */
short a[2]; /* Coefficients of pole portion of prediction filter. */
short b[6]; /* Coefficients of zero portion of prediction filter. */
short pk[2]; /*
* Signs of previous two samples of a partially
* reconstructed signal.
*/
short dq[6]; /*
* Previous 6 samples of the quantized difference
* signal represented in an internal floating point
* format.
*/
short sr[2]; /*
* Previous 2 samples of the quantized difference
* signal represented in an internal floating point
* format.
*/
char td; /* delayed tone detect, new in 1988 version */
};
/* External function definitions. */
extern void g72x_init_state(struct g72x_state *);
extern int g721_encoder(
int sample,
int in_coding,
struct g72x_state *state_ptr);
extern int g721_decoder(
int code,
int out_coding,
struct g72x_state *state_ptr);
extern int g723_16_encoder(
int sample,
int in_coding,
struct g72x_state *state_ptr);
extern int g723_16_decoder(
int code,
int out_coding,
struct g72x_state *state_ptr);
extern int g723_24_encoder(
int sample,
int in_coding,
struct g72x_state *state_ptr);
extern int g723_24_decoder(
int code,
int out_coding,
struct g72x_state *state_ptr);
extern int g723_40_encoder(
int sample,
int in_coding,
struct g72x_state *state_ptr);
extern int g723_40_decoder(
int code,
int out_coding,
struct g72x_state *state_ptr);
int predictor_zero(struct g72x_state *state_ptr);
int predictor_pole(struct g72x_state *state_ptr);
int step_size(struct g72x_state *state_ptr);
int quantize(int d,
int y,
short const *table,
int size);
int reconstruct(int sign,
int dqln,
int y);
void update(int code_size,
int y,
int wi,
int fi,
int dq,
int sr,
int dqsez,
struct g72x_state *state_ptr);
int tandem_adjust_alaw(int sr,
int se,
int y,
int i,
int sign,
short const *qtab);
int tandem_adjust_ulaw(int sr,
int se,
int y,
int i,
int sign,
short const *qtab);
#endif /* !_G72X_H */

276
app/src/main/cpp/sox/gain.c
Unescape View File

@ -0,0 +1,276 @@
/* libSoX effect: gain/norm/etc. (c) 2008-9 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define LSX_EFF_ALIAS
#include "sox_i.h"
#include <ctype.h>
#include <string.h>
typedef struct {
sox_bool do_equalise, do_balance, do_balance_no_clip, do_limiter;
sox_bool do_restore, make_headroom, do_normalise, do_scan;
double fixed_gain; /* Valid only in channel 0 */
double mult, reclaim, rms, limiter;
off_t num_samples;
sox_sample_t min, max;
FILE * tmp_file;
} priv_t;
static int create(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
char const * q;
for (--argc, ++argv; argc && **argv == '-' && argv[0][1] &&
!isdigit((unsigned char)argv[0][1]) && argv[0][1] != '.'; --argc, ++argv)
for (q = &argv[0][1]; *q; ++q) switch (*q) {
case 'n': p->do_scan = p->do_normalise = sox_true; break;
case 'e': p->do_scan = p->do_equalise = sox_true; break;
case 'B': p->do_scan = p->do_balance = sox_true; break;
case 'b': p->do_scan = p->do_balance_no_clip = sox_true; break;
case 'r': p->do_scan = p->do_restore = sox_true; break;
case 'h': p->make_headroom = sox_true; break;
case 'l': p->do_limiter = sox_true; break;
default: lsx_fail("invalid option `-%c'", *q); return lsx_usage(effp);
}
if ((p->do_equalise + p->do_balance + p->do_balance_no_clip + p->do_restore)/ sox_true > 1) {
lsx_fail("only one of -e, -B, -b, -r may be given");
return SOX_EOF;
}
if (p->do_normalise && p->do_restore) {
lsx_fail("only one of -n, -r may be given");
return SOX_EOF;
}
if (p->do_limiter && p->make_headroom) {
lsx_fail("only one of -l, -h may be given");
return SOX_EOF;
}
do {NUMERIC_PARAMETER(fixed_gain, -HUGE_VAL, HUGE_VAL)} while (0);
p->fixed_gain = dB_to_linear(p->fixed_gain);
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
if (effp->flow == 0) {
if (p->do_restore) {
if (!effp->in_signal.mult || *effp->in_signal.mult >= 1) {
lsx_fail("can't reclaim headroom");
return SOX_EOF;
}
p->reclaim = 1 / *effp->in_signal.mult;
}
effp->out_signal.mult = p->make_headroom? &p->fixed_gain : NULL;
if (!p->do_equalise && !p->do_balance && !p->do_balance_no_clip)
effp->flows = 1; /* essentially a conditional SOX_EFF_MCHAN */
}
p->mult = 0;
p->max = 1;
p->min = -1;
if (p->do_scan) {
p->tmp_file = lsx_tmpfile();
if (p->tmp_file == NULL) {
lsx_fail("can't create temporary file: %s", strerror(errno));
return SOX_EOF;
}
}
if (p->do_limiter)
p->limiter = (1 - 1 / p->fixed_gain) * (1. / SOX_SAMPLE_MAX);
else if (p->fixed_gain == floor(p->fixed_gain) && !p->do_scan)
effp->out_signal.precision = effp->in_signal.precision;
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len;
if (p->do_scan) {
if (fwrite(ibuf, sizeof(*ibuf), *isamp, p->tmp_file) != *isamp) {
lsx_fail("error writing temporary file: %s", strerror(errno));
return SOX_EOF;
}
if (p->do_balance && !p->do_normalise)
for (len = *isamp; len; --len, ++ibuf) {
double d = SOX_SAMPLE_TO_FLOAT_64BIT(*ibuf, effp->clips);
p->rms += sqr(d);
++p->num_samples;
}
else if (p->do_balance || p->do_balance_no_clip)
for (len = *isamp; len; --len, ++ibuf) {
double d = SOX_SAMPLE_TO_FLOAT_64BIT(*ibuf, effp->clips);
p->rms += sqr(d);
++p->num_samples;
p->max = max(p->max, *ibuf);
p->min = min(p->min, *ibuf);
}
else for (len = *isamp; len; --len, ++ibuf) {
p->max = max(p->max, *ibuf);
p->min = min(p->min, *ibuf);
}
*osamp = 0; /* samples not output until drain */
}
else {
double mult = ((priv_t *)(effp - effp->flow)->priv)->fixed_gain;
len = *isamp = *osamp = min(*isamp, *osamp);
if (!p->do_limiter) for (; len; --len, ++ibuf)
*obuf++ = SOX_ROUND_CLIP_COUNT(*ibuf * mult, effp->clips);
else for (; len; --len, ++ibuf) {
double d = *ibuf * mult;
*obuf++ = d < 0 ? 1 / (1 / d - p->limiter) - .5 :
d > 0 ? 1 / (1 / d + p->limiter) + .5 : 0;
}
}
return SOX_SUCCESS;
}
static void start_drain(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
double max = SOX_SAMPLE_MAX, max_peak = 0, max_rms = 0;
size_t i;
if (p->do_balance || p->do_balance_no_clip) {
for (i = 0; i < effp->flows; ++i) {
priv_t * q = (priv_t *)(effp - effp->flow + i)->priv;
max_rms = max(max_rms, sqrt(q->rms / q->num_samples));
rewind(q->tmp_file);
}
for (i = 0; i < effp->flows; ++i) {
priv_t * q = (priv_t *)(effp - effp->flow + i)->priv;
double this_rms = sqrt(q->rms / q->num_samples);
double this_peak = max(q->max / max, q->min / (double)SOX_SAMPLE_MIN);
q->mult = this_rms != 0? max_rms / this_rms : 1;
max_peak = max(max_peak, q->mult * this_peak);
q->mult *= p->fixed_gain;
}
if (p->do_normalise || (p->do_balance_no_clip && max_peak > 1))
for (i = 0; i < effp->flows; ++i) {
priv_t * q = (priv_t *)(effp - effp->flow + i)->priv;
q->mult /= max_peak;
}
} else if (p->do_equalise && !p->do_normalise) {
for (i = 0; i < effp->flows; ++i) {
priv_t * q = (priv_t *)(effp - effp->flow + i)->priv;
double this_peak = max(q->max / max, q->min / (double)SOX_SAMPLE_MIN);
max_peak = max(max_peak, this_peak);
q->mult = p->fixed_gain / this_peak;
rewind(q->tmp_file);
}
for (i = 0; i < effp->flows; ++i) {
priv_t * q = (priv_t *)(effp - effp->flow + i)->priv;
q->mult *= max_peak;
}
} else {
p->mult = min(max / p->max, (double)SOX_SAMPLE_MIN / p->min);
if (p->do_restore) {
if (p->reclaim > p->mult)
lsx_report("%.3gdB not reclaimed", linear_to_dB(p->reclaim / p->mult));
else p->mult = p->reclaim;
}
p->mult *= p->fixed_gain;
rewind(p->tmp_file);
}
}
static int drain(sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t len;
int result = SOX_SUCCESS;
*osamp -= *osamp % effp->in_signal.channels;
if (p->do_scan) {
if (!p->mult)
start_drain(effp);
len = fread(obuf, sizeof(*obuf), *osamp, p->tmp_file);
if (len != *osamp && !feof(p->tmp_file)) {
lsx_fail("error reading temporary file: %s", strerror(errno));
result = SOX_EOF;
}
if (!p->do_limiter) for (*osamp = len; len; --len, ++obuf)
*obuf = SOX_ROUND_CLIP_COUNT(*obuf * p->mult, effp->clips);
else for (*osamp = len; len; --len) {
double d = *obuf * p->mult;
*obuf++ = d < 0 ? 1 / (1 / d - p->limiter) - .5 :
d > 0 ? 1 / (1 / d + p->limiter) + .5 : 0;
}
}
else *osamp = 0;
return result;
}
static int stop(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
if (p->do_scan)
fclose(p->tmp_file); /* auto-deleted by lsx_tmpfile */
return SOX_SUCCESS;
}
sox_effect_handler_t const * lsx_gain_effect_fn(void)
{
static sox_effect_handler_t handler = {
"gain", NULL, SOX_EFF_GAIN,
create, start, flow, drain, stop, NULL, sizeof(priv_t)};
static char const * lines[] = {
"[-e|-b|-B|-r] [-n] [-l|-h] [gain-dB]",
"-e\t Equalise channels: peak to that with max peak;",
"-B\t Balance channels: rms to that with max rms; no clip protection",
"-b\t Balance channels: rms to that with max rms; clip protection",
"\t Note -Bn = -bn",
"-r\t Reclaim headroom (as much as possible without clipping); see -h",
"-n\t Norm file to 0dBfs(output precision); gain-dB, if present, usually <0",
"-l\t Use simple limiter",
"-h\t Apply attenuation for headroom for subsequent effects; gain-dB, if",
"\t present, is subject to reclaim by a subsequent gain -r",
"gain-dB\t Apply gain in dB",
};
static char * usage;
handler.usage = lsx_usage_lines(&usage, lines, array_length(lines));
return &handler;
}
/*------------------ emulation of the old `normalise' effect -----------------*/
static int norm_getopts(sox_effect_t * effp, int argc, char * * argv)
{
char * argv2[3];
int argc2 = 2;
argv2[0] = argv[0], --argc, ++argv;
argv2[1] = "-n";
if (argc)
argv2[argc2++] = *argv, --argc, ++argv;
return argc? lsx_usage(effp) :
lsx_gain_effect_fn()->getopts(effp, argc2, argv2);
}
sox_effect_handler_t const * lsx_norm_effect_fn(void)
{
static sox_effect_handler_t handler;
handler = *lsx_gain_effect_fn();
handler.name = "norm";
handler.usage = "[level]";
handler.getopts = norm_getopts;
return &handler;
}

349
app/src/main/cpp/sox/getopt.c
Unescape View File

@ -0,0 +1,349 @@
/* lsx_getopt for SoX
*
* (c) 2011 Doug Cook and SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox.h"
#include <assert.h>
#include <stdlib.h>
#include <string.h>
void
lsx_getopt_init(
LSX_PARAM_IN int argc, /* Number of arguments in argv */
LSX_PARAM_IN_COUNT(argc) char * const * argv, /* Array of arguments */
LSX_PARAM_IN_Z char const * shortopts, /* Short option characters */
LSX_PARAM_IN_OPT lsx_option_t const * longopts, /* Array of long option descriptors */
LSX_PARAM_IN lsx_getopt_flags_t flags, /* Flags for longonly and opterr */
LSX_PARAM_IN int first, /* First argument to check (usually 1) */
LSX_PARAM_OUT lsx_getopt_t * state) /* State object to initialize */
{
assert(argc >= 0);
assert(argv != NULL);
assert(shortopts);
assert(first >= 0);
assert(first <= argc);
assert(state);
if (state)
{
if (argc < 0 ||
!argv ||
!shortopts ||
first < 0 ||
first > argc)
{
memset(state, 0, sizeof(*state));
}
else
{
state->argc = argc;
state->argv = argv;
state->shortopts =
(shortopts[0] == '+' || shortopts[0] == '-') /* Requesting GNU special behavior? */
? shortopts + 1 /* Ignore request. */
: shortopts; /* No special behavior requested. */
state->longopts = longopts;
state->flags = flags;
state->curpos = NULL;
state->ind = first;
state->opt = '?';
state->arg = NULL;
state->lngind = -1;
}
}
}
static void CheckCurPosEnd(
LSX_PARAM_INOUT lsx_getopt_t * state)
{
if (!state->curpos[0])
{
state->curpos = NULL;
state->ind++;
}
}
int
lsx_getopt(
LSX_PARAM_INOUT lsx_getopt_t * state)
{
int oerr;
assert(state);
if (!state)
{
lsx_fail("lsx_getopt called with state=NULL");
return -1;
}
assert(state->argc >= 0);
assert(state->argv != NULL);
assert(state->shortopts);
assert(state->ind >= 0);
assert(state->ind <= state->argc + 1);
oerr = 0 != (state->flags & lsx_getopt_flag_opterr);
state->opt = 0;
state->arg = NULL;
state->lngind = -1;
if (state->argc < 0 ||
!state->argv ||
!state->shortopts ||
state->ind < 0)
{ /* programmer error */
lsx_fail("lsx_getopt called with invalid information");
state->curpos = NULL;
return -1;
}
else if (
state->argc <= state->ind ||
!state->argv[state->ind] ||
state->argv[state->ind][0] != '-' ||
state->argv[state->ind][1] == '\0')
{ /* return no more options */
state->curpos = NULL;
return -1;
}
else if (state->argv[state->ind][1] == '-' && state->argv[state->ind][2] == '\0')
{ /* skip "--", return no more options. */
state->curpos = NULL;
state->ind++;
return -1;
}
else
{ /* Look for the next option */
char const * current = state->argv[state->ind];
char const * param = current + 1;
if (state->curpos == NULL ||
state->curpos <= param ||
param + strlen(param) <= state->curpos)
{ /* Start parsing a new parameter - check for a long option */
state->curpos = NULL;
if (state->longopts &&
(param[0] == '-' || (state->flags & lsx_getopt_flag_longonly)))
{
size_t nameLen;
int doubleDash = param[0] == '-';
if (doubleDash)
{
param++;
}
for (nameLen = 0; param[nameLen] && param[nameLen] != '='; nameLen++)
{}
/* For single-dash, you have to specify at least two letters in the name. */
if (doubleDash || nameLen >= 2)
{
lsx_option_t const * pCur;
lsx_option_t const * pMatch = NULL;
int matches = 0;
for (pCur = state->longopts; pCur->name; pCur++)
{
if (0 == strncmp(pCur->name, param, nameLen))
{ /* Prefix match. */
matches++;
pMatch = pCur;
if (nameLen == strlen(pCur->name))
{ /* Exact match - no ambiguity, stop search. */
matches = 1;
break;
}
}
}
if (matches == 1)
{ /* Matched. */
state->ind++;
if (param[nameLen])
{ /* --name=value */
if (pMatch->has_arg)
{ /* Required or optional arg - done. */
state->arg = param + nameLen + 1;
}
else
{ /* No arg expected. */
if (oerr)
{
lsx_warn("`%s' did not expect an argument from `%s'",
pMatch->name,
current);
}
return '?';
}
}
else if (pMatch->has_arg == lsx_option_arg_required)
{ /* Arg required. */
state->arg = state->argv[state->ind];
state->ind++;
if (state->ind > state->argc)
{
if (oerr)
{
lsx_warn("`%s' requires an argument from `%s'",
pMatch->name,
current);
}
return state->shortopts[0] == ':' ? ':' : '?'; /* Missing required value. */
}
}
state->lngind = pMatch - state->longopts;
if (pMatch->flag)
{
*pMatch->flag = pMatch->val;
return 0;
}
else
{
return pMatch->val;
}
}
else if (matches == 0 && doubleDash)
{ /* No match */
if (oerr)
{
lsx_warn("parameter not recognized from `%s'", current);
}
state->ind++;
return '?';
}
else if (matches > 1)
{ /* Ambiguous. */
if (oerr)
{
lsx_warn("parameter `%s' is ambiguous:", current);
for (pCur = state->longopts; pCur->name; pCur++)
{
if (0 == strncmp(pCur->name, param, nameLen))
{
lsx_warn("parameter `%s' could be `--%s'", current, pCur->name);
}
}
}
state->ind++;
return '?';
}
}
}
state->curpos = param;
}
state->opt = state->curpos[0];
if (state->opt == ':')
{ /* ':' is never a valid short option character */
if (oerr)
{
lsx_warn("option `%c' not recognized", state->opt);
}
state->curpos++;
CheckCurPosEnd(state);
return '?'; /* unrecognized option */
}
else
{ /* Short option needs to be matched from option list */
char const * pShortopt = strchr(state->shortopts, state->opt);
state->curpos++;
if (!pShortopt)
{ /* unrecognized option */
if (oerr)
{
lsx_warn("option `%c' not recognized", state->opt);
}
CheckCurPosEnd(state);
return '?';
}
else if (pShortopt[1] == ':' && state->curpos[0])
{ /* Return the rest of the parameter as the option's value */
state->arg = state->curpos;
state->curpos = NULL;
state->ind++;
return state->opt;
}
else if (pShortopt[1] == ':' && pShortopt[2] != ':')
{ /* Option requires a value */
state->curpos = NULL;
state->ind++;
state->arg = state->argv[state->ind];
state->ind++;
if (state->ind <= state->argc)
{ /* A value was present, so we're good. */
return state->opt;
}
else
{ /* Missing required value. */
if (oerr)
{
lsx_warn("option `%c' requires an argument",
state->opt);
}
return state->shortopts[0] == ':' ? ':' : '?';
}
}
else
{ /* Option without a value. */
CheckCurPosEnd(state);
return state->opt;
}
}
}
}
#ifdef TEST_GETOPT
#include <stdio.h>
int main(int argc, char const * argv[])
{
static int help = 0;
static lsx_option_t longopts[] =
{
{"a11", 0, 0, 101},
{"a12", 0, 0, 102},
{"a122", 0, 0, 103},
{"rarg", 1, 0, 104},
{"oarg", 2, 0, 105},
{"help", 0, &help, 106},
{0}
};
int ch;
lsx_getopt_t state;
lsx_getopt_init(argc, argv, "abc:d:v::0123456789", longopts, sox_true, 1, &state);
while (-1 != (ch = lsx_getopt(&state)))
{
printf(
"H=%d ch=%d, ind=%d opt=%d lng=%d arg=%s\n",
help,
ch,
state.ind,
state.opt,
state.lngind,
state.arg ? state.arg : "NULL");
}
return 0;
}
#endif /* TEST_GETOPT */

247
app/src/main/cpp/sox/gsm.c
Unescape View File

@ -0,0 +1,247 @@
/* Copyright 1991, 1992, 1993 Guido van Rossum And Sundry Contributors.
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Guido van Rossum And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
/*
* GSM 06.10 courtesy Communications and Operating Systems Research Group,
* Technische Universitaet Berlin
*
* Source code and more information on this format can be obtained from
* http://www.quut.com/gsm/
*
* Written 26 Jan 1995 by Andrew Pam
* Portions Copyright (c) 1995 Serious Cybernetics
*
* July 19, 1998 - Chris Bagwell (cbagwell@sprynet.com)
* Added GSM support to SOX from patches floating around with the help
* of Dima Barsky (ess2db@ee.surrey.ac.uk).
*
* Nov. 26, 1999 - Stan Brooks (stabro@megsinet.com)
* Rewritten to support multiple channels
*/
#include "sox_i.h"
#ifdef HAVE_GSM_GSM_H
#include <gsm/gsm.h>
#else
#include <gsm.h>
#endif
#include <errno.h>
#define MAXCHANS 16
/* sizeof(gsm_frame) */
#define FRAMESIZE (size_t)33
/* samples per gsm_frame */
#define BLOCKSIZE 160
/* Private data */
typedef struct {
unsigned channels;
gsm_signal *samples;
gsm_signal *samplePtr;
gsm_signal *sampleTop;
gsm_byte *frames;
gsm handle[MAXCHANS];
} priv_t;
static int gsmstart_rw(sox_format_t * ft, int w)
{
priv_t *p = (priv_t *) ft->priv;
unsigned ch;
ft->encoding.encoding = SOX_ENCODING_GSM;
if (!ft->signal.rate)
ft->signal.rate = 8000;
if (ft->signal.channels == 0)
ft->signal.channels = 1;
p->channels = ft->signal.channels;
if (p->channels > MAXCHANS || p->channels <= 0)
{
lsx_fail_errno(ft,SOX_EFMT,"gsm: channels(%d) must be in 1-16", ft->signal.channels);
return(SOX_EOF);
}
for (ch=0; ch<p->channels; ch++) {
p->handle[ch] = gsm_create();
if (!p->handle[ch])
{
lsx_fail_errno(ft,errno,"unable to create GSM stream");
return (SOX_EOF);
}
}
p->frames = lsx_malloc(p->channels*FRAMESIZE);
p->samples = lsx_malloc(BLOCKSIZE * (p->channels+1) * sizeof(gsm_signal));
p->sampleTop = p->samples + BLOCKSIZE*p->channels;
p->samplePtr = (w)? p->samples : p->sampleTop;
return (SOX_SUCCESS);
}
static int sox_gsmstartread(sox_format_t * ft)
{
return gsmstart_rw(ft,0);
}
static int sox_gsmstartwrite(sox_format_t * ft)
{
return gsmstart_rw(ft,1);
}
/*
* Read up to len samples from file.
* Convert to signed longs.
* Place in buf[].
* Return number of samples read.
*/
static size_t sox_gsmread(sox_format_t * ft, sox_sample_t *buf, size_t samp)
{
size_t done = 0, r;
int ch, chans;
gsm_signal *gbuff;
priv_t *p = (priv_t *) ft->priv;
chans = p->channels;
while (done < samp)
{
while (p->samplePtr < p->sampleTop && done < samp)
buf[done++] =
SOX_SIGNED_16BIT_TO_SAMPLE(*(p->samplePtr)++,);
if (done>=samp) break;
r = lsx_readbuf(ft, p->frames, p->channels * FRAMESIZE);
if (r != p->channels * FRAMESIZE)
break;
p->samplePtr = p->samples;
for (ch=0; ch<chans; ch++) {
int i;
gsm_signal *gsp;
gbuff = p->sampleTop;
if (gsm_decode(p->handle[ch], p->frames + ch*FRAMESIZE, gbuff) < 0)
{
lsx_fail_errno(ft,errno,"error during GSM decode");
return (0);
}
gsp = p->samples + ch;
for (i=0; i<BLOCKSIZE; i++) {
*gsp = *gbuff++;
gsp += chans;
}
}
}
return done;
}
static int gsmflush(sox_format_t * ft)
{
int r, ch, chans;
gsm_signal *gbuff;
priv_t *p = (priv_t *) ft->priv;
chans = p->channels;
/* zero-fill samples as needed */
while (p->samplePtr < p->sampleTop)
*(p->samplePtr)++ = 0;
gbuff = p->sampleTop;
for (ch=0; ch<chans; ch++) {
int i;
gsm_signal *gsp;
gsp = p->samples + ch;
for (i=0; i<BLOCKSIZE; i++) {
gbuff[i] = *gsp;
gsp += chans;
}
gsm_encode(p->handle[ch], gbuff, p->frames);
r = lsx_writebuf(ft, p->frames, FRAMESIZE);
if (r != FRAMESIZE)
{
lsx_fail_errno(ft,errno,"write error");
return(SOX_EOF);
}
}
p->samplePtr = p->samples;
return (SOX_SUCCESS);
}
static size_t sox_gsmwrite(sox_format_t * ft, const sox_sample_t *buf, size_t samp)
{
size_t done = 0;
priv_t *p = (priv_t *) ft->priv;
while (done < samp)
{
SOX_SAMPLE_LOCALS;
while ((p->samplePtr < p->sampleTop) && (done < samp))
*(p->samplePtr)++ =
SOX_SAMPLE_TO_SIGNED_16BIT(buf[done++], ft->clips);
if (p->samplePtr == p->sampleTop)
{
if(gsmflush(ft))
{
return 0;
}
}
}
return done;
}
static int sox_gsmstopread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
unsigned ch;
for (ch=0; ch<p->channels; ch++)
gsm_destroy(p->handle[ch]);
free(p->samples);
free(p->frames);
return (SOX_SUCCESS);
}
static int sox_gsmstopwrite(sox_format_t * ft)
{
int rc;
priv_t *p = (priv_t *) ft->priv;
if (p->samplePtr > p->samples)
{
rc = gsmflush(ft);
if (rc)
return rc;
}
return sox_gsmstopread(ft); /* destroy handles and free buffers */
}
LSX_FORMAT_HANDLER(gsm)
{
static char const * const names[] = {"gsm", NULL};
static sox_rate_t const write_rates[] = {8000, 0};
static unsigned const write_encodings[] = {SOX_ENCODING_GSM, 0, 0};
static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,
"GSM 06.10 (full-rate) lossy speech compression", names, 0,
sox_gsmstartread, sox_gsmread, sox_gsmstopread,
sox_gsmstartwrite, sox_gsmwrite, sox_gsmstopwrite,
NULL, write_encodings, write_rates, sizeof(priv_t)
};
return &handler;
}

207
app/src/main/cpp/sox/gsrt.c
Unescape View File

@ -0,0 +1,207 @@
/* libSoX file format: Grandstream ring tone (c) 2009 robs@users.sourceforge.net
*
* See https://web.archive.org/web/20101128121923/http://grandstream.com/ringtone.html
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include <time.h>
#define VERSION_ 0x1000000
#define MAX_FILE_SIZE 0x10000
#define HEADER_SIZE (size_t)512
#define PADDING_SIZE (size_t)478
static char const id[16] = "ring.bin";
typedef struct {
char const * string;
int ft_encoding;
unsigned bits_per_sample;
sox_encoding_t sox_encoding;
} table_t;
static table_t const table[] = {
{NULL, 0, 8, SOX_ENCODING_ULAW},
{"G726", 2, 0, SOX_ENCODING_UNKNOWN},
{NULL, 3, 0, SOX_ENCODING_GSM},
{NULL, 4, 0, SOX_ENCODING_G723},
{NULL, 8, 8, SOX_ENCODING_ALAW},
{"G722", 9, 0, SOX_ENCODING_UNKNOWN},
{"G728", 15, 2, SOX_ENCODING_UNKNOWN},
{"iLBC", 98, 0, SOX_ENCODING_UNKNOWN},
};
static int ft_enc(unsigned bits_per_sample, sox_encoding_t encoding)
{
size_t i;
for (i = 0; i < array_length(table); ++i) {
table_t const * t = &table[i];
if (t->sox_encoding == encoding && t->bits_per_sample == bits_per_sample)
return t->ft_encoding;
}
return -1; /* Should never get here. */
}
static sox_encoding_t sox_enc(int ft_encoding, unsigned * bits_per_sample)
{
size_t i;
for (i = 0; i < array_length(table); ++i) {
table_t const * t = &table[i];
if (t->ft_encoding == ft_encoding) {
*bits_per_sample = t->bits_per_sample;
if (t->sox_encoding == SOX_ENCODING_UNKNOWN)
lsx_report("unsupported encoding: %s", t->string);
return t->sox_encoding;
}
}
*bits_per_sample = 0;
return SOX_ENCODING_UNKNOWN;
}
static int start_read(sox_format_t * ft)
{
off_t num_samples;
char read_id[array_length(id)];
uint32_t file_size;
int16_t ft_encoding;
sox_encoding_t encoding;
unsigned bits_per_sample;
lsx_readdw(ft, &file_size);
num_samples = file_size? file_size * 2 - HEADER_SIZE : SOX_UNSPEC;
if (file_size >= 2 && ft->seekable) {
int i, checksum = (file_size >> 16) + file_size;
for (i = file_size - 2; i; --i) {
int16_t int16;
lsx_readsw(ft, &int16);
checksum += int16;
}
if (lsx_seeki(ft, (off_t)sizeof(file_size), SEEK_SET) != 0)
return SOX_EOF;
if (checksum & 0xffff)
lsx_warn("invalid checksum in input file %s", ft->filename);
}
lsx_skipbytes(ft, (size_t)(2 + 4 + 6)); /* Checksum, version, time stamp. */
lsx_readchars(ft, read_id, sizeof(read_id));
if (memcmp(read_id, id, strlen(id))) {
lsx_fail_errno(ft, SOX_EHDR, "gsrt: invalid file name in header");
return SOX_EOF;
}
lsx_readsw(ft, &ft_encoding);
encoding = sox_enc(ft_encoding, &bits_per_sample);
if (encoding != SOX_ENCODING_ALAW &&
encoding != SOX_ENCODING_ULAW)
ft->handler.read = NULL;
lsx_skipbytes(ft, PADDING_SIZE);
return lsx_check_read_params(ft, 1, 8000., encoding,
bits_per_sample, (uint64_t)num_samples, sox_true);
}
static int start_write(sox_format_t * ft)
{
int i, encoding = ft_enc(ft->encoding.bits_per_sample, ft->encoding.encoding);
time_t now = sox_globals.repeatable? 0 : time(NULL);
struct tm const * t = sox_globals.repeatable? gmtime(&now) : localtime(&now);
int checksum = (VERSION_ >> 16) + VERSION_;
checksum += t->tm_year + 1900;
checksum += ((t->tm_mon + 1) << 8) + t->tm_mday;
checksum += (t->tm_hour << 8) + t->tm_min;
for (i = sizeof(id) - 2; i >= 0; i -= 2)
checksum += (id[i] << 8) + id[i + 1];
checksum += encoding;
return lsx_writedw(ft, 0)
|| lsx_writesw(ft, -checksum)
|| lsx_writedw(ft, VERSION_)
|| lsx_writesw(ft, t->tm_year + 1900)
|| lsx_writesb(ft, t->tm_mon + 1)
|| lsx_writesb(ft, t->tm_mday)
|| lsx_writesb(ft, t->tm_hour)
|| lsx_writesb(ft, t->tm_min)
|| lsx_writechars(ft, id, sizeof(id))
|| lsx_writesw(ft, encoding)
|| lsx_padbytes(ft, PADDING_SIZE) ? SOX_EOF : SOX_SUCCESS;
}
static size_t write_samples(
sox_format_t * ft, sox_sample_t const * buf, size_t nsamp)
{
size_t n = min(nsamp, MAX_FILE_SIZE - (size_t)ft->tell_off);
if (n != nsamp)
lsx_warn("audio truncated");
return lsx_rawwrite(ft, buf, n);
}
static int stop_write(sox_format_t * ft)
{
long num_samples = ft->tell_off - HEADER_SIZE;
if (num_samples & 1) {
sox_sample_t pad = 0;
lsx_rawwrite(ft, &pad, 1);
}
if (ft->seekable) {
unsigned i, file_size = ft->tell_off >> 1;
int16_t int16;
int checksum;
if (!lsx_seeki(ft, (off_t)sizeof(uint32_t), SEEK_SET)) {
lsx_readsw(ft, &int16);
checksum = (file_size >> 16) + file_size - int16;
if (!lsx_seeki(ft, (off_t)HEADER_SIZE, SEEK_SET)) {
for (i = (num_samples + 1) >> 1; i; --i) {
lsx_readsw(ft, &int16);
checksum += int16;
}
if (!lsx_seeki(ft, (off_t)0, SEEK_SET)) {
lsx_writedw(ft, file_size);
lsx_writesw(ft, -checksum);
return SOX_SUCCESS;
}
}
}
}
lsx_warn("can't seek in output file `%s'; "
"length in file header will be unspecified", ft->filename);
return SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(gsrt)
{
static char const *const names[] = { "gsrt", NULL };
static sox_rate_t const write_rates[] = { 8000, 0 };
static unsigned const write_encodings[] = {
SOX_ENCODING_ALAW, 8, 0,
SOX_ENCODING_ULAW, 8, 0,
0
};
static sox_format_handler_t const handler = {
SOX_LIB_VERSION_CODE, "Grandstream ring tone",
names, SOX_FILE_BIG_END | SOX_FILE_MONO,
start_read, lsx_rawread, NULL,
start_write, write_samples, stop_write,
lsx_rawseek, write_encodings, write_rates, 0
};
return &handler;
}

502
app/src/main/cpp/sox/hcom.c
Unescape View File

@ -0,0 +1,502 @@
/* libSoX Macintosh HCOM format.
* These are really FSSD type files with Huffman compression,
* in MacBinary format.
* TODO: make the MacBinary format optional (so that .data files
* are also acceptable). (How to do this on output?)
*
* September 25, 1991
* Copyright 1991 Guido van Rossum And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Guido van Rossum And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* April 28, 1998 - Chris Bagwell (cbagwell@sprynet.com)
*
* Rearranged some functions so that they are declared before they are
* used, clearing up some compiler warnings. Because these functions
* passed floats, it helped some dumb compilers pass stuff on the
* stack correctly.
*
*/
#include "sox_i.h"
#include <assert.h>
#include <string.h>
#include <stdlib.h>
#include <errno.h>
/* FIXME: eliminate these 2 functions */
static void put32_be(unsigned char **p, int32_t val)
{
*(*p)++ = (val >> 24) & 0xff;
*(*p)++ = (val >> 16) & 0xff;
*(*p)++ = (val >> 8) & 0xff;
*(*p)++ = val & 0xff;
}
static void put16_be(unsigned char **p, int val)
{
*(*p)++ = (val >> 8) & 0xff;
*(*p)++ = val & 0xff;
}
/* Dictionary entry for Huffman (de)compression */
typedef struct {
long frequ;
short dict_leftson;
short dict_rightson;
} dictent;
typedef struct {
/* Static data from the header */
dictent *dictionary;
int32_t checksum;
int deltacompression;
/* Engine state */
long huffcount;
long cksum;
int dictentry;
int nrbits;
uint32_t current;
short sample;
/* Dictionary */
dictent *de;
int32_t new_checksum;
int nbits;
int32_t curword;
/* Private data used by writer */
unsigned char *data; /* Buffer allocated with lsx_malloc */
size_t size; /* Size of allocated buffer */
size_t pos; /* Where next byte goes */
} priv_t;
static int dictvalid(int n, int size, int left, int right)
{
if (n > 0 && left < 0)
return 1;
return (unsigned)left < size && (unsigned)right < size;
}
static int startread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
int i;
char buf[5];
uint32_t datasize, rsrcsize;
uint32_t huffcount, checksum, compresstype, divisor;
unsigned short dictsize;
int rc;
/* Skip first 65 bytes of header */
rc = lsx_skipbytes(ft, (size_t) 65);
if (rc)
return rc;
/* Check the file type (bytes 65-68) */
if (lsx_reads(ft, buf, (size_t)4) == SOX_EOF || strncmp(buf, "FSSD", (size_t)4) != 0)
{
lsx_fail_errno(ft,SOX_EHDR,"Mac header type is not FSSD");
return (SOX_EOF);
}
/* Skip to byte 83 */
rc = lsx_skipbytes(ft, (size_t) 83-69);
if (rc)
return rc;
/* Get essential numbers from the header */
lsx_readdw(ft, &datasize); /* bytes 83-86 */
lsx_readdw(ft, &rsrcsize); /* bytes 87-90 */
/* Skip the rest of the header (total 128 bytes) */
rc = lsx_skipbytes(ft, (size_t) 128-91);
if (rc != 0)
return rc;
/* The data fork must contain a "HCOM" header */
if (lsx_reads(ft, buf, (size_t)4) == SOX_EOF || strncmp(buf, "HCOM", (size_t)4) != 0)
{
lsx_fail_errno(ft,SOX_EHDR,"Mac data fork is not HCOM");
return (SOX_EOF);
}
/* Then follow various parameters */
lsx_readdw(ft, &huffcount);
lsx_readdw(ft, &checksum);
lsx_readdw(ft, &compresstype);
if (compresstype > 1)
{
lsx_fail_errno(ft,SOX_EHDR,"Bad compression type in HCOM header");
return (SOX_EOF);
}
lsx_readdw(ft, &divisor);
if (divisor == 0 || divisor > 4)
{
lsx_fail_errno(ft,SOX_EHDR,"Bad sampling rate divisor in HCOM header");
return (SOX_EOF);
}
lsx_readw(ft, &dictsize);
/* Translate to sox parameters */
ft->encoding.encoding = SOX_ENCODING_HCOM;
ft->encoding.bits_per_sample = 8;
ft->signal.rate = 22050 / divisor;
ft->signal.channels = 1;
ft->signal.length = huffcount;
/* Allocate memory for the dictionary */
p->dictionary = lsx_malloc(511 * sizeof(dictent));
/* Read dictionary */
for(i = 0; i < dictsize; i++) {
lsx_readsw(ft, &(p->dictionary[i].dict_leftson));
lsx_readsw(ft, &(p->dictionary[i].dict_rightson));
lsx_debug("%d %d",
p->dictionary[i].dict_leftson,
p->dictionary[i].dict_rightson);
if (!dictvalid(i, dictsize, p->dictionary[i].dict_leftson,
p->dictionary[i].dict_rightson)) {
lsx_fail_errno(ft, SOX_EHDR, "Invalid dictionary");
return SOX_EOF;
}
}
rc = lsx_skipbytes(ft, (size_t) 1); /* skip pad byte */
if (rc)
return rc;
/* Initialized the decompression engine */
p->checksum = checksum;
p->deltacompression = compresstype;
if (!p->deltacompression)
lsx_debug("HCOM data using value compression");
p->huffcount = huffcount;
p->cksum = 0;
p->dictentry = 0;
p->nrbits = -1; /* Special case to get first byte */
return (SOX_SUCCESS);
}
static size_t read_samples(sox_format_t * ft, sox_sample_t *buf, size_t len)
{
register priv_t *p = (priv_t *) ft->priv;
int done = 0;
unsigned char sample_rate;
if (p->nrbits < 0) {
/* The first byte is special */
if (p->huffcount == 0)
return 0; /* Don't know if this can happen... */
if (lsx_readb(ft, &sample_rate) == SOX_EOF)
{
return (0);
}
p->sample = sample_rate;
*buf++ = SOX_UNSIGNED_8BIT_TO_SAMPLE(p->sample,);
p->huffcount--;
p->nrbits = 0;
done++;
len--;
if (len == 0)
return done;
}
while (p->huffcount > 0) {
if(p->nrbits == 0) {
lsx_readdw(ft, &(p->current));
if (lsx_eof(ft))
{
lsx_fail_errno(ft,SOX_EOF,"unexpected EOF in HCOM data");
return (0);
}
p->cksum += p->current;
p->nrbits = 32;
}
if(p->current & 0x80000000) {
p->dictentry =
p->dictionary[p->dictentry].dict_rightson;
} else {
p->dictentry =
p->dictionary[p->dictentry].dict_leftson;
}
p->current = p->current << 1;
p->nrbits--;
if(p->dictionary[p->dictentry].dict_leftson < 0) {
short datum;
datum = p->dictionary[p->dictentry].dict_rightson;
if (!p->deltacompression)
p->sample = 0;
p->sample = (p->sample + datum) & 0xff;
p->huffcount--;
*buf++ = SOX_UNSIGNED_8BIT_TO_SAMPLE(p->sample,);
p->dictentry = 0;
done++;
len--;
if (len == 0)
break;
}
}
return done;
}
static int stopread(sox_format_t * ft)
{
register priv_t *p = (priv_t *) ft->priv;
if (p->huffcount != 0)
{
lsx_fail_errno(ft,SOX_EFMT,"not all HCOM data read");
return (SOX_EOF);
}
if(p->cksum != p->checksum)
{
lsx_fail_errno(ft,SOX_EFMT,"checksum error in HCOM data");
return (SOX_EOF);
}
free(p->dictionary);
p->dictionary = NULL;
return (SOX_SUCCESS);
}
#define BUFINCR (10*BUFSIZ)
static int startwrite(sox_format_t * ft)
{
priv_t * p = (priv_t *) ft->priv;
p->size = BUFINCR;
p->pos = 0;
p->data = lsx_malloc(p->size);
return SOX_SUCCESS;
}
static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t len)
{
priv_t *p = (priv_t *) ft->priv;
sox_sample_t datum;
size_t i;
if (len == 0)
return 0;
if (p->pos == INT32_MAX)
return SOX_EOF;
if (p->pos + len > INT32_MAX) {
lsx_warn("maximum file size exceeded");
len = INT32_MAX - p->pos;
}
if (p->pos + len > p->size) {
p->size = ((p->pos + len) / BUFINCR + 1) * BUFINCR;
p->data = lsx_realloc(p->data, p->size);
}
for (i = 0; i < len; i++) {
SOX_SAMPLE_LOCALS;
datum = *buf++;
p->data[p->pos++] = SOX_SAMPLE_TO_UNSIGNED_8BIT(datum, ft->clips);
}
return len;
}
static void makecodes(int e, int c, int s, int b, dictent newdict[511], long codes[256], long codesize[256])
{
assert(b); /* Prevent stack overflow */
if (newdict[e].dict_leftson < 0) {
codes[newdict[e].dict_rightson] = c;
codesize[newdict[e].dict_rightson] = s;
} else {
makecodes(newdict[e].dict_leftson, c, s + 1, b << 1, newdict, codes, codesize);
makecodes(newdict[e].dict_rightson, c + b, s + 1, b << 1, newdict, codes, codesize);
}
}
static void putcode(sox_format_t * ft, long codes[256], long codesize[256], unsigned c, unsigned char **df)
{
priv_t *p = (priv_t *) ft->priv;
long code, size;
int i;
code = codes[c];
size = codesize[c];
for(i = 0; i < size; i++) {
p->curword <<= 1;
if (code & 1)
p->curword += 1;
p->nbits++;
if (p->nbits == 32) {
put32_be(df, p->curword);
p->new_checksum += p->curword;
p->nbits = 0;
p->curword = 0;
}
code >>= 1;
}
}
static void compress(sox_format_t * ft, unsigned char **df, int32_t *dl)
{
priv_t *p = (priv_t *) ft->priv;
int samplerate;
unsigned char *datafork = *df;
unsigned char *ddf, *dfp;
short dictsize;
int frequtable[256];
long codes[256], codesize[256];
dictent newdict[511];
int i, sample, j, k, d, l, frequcount;
int64_t csize;
sample = *datafork;
memset(frequtable, 0, sizeof(frequtable));
memset(codes, 0, sizeof(codes));
memset(codesize, 0, sizeof(codesize));
memset(newdict, 0, sizeof(newdict));
for (i = 1; i < *dl; i++) {
d = (datafork[i] - (sample & 0xff)) & 0xff; /* creates absolute entries LMS */
sample = datafork[i];
datafork[i] = d;
assert(d >= 0 && d <= 255); /* check our table is accessed correctly */
frequtable[d]++;
}
p->de = newdict;
for (i = 0; i < 256; i++)
if (frequtable[i] != 0) {
p->de->frequ = -frequtable[i];
p->de->dict_leftson = -1;
p->de->dict_rightson = i;
p->de++;
}
frequcount = p->de - newdict;
for (i = 0; i < frequcount; i++) {
for (j = i + 1; j < frequcount; j++) {
if (newdict[i].frequ > newdict[j].frequ) {
k = newdict[i].frequ;
newdict[i].frequ = newdict[j].frequ;
newdict[j].frequ = k;
k = newdict[i].dict_leftson;
newdict[i].dict_leftson = newdict[j].dict_leftson;
newdict[j].dict_leftson = k;
k = newdict[i].dict_rightson;
newdict[i].dict_rightson = newdict[j].dict_rightson;
newdict[j].dict_rightson = k;
}
}
}
while (frequcount > 1) {
j = frequcount - 1;
p->de->frequ = newdict[j - 1].frequ;
p->de->dict_leftson = newdict[j - 1].dict_leftson;
p->de->dict_rightson = newdict[j - 1].dict_rightson;
l = newdict[j - 1].frequ + newdict[j].frequ;
for (i = j - 2; i >= 0 && l < newdict[i].frequ; i--)
newdict[i + 1] = newdict[i];
i = i + 1;
newdict[i].frequ = l;
newdict[i].dict_leftson = j;
newdict[i].dict_rightson = p->de - newdict;
p->de++;
frequcount--;
}
dictsize = p->de - newdict;
makecodes(0, 0, 0, 1, newdict, codes, codesize);
csize = 0;
for (i = 0; i < 256; i++)
csize += frequtable[i] * codesize[i];
l = (((csize + 31) >> 5) << 2) + 24 + dictsize * 4;
lsx_debug(" Original size: %6d bytes", *dl);
lsx_debug("Compressed size: %6d bytes", l);
datafork = lsx_malloc((size_t)l);
ddf = datafork + 22;
for(i = 0; i < dictsize; i++) {
put16_be(&ddf, newdict[i].dict_leftson);
put16_be(&ddf, newdict[i].dict_rightson);
}
*ddf++ = 0;
*ddf++ = *(*df)++;
p->new_checksum = 0;
p->nbits = 0;
p->curword = 0;
for (i = 1; i < *dl; i++)
putcode(ft, codes, codesize, *(*df)++, &ddf);
if (p->nbits != 0) {
codes[0] = 0;
codesize[0] = 32 - p->nbits;
putcode(ft, codes, codesize, 0, &ddf);
}
memcpy(datafork, "HCOM", (size_t)4);
dfp = datafork + 4;
put32_be(&dfp, *dl);
put32_be(&dfp, p->new_checksum);
put32_be(&dfp, 1);
samplerate = 22050 / ft->signal.rate + .5;
put32_be(&dfp, samplerate);
put16_be(&dfp, dictsize);
*df = datafork; /* reassign passed pointer to new datafork */
*dl = l; /* and its compressed length */
}
/* End of hcom utility routines */
static int stopwrite(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
unsigned char *compressed_data = p->data;
int32_t compressed_len = p->pos;
int rc = SOX_SUCCESS;
/* Compress it all at once */
if (compressed_len) {
compress(ft, &compressed_data, &compressed_len);
free(p->data);
}
/* Write the header */
lsx_writebuf(ft, "\000\001A", (size_t) 3); /* Dummy file name "A" */
lsx_padbytes(ft, (size_t) 65-3);
lsx_writes(ft, "FSSD");
lsx_padbytes(ft, (size_t) 83-69);
lsx_writedw(ft, (unsigned) compressed_len); /* compressed_data size */
lsx_writedw(ft, 0); /* rsrc size */
lsx_padbytes(ft, (size_t) 128 - 91);
if (lsx_error(ft)) {
lsx_fail_errno(ft, errno, "write error in HCOM header");
rc = SOX_EOF;
} else if (lsx_writebuf(ft, compressed_data, compressed_len) != compressed_len) {
/* Write the compressed_data fork */
lsx_fail_errno(ft, errno, "can't write compressed HCOM data");
rc = SOX_EOF;
}
free(compressed_data);
if (rc == SOX_SUCCESS)
/* Pad the compressed_data fork to a multiple of 128 bytes */
lsx_padbytes(ft, 128u - (compressed_len % 128));
return rc;
}
LSX_FORMAT_HANDLER(hcom)
{
static char const * const names[] = {"hcom", NULL};
static sox_rate_t const write_rates[] = {22050,22050/2,22050/3,22050/4.,0};
static unsigned const write_encodings[] = {
SOX_ENCODING_HCOM, 8, 0, 0};
static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,
"Mac FSSD files with Huffman compression",
names, SOX_FILE_BIG_END|SOX_FILE_MONO,
startread, read_samples, stopread,
startwrite, write_samples, stopwrite,
NULL, write_encodings, write_rates, sizeof(priv_t)
};
return &handler;
}

102
app/src/main/cpp/sox/hilbert.c
Unescape View File

@ -0,0 +1,102 @@
/* libSoX effect: Hilbert transform filter
*
* First version of this effect written 11/2011 by Ulrich Klauer, using maths
* from "Understanding digital signal processing" by Richard G. Lyons.
*
* Copyright 2011 Chris Bagwell and SoX Contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "dft_filter.h"
typedef struct {
dft_filter_priv_t base;
double *h;
int taps;
} priv_t;
static int getopts(sox_effect_t *effp, int argc, char **argv)
{
lsx_getopt_t optstate;
int c;
priv_t *p = (priv_t*)effp->priv;
dft_filter_priv_t *b = &p->base;
b->filter_ptr = &b->filter;
lsx_getopt_init(argc, argv, "+n:", NULL, lsx_getopt_flag_none, 1, &optstate);
while ((c = lsx_getopt(&optstate)) != -1) switch (c) {
GETOPT_NUMERIC(optstate, 'n', taps, 3, 32767)
default: lsx_fail("invalid option `-%c'", optstate.opt); return lsx_usage(effp);
}
if (p->taps && p->taps%2 == 0) {
lsx_fail("only filters with an odd number of taps are supported");
return SOX_EOF;
}
return optstate.ind != argc ? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t *effp)
{
priv_t *p = (priv_t*)effp->priv;
dft_filter_t *f = p->base.filter_ptr;
if (!f->num_taps) {
int i;
if (!p->taps) {
p->taps = effp->in_signal.rate/76.5 + 2;
p->taps += 1 - (p->taps%2);
/* results in a cutoff frequency of about 75 Hz with a Blackman window */
lsx_debug("choosing number of taps = %d (override with -n)", p->taps);
}
lsx_valloc(p->h, p->taps);
for (i = 0; i < p->taps; i++) {
int k = -(p->taps/2) + i;
if (k%2 == 0) {
p->h[i] = 0.0;
} else {
double pk = M_PI * k;
p->h[i] = (1 - cos(pk))/pk;
}
}
lsx_apply_blackman(p->h, p->taps, .16);
if (effp->global_info->plot != sox_plot_off) {
char title[100];
sprintf(title, "SoX effect: hilbert (%d taps)", p->taps);
lsx_plot_fir(p->h, p->taps, effp->in_signal.rate,
effp->global_info->plot, title, -20., 5.);
free(p->h);
return SOX_EOF;
}
lsx_set_dft_filter(f, p->h, p->taps, p->taps/2);
}
return lsx_dft_filter_effect_fn()->start(effp);
}
sox_effect_handler_t const *lsx_hilbert_effect_fn(void)
{
static sox_effect_handler_t handler;
handler = *lsx_dft_filter_effect_fn();
handler.name = "hilbert";
handler.usage = "[-n taps]";
handler.getopts = getopts;
handler.start = start;
handler.priv_size = sizeof(priv_t);
return &handler;
}

80
app/src/main/cpp/sox/htk.c
Unescape View File

@ -0,0 +1,80 @@
/* libSoX file format: HTK (c) 2008 robs@users.sourceforge.net
*
* See http://labrosa.ee.columbia.edu/doc/HTKBook21/HTKBook.html
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
typedef enum {
Waveform, Lpc, Lprefc, Lpcepstra, Lpdelcep, Irefc,
Mfcc, Fbank, Melspec, User, Discrete, Unknown} kind_t;
static char const * const str[] = {
"Sampled waveform", "Linear prediction filter", "Linear prediction",
"LPC cepstral", "LPC cepstra plus delta", "LPC reflection coef in",
"Mel-frequency cepstral", "Log mel-filter bank", "Linear mel-filter bank",
"User defined sample", "Vector quantised data", "Unknown"};
static int start_read(sox_format_t * ft)
{
uint32_t period_100ns, num_samples;
uint16_t bytes_per_sample, parmKind;
if (lsx_readdw(ft, &num_samples ) ||
lsx_readdw(ft, &period_100ns ) ||
lsx_readw (ft, &bytes_per_sample) ||
lsx_readw (ft, &parmKind )) return SOX_EOF;
if (parmKind != Waveform) {
int n = min(parmKind & 077, Unknown);
lsx_fail_errno(ft, SOX_EFMT, "unsupported HTK type `%s' (0%o)", str[n], parmKind);
return SOX_EOF;
}
return lsx_check_read_params(ft, 1, 1e7 / period_100ns, SOX_ENCODING_SIGN2,
(unsigned)bytes_per_sample << 3, (uint64_t)num_samples, sox_true);
}
static int write_header(sox_format_t * ft)
{
double period_100ns = 1e7 / ft->signal.rate;
uint64_t len = ft->olength? ft->olength:ft->signal.length;
if (len > UINT_MAX)
{
lsx_warn("length greater than 32 bits - cannot fit actual length in header");
len = UINT_MAX;
}
if (!ft->olength && floor(period_100ns) != period_100ns)
lsx_warn("rounding sample period %f (x 100ns) to nearest integer", period_100ns);
return lsx_writedw(ft, (unsigned)len)
|| lsx_writedw(ft, (unsigned)(period_100ns + .5))
|| lsx_writew(ft, ft->encoding.bits_per_sample >> 3)
|| lsx_writew(ft, Waveform) ? SOX_EOF : SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(htk)
{
static char const * const names[] = {"htk", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_SIGN2, 16, 0, 0};
static sox_format_handler_t handler = {
SOX_LIB_VERSION_CODE,
"PCM format used for Hidden Markov Model speech processing",
names, SOX_FILE_BIG_END | SOX_FILE_MONO | SOX_FILE_REWIND,
start_read, lsx_rawread, NULL,
write_header, lsx_rawwrite, NULL,
lsx_rawseek, write_encodings, NULL, 0
};
return &handler;
}

227
app/src/main/cpp/sox/id3.c
Unescape View File

@ -0,0 +1,227 @@
/* libSoX MP3 utilities Copyright (c) 2007-9 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "id3.h"
#ifdef HAVE_ID3TAG
#include <id3tag.h>
static char const * id3tagmap[][2] =
{
{"TIT2", "Title"},
{"TPE1", "Artist"},
{"TALB", "Album"},
{"TCOM", "Composer"},
{"TRCK", "Tracknumber"},
{"TDRC", "Year"},
{"TCON", "Genre"},
{"COMM", "Comment"},
{"TPOS", "Discnumber"},
{NULL, NULL}
};
static id3_utf8_t * utf8_id3tag_findframe(
struct id3_tag * tag, const char * const frameid, unsigned index)
{
struct id3_frame const * frame = id3_tag_findframe(tag, frameid, index);
if (frame) {
unsigned nfields = frame->nfields;
while (nfields--) {
union id3_field const *field = id3_frame_field(frame, nfields);
int ftype = id3_field_type(field);
const id3_ucs4_t *ucs4 = NULL;
unsigned nstrings;
switch (ftype) {
case ID3_FIELD_TYPE_STRING:
ucs4 = id3_field_getstring(field);
break;
case ID3_FIELD_TYPE_STRINGFULL:
ucs4 = id3_field_getfullstring(field);
break;
case ID3_FIELD_TYPE_STRINGLIST:
nstrings = id3_field_getnstrings(field);
while (nstrings--) {
ucs4 = id3_field_getstrings(field, nstrings);
if (ucs4)
break;
}
break;
}
if (ucs4)
return id3_ucs4_utf8duplicate(ucs4); /* Must call free() on this */
}
}
return NULL;
}
struct tag_info_node
{
struct tag_info_node * next;
off_t start;
off_t end;
};
struct tag_info {
sox_format_t * ft;
struct tag_info_node * head;
struct id3_tag * tag;
};
static int add_tag(struct tag_info * info)
{
struct tag_info_node * current;
off_t start, end;
id3_byte_t query[ID3_TAG_QUERYSIZE];
id3_byte_t * buffer;
long size;
int result = 0;
/* Ensure we're at the start of a valid tag and get its size. */
if (ID3_TAG_QUERYSIZE != lsx_readbuf(info->ft, query, ID3_TAG_QUERYSIZE) ||
!(size = id3_tag_query(query, ID3_TAG_QUERYSIZE))) {
return 0;
}
if (size < 0) {
if (0 != lsx_seeki(info->ft, size, SEEK_CUR) ||
ID3_TAG_QUERYSIZE != lsx_readbuf(info->ft, query, ID3_TAG_QUERYSIZE) ||
(size = id3_tag_query(query, ID3_TAG_QUERYSIZE)) <= 0) {
return 0;
}
}
/* Don't read a tag more than once. */
start = lsx_tell(info->ft);
end = start + size;
for (current = info->head; current; current = current->next) {
if (start == current->start && end == current->end) {
return 1;
} else if (start < current->end && current->start < end) {
return 0;
}
}
buffer = lsx_malloc((size_t)size);
if (!buffer) {
return 0;
}
memcpy(buffer, query, ID3_TAG_QUERYSIZE);
if ((unsigned long)size - ID3_TAG_QUERYSIZE ==
lsx_readbuf(info->ft, buffer + ID3_TAG_QUERYSIZE, (size_t)size - ID3_TAG_QUERYSIZE)) {
struct id3_tag * tag = id3_tag_parse(buffer, (size_t)size);
if (tag) {
current = lsx_malloc(sizeof(struct tag_info_node));
if (current) {
current->next = info->head;
current->start = start;
current->end = end;
info->head = current;
if (info->tag && (info->tag->extendedflags & ID3_TAG_EXTENDEDFLAG_TAGISANUPDATE)) {
struct id3_frame * frame;
unsigned i;
for (i = 0; (frame = id3_tag_findframe(tag, NULL, i)); i++) {
id3_tag_attachframe(info->tag, frame);
}
id3_tag_delete(tag);
} else {
if (info->tag) {
id3_tag_delete(info->tag);
}
info->tag = tag;
}
}
}
}
free(buffer);
return result;
}
void lsx_id3_read_tag(sox_format_t * ft, sox_bool search)
{
struct tag_info info;
id3_utf8_t * utf8;
int i;
int has_id3v1 = 0;
info.ft = ft;
info.head = NULL;
info.tag = NULL;
/*
We look for:
ID3v1 at end (EOF - 128).
ID3v2 at start.
ID3v2 at end (but before ID3v1 from end if there was one).
*/
if (search) {
if (0 == lsx_seeki(ft, -128, SEEK_END)) {
has_id3v1 =
add_tag(&info) &&
1 == ID3_TAG_VERSION_MAJOR(id3_tag_version(info.tag));
}
if (0 == lsx_seeki(ft, 0, SEEK_SET)) {
add_tag(&info);
}
if (0 == lsx_seeki(ft, has_id3v1 ? -138 : -10, SEEK_END)) {
add_tag(&info);
}
} else {
add_tag(&info);
}
if (info.tag && info.tag->frames) {
for (i = 0; id3tagmap[i][0]; ++i) {
if ((utf8 = utf8_id3tag_findframe(info.tag, id3tagmap[i][0], 0))) {
char * comment = lsx_malloc(strlen(id3tagmap[i][1]) + 1 + strlen((char *)utf8) + 1);
sprintf(comment, "%s=%s", id3tagmap[i][1], utf8);
sox_append_comment(&ft->oob.comments, comment);
free(comment);
free(utf8);
}
}
if ((utf8 = utf8_id3tag_findframe(info.tag, "TLEN", 0))) {
unsigned long tlen = strtoul((char *)utf8, NULL, 10);
if (tlen > 0 && tlen < ULONG_MAX) {
ft->signal.length= tlen; /* In ms; convert to samples later */
lsx_debug("got exact duration from ID3 TLEN");
}
free(utf8);
}
}
while (info.head) {
struct tag_info_node * head = info.head;
info.head = head->next;
free(head);
}
if (info.tag) {
id3_tag_delete(info.tag);
}
}
#else
/* Stub for format modules */
void lsx_id3_read_tag(sox_format_t *ft, sox_bool search) { }
#endif

25
app/src/main/cpp/sox/id3.h
Unescape View File

@ -0,0 +1,25 @@
/* libSoX MP3 utilities Copyright (c) 2007-9 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SOX_ID3_H
#define SOX_ID3_H
#include "sox_i.h"
void lsx_id3_read_tag(sox_format_t *ft, sox_bool search);
#endif

33
app/src/main/cpp/sox/ima-fmt.c
Unescape View File

@ -0,0 +1,33 @@
/* libSoX format: raw IMA ADPCM (c) 2007-8 SoX contributors
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "adpcms.h"
#include "vox.h"
LSX_FORMAT_HANDLER(ima)
{
static char const * const names[] = {"ima", NULL};
static unsigned const write_encodings[] = {SOX_ENCODING_IMA_ADPCM, 4, 0, 0};
static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,
"Raw IMA ADPCM", names, SOX_FILE_MONO,
lsx_ima_start, lsx_vox_read, lsx_vox_stopread,
lsx_ima_start, lsx_vox_write, lsx_vox_stopwrite,
lsx_rawseek, write_encodings, NULL, sizeof(adpcm_io_t)
};
return &handler;
}

365
app/src/main/cpp/sox/ima_rw.c
Unescape View File

@ -0,0 +1,365 @@
/* libSoX ima_rw.c -- codex utilities for WAV_FORMAT_IMA_ADPCM
* Copyright (C) 1999 Stanley J. Brooks <stabro@megsinet.net>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
#include "ima_rw.h"
#include <sys/types.h>
#include <stdio.h>
#include <stdlib.h>
/*
*
* Lookup tables for IMA ADPCM format
*
*/
#define ISSTMAX 88
static const int imaStepSizeTable[ISSTMAX + 1] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34,
37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494,
544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552,
1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026,
4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623,
27086, 29794, 32767
};
#define imaStateAdjust(c) (((c)<4)? -1:(2*(c)-6))
/* +0 - +3, decrease step size */
/* +4 - +7, increase step size */
/* -0 - -3, decrease step size */
/* -4 - -7, increase step size */
static unsigned char imaStateAdjustTable[ISSTMAX+1][8];
void lsx_ima_init_table(void)
{
int i,j,k;
for (i=0; i<=ISSTMAX; i++) {
for (j=0; j<8; j++) {
k = i + imaStateAdjust(j);
if (k<0) k=0;
else if (k>ISSTMAX) k=ISSTMAX;
imaStateAdjustTable[i][j] = k;
}
}
}
static void ImaExpandS(
unsigned ch, /* channel number to decode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* obuff[n] will be output samples */
int n, /* samples to decode PER channel, REQUIRE n % 8 == 1 */
unsigned o_inc /* index difference between successive output samples */
)
{
const unsigned char *ip;
int i_inc;
SAMPL *op;
int i, val, state;
ip = ibuff + 4*ch; /* input pointer to 4-byte block state-initializer */
i_inc = 4*(chans-1); /* amount by which to incr ip after each 4-byte read */
val = (short)(ip[0] + (ip[1]<<8)); /* need cast for sign-extend */
state = ip[2];
if (state > ISSTMAX) {
lsx_warn("IMA_ADPCM block ch%d initial-state (%d) out of range", ch, state);
state = 0;
}
/* specs say to ignore ip[3] , but write it as 0 */
ip += 4+i_inc;
op = obuff;
*op = val; /* 1st output sample for this channel */
op += o_inc;
for (i = 1; i < n; i++) {
int step,dp,c,cm;
if (i&1) { /* 1st of pair */
cm = *ip & 0x0f;
} else {
cm = (*ip++)>>4;
if ((i&7) == 0) /* ends the 8-sample input block for this channel */
ip += i_inc; /* skip ip for next group */
}
step = imaStepSizeTable[state];
/* Update the state for the next sample */
c = cm & 0x07;
state = imaStateAdjustTable[state][c];
dp = 0;
if (c & 4) dp += step;
step = step >> 1;
if (c & 2) dp += step;
step = step >> 1;
if (c & 1) dp += step;
step = step >> 1;
dp += step;
if (c != cm) {
val -= dp;
if (val<-0x8000) val = -0x8000;
} else {
val += dp;
if (val>0x7fff) val = 0x7fff;
}
*op = val;
op += o_inc;
}
return;
}
/* lsx_ima_block_expand_i() outputs interleaved samples into one output buffer */
void lsx_ima_block_expand_i(
unsigned chans, /* total channels */
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* output samples, n*chans */
int n /* samples to decode PER channel, REQUIRE n % 8 == 1 */
)
{
unsigned ch;
for (ch=0; ch<chans; ch++)
ImaExpandS(ch, chans, ibuff, obuff+ch, n, chans);
}
/* lsx_ima_block_expand_m() outputs non-interleaved samples into chan separate output buffers */
void lsx_ima_block_expand_m(
unsigned chans, /* total channels */
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL **obuffs, /* chan output sample buffers, each takes n samples */
int n /* samples to decode PER channel, REQUIRE n % 8 == 1 */
)
{
unsigned ch;
for (ch=0; ch<chans; ch++)
ImaExpandS(ch, chans, ibuff, obuffs[ch], n, 1);
}
static int ImaMashS(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
int v0, /* value to use as starting prediction0 */
const SAMPL *ibuff, /* ibuff[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE n % 8 == 1 */
int *st, /* input/output state, REQUIRE 0 <= *st <= ISSTMAX */
unsigned char *obuff /* output buffer[blockAlign], or NULL for no output */
)
{
const SAMPL *ip, *itop;
unsigned char *op;
int o_inc = 0; /* set 0 only to shut up gcc's 'might be uninitialized' */
int i, val;
int state;
double d2; /* long long is okay also, speed abt the same */
ip = ibuff + ch; /* point ip to 1st input sample for this channel */
itop = ibuff + n*chans;
val = *ip - v0; ip += chans;/* 1st input sample for this channel */
d2 = val*val;/* d2 will be sum of squares of errors, given input v0 and *st */
val = v0;
op = obuff; /* output pointer (or NULL) */
if (op) { /* NULL means don't output, just compute the rms error */
op += 4*ch; /* where to put this channel's 4-byte block state-initializer */
o_inc = 4*(chans-1); /* amount by which to incr op after each 4-byte written */
*op++ = val; *op++ = val>>8;
*op++ = *st; *op++ = 0; /* they could have put a mid-block state-correction here */
op += o_inc; /* _sigh_ NEVER waste a byte. It's a rule! */
}
state = *st;
for (i = 0; ip < itop; ip+=chans) {
int step,d,dp,c;
d = *ip - val; /* difference between last prediction and current sample */
step = imaStepSizeTable[state];
c = (abs(d)<<2)/step;
if (c > 7) c = 7;
/* Update the state for the next sample */
state = imaStateAdjustTable[state][c];
if (op) { /* if we want output, put it in proper place */
int cm = c;
if (d<0) cm |= 8;
if (i&1) { /* odd numbered output */
*op++ |= (cm<<4);
if (i == 7) /* ends the 8-sample output block for this channel */
op += o_inc; /* skip op for next group */
} else {
*op = cm;
}
i = (i+1) & 0x07;
}
dp = 0;
if (c & 4) dp += step;
step = step >> 1;
if (c & 2) dp += step;
step = step >> 1;
if (c & 1) dp += step;
step = step >> 1;
dp += step;
if (d<0) {
val -= dp;
if (val<-0x8000) val = -0x8000;
} else {
val += dp;
if (val>0x7fff) val = 0x7fff;
}
{
int x = *ip - val;
d2 += x*x;
}
}
d2 /= n; /* be sure it's non-negative */
*st = state;
return (int) sqrt(d2);
}
/* mash one channel... if you want to use opt>0, 9 is a reasonable value */
inline static void ImaMashChannel(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE n % 8 == 1 */
int *st, /* input/output state, REQUIRE 0 <= *st <= ISSTMAX */
unsigned char *obuff, /* output buffer[blockAlign] */
int opt /* non-zero allows some cpu-intensive code to improve output */
)
{
int snext;
int s0,d0;
s0 = *st;
if (opt>0) {
int low,hi,w;
int low0,hi0;
snext = s0;
d0 = ImaMashS(ch, chans, ip[ch], ip,n,&snext, NULL);
w = 0;
low=hi=s0;
low0 = low-opt; if (low0<0) low0=0;
hi0 = hi+opt; if (hi0>ISSTMAX) hi0=ISSTMAX;
while (low>low0 || hi<hi0) {
if (!w && low>low0) {
int d2;
snext = --low;
d2 = ImaMashS(ch, chans, ip[ch], ip,n,&snext, NULL);
if (d2<d0) {
d0=d2; s0=low;
low0 = low-opt; if (low0<0) low0=0;
hi0 = low+opt; if (hi0>ISSTMAX) hi0=ISSTMAX;
}
}
if (w && hi<hi0) {
int d2;
snext = ++hi;
d2 = ImaMashS(ch, chans, ip[ch], ip,n,&snext, NULL);
if (d2<d0) {
d0=d2; s0=hi;
low0 = hi-opt; if (low0<0) low0=0;
hi0 = hi+opt; if (hi0>ISSTMAX) hi0=ISSTMAX;
}
}
w=1-w;
}
*st = s0;
}
ImaMashS(ch, chans, ip[ch], ip,n,st, obuff);
}
/* mash one block. if you want to use opt>0, 9 is a reasonable value */
void lsx_ima_block_mash_i(
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE n % 8 == 1 */
int *st, /* input/output state, REQUIRE 0 <= *st <= ISSTMAX */
unsigned char *obuff, /* output buffer[blockAlign] */
int opt /* non-zero allows some cpu-intensive code to improve output */
)
{
unsigned ch;
for (ch=0; ch<chans; ch++)
ImaMashChannel(ch, chans, ip, n, st+ch, obuff, opt);
}
/*
* lsx_ima_samples_in(dataLen, chans, blockAlign, samplesPerBlock)
* returns the number of samples/channel which would go
* in the dataLen, given the other parameters ...
* if input samplesPerBlock is 0, then returns the max
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing.
*/
size_t lsx_ima_samples_in(
size_t dataLen,
size_t chans,
size_t blockAlign,
size_t samplesPerBlock
)
{
size_t m, n;
if (samplesPerBlock) {
n = (dataLen / blockAlign) * samplesPerBlock;
m = (dataLen % blockAlign);
} else {
n = 0;
m = blockAlign;
}
if (m >= (size_t)4*chans) {
m -= 4*chans; /* number of bytes beyond block-header */
m /= 4*chans; /* number of 4-byte blocks/channel beyond header */
m = 8*m + 1; /* samples/chan beyond header + 1 in header */
if (samplesPerBlock && m > samplesPerBlock) m = samplesPerBlock;
n += m;
}
return n;
/*wSamplesPerBlock = ((wBlockAlign - 4*wChannels)/(4*wChannels))*8 + 1;*/
}
/*
* size_t lsx_ima_bytes_per_block(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
size_t lsx_ima_bytes_per_block(
size_t chans,
size_t samplesPerBlock
)
{
size_t n;
/* per channel, ima has blocks of len 4, the 1st has 1st sample, the others
* up to 8 samples per block,
* so number of later blocks is (nsamp-1 + 7)/8, total blocks/chan is
* (nsamp-1+7)/8 + 1 = (nsamp+14)/8
*/
n = ((size_t)samplesPerBlock + 14)/8 * 4 * chans;
return n;
}

84
app/src/main/cpp/sox/ima_rw.h
Unescape View File

@ -0,0 +1,84 @@
/* libSoX ima_rw.h -- codex utilities for WAV_FORMAT_IMA_ADPCM
* Copyright (C) 1999 Stanley J. Brooks <stabro@megsinet.net>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox.h"
#ifndef SAMPL
#define SAMPL short
#endif
/*
* call lsx_ima_init_table() before any other Ima* functions,
* to create the fast lookup tables
*/
extern void lsx_ima_init_table(void);
/* lsx_ima_block_expand_i() outputs interleaved samples into one output buffer */
extern void lsx_ima_block_expand_i(
unsigned chans, /* total channels */
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* output samples, n*chans */
int n /* samples to decode PER channel, REQUIRE n % 8 == 1 */
);
/* lsx_ima_block_expand_m() outputs non-interleaved samples into chan separate output buffers */
extern void lsx_ima_block_expand_m(
unsigned chans, /* total channels */
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL **obuffs, /* chan output sample buffers, each takes n samples */
int n /* samples to decode PER channel, REQUIRE n % 8 == 1 */
);
/* mash one block. if you want to use opt>0, 9 is a reasonable value */
extern void lsx_ima_block_mash_i(
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE n % 8 == 1 */
int *st, /* input/output state[chans], REQUIRE 0 <= st[ch] <= ISSTMAX */
unsigned char *obuff, /* output buffer[blockAlign] */
int opt /* non-zero allows some cpu-intensive code to improve output */
);
/* Some helper functions for computing samples/block and blockalign */
/*
* lsx_ima_samples_in(dataLen, chans, blockAlign, samplesPerBlock)
* returns the number of samples/channel which would go
* in the dataLen, given the other parameters ...
* if input samplesPerBlock is 0, then returns the max
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
extern size_t lsx_ima_samples_in(
size_t dataLen,
size_t chans,
size_t blockAlign,
size_t samplesPerBlock
);
/*
* size_t lsx_ima_bytes_per_block(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
extern size_t lsx_ima_bytes_per_block(
size_t chans,
size_t samplesPerBlock
);

58
app/src/main/cpp/sox/input.c
Unescape View File

@ -0,0 +1,58 @@
/* libSoX effect: Input audio from a file (c) 2008 robs@users.sourceforge.net
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "sox_i.h"
typedef struct {sox_format_t * file;} priv_t;
static int getopts(sox_effect_t * effp, int argc, char * * argv)
{
priv_t * p = (priv_t *)effp->priv;
if (argc != 2 || !(p->file = (sox_format_t *)argv[1]) || p->file->mode != 'r')
return SOX_EOF;
return SOX_SUCCESS;
}
static int drain(
sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
/* ensure that *osamp is a multiple of the number of channels. */
*osamp -= *osamp % effp->out_signal.channels;
/* Read up to *osamp samples into obuf; store the actual number read
* back to *osamp */
*osamp = sox_read(p->file, obuf, *osamp);
/* sox_read may return a number that is less than was requested; only if
* 0 samples is returned does it indicate that end-of-file has been reached
* or an error has occurred */
if (!*osamp && p->file->sox_errno)
lsx_fail("%s: %s", p->file->filename, p->file->sox_errstr);
return *osamp? SOX_SUCCESS : SOX_EOF;
}
sox_effect_handler_t const * lsx_input_effect_fn(void)
{
static sox_effect_handler_t handler = {
"input", NULL, SOX_EFF_MCHAN | SOX_EFF_LENGTH | SOX_EFF_INTERNAL,
getopts, NULL, NULL, drain, NULL, NULL, sizeof(priv_t)
};
return &handler;
}

Some files were not shown because too many files have changed in this diff Show More

Write Preview
Loading…
Cancel
Save