mirrors/cosmopolitan
1
0
Fork 0
mirror of https://github.com/jart/cosmopolitan.git synced 2025-02-01 03:53:33 +00:00
Code Issues Projects Releases Packages Wiki Activity Actions
cosmopolitan/third_party/python/Modules/audioop.c
Justine Tunney 9b29358511 Make whitespace changes 
Status lines for Emacs and Vim have been added to Python sources so
they'll be easier to edit using Python's preferred coding style.

Some DNS helper functions have been broken up into multiple files. It's
nice to have one function per file whenever possible, since that way we
don't need -ffunction-sections.  Another reason it's good to have small
source files, is because the build will be enforcing resource limits on
compilation and testing soon.
2021-08-13 03:20:45 -07:00

1935 lines
58 KiB
C
Raw Blame History

/*-*- mode:c;indent-tabs-mode:nil;c-basic-offset:4;tab-width:8;coding:utf-8 -*-│
│vi: set net ft=c ts=4 sts=4 sw=4 fenc=utf-8 :vi│
╞══════════════════════════════════════════════════════════════════════════════╡
│ Python 3 │
│ https://docs.python.org/3/license.html │
╚─────────────────────────────────────────────────────────────────────────────*/
#define PY_SSIZE_T_CLEAN
#include "libc/math.h"
#include "third_party/python/Include/dictobject.h"
#include "third_party/python/Include/floatobject.h"
#include "third_party/python/Include/longobject.h"
#include "third_party/python/Include/modsupport.h"
#include "third_party/python/Include/object.h"
#include "third_party/python/Include/pyerrors.h"
#include "third_party/python/Include/pymacro.h"
#include "third_party/python/Include/pymem.h"
#include "third_party/python/Include/tupleobject.h"
/* clang-format off */
/* audioopmodule - Module to detect peak values in arrays */
#if defined(__CHAR_UNSIGNED__)
#if defined(signed)
/* This module currently does not work on systems where only unsigned
characters are available. Take it out of Setup. Sorry. */
#endif
#endif
static const int maxvals[] = {0, 0x7F, 0x7FFF, 0x7FFFFF, 0x7FFFFFFF};
/* -1 trick is needed on Windows to support -0x80000000 without a warning */
static const int minvals[] = {0, -0x80, -0x8000, -0x800000, -0x7FFFFFFF-1};
static const unsigned int masks[] = {0, 0xFF, 0xFFFF, 0xFFFFFF, 0xFFFFFFFF};
static int
fbound(double val, double minval, double maxval)
{
if (val > maxval) {
val = maxval;
}
else if (val < minval + 1.0) {
val = minval;
}
/* Round towards minus infinity (-inf) */
val = floor(val);
/* Cast double to integer: round towards zero */
return (int)val;
}
/* Code shamelessly stolen from sox, 12.17.7, g711.c
** (c) Craig Reese, Joe Campbell and Jeff Poskanzer 1989 */
/* From g711.c:
*
* December 30, 1994:
* Functions linear2alaw, linear2ulaw have been updated to correctly
* convert unquantized 16 bit values.
* Tables for direct u- to A-law and A- to u-law conversions have been
* corrected.
* Borge Lindberg, Center for PersonKommunikation, Aalborg University.
* bli@cpk.auc.dk
*
*/
#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
#define CLIP 32635
#define SIGN_BIT (0x80) /* Sign bit for an A-law byte. */
#define QUANT_MASK (0xf) /* Quantization field mask. */
#define SEG_SHIFT (4) /* Left shift for segment number. */
#define SEG_MASK (0x70) /* Segment field mask. */
static const int16_t seg_aend[8] = {
0x1F, 0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF
};
static const int16_t seg_uend[8] = {
0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF
};
static int16_t
search(int16_t val, const int16_t *table, int size)
{
int i;
for (i = 0; i < size; i++) {
if (val <= *table++)
return (i);
}
return (size);
}
#define st_ulaw2linear16(uc) (_st_ulaw2linear16[uc])
#define st_alaw2linear16(uc) (_st_alaw2linear16[uc])
static const int16_t _st_ulaw2linear16[256] = {
-32124, -31100, -30076, -29052, -28028, -27004, -25980,
-24956, -23932, -22908, -21884, -20860, -19836, -18812,
-17788, -16764, -15996, -15484, -14972, -14460, -13948,
-13436, -12924, -12412, -11900, -11388, -10876, -10364,
-9852, -9340, -8828, -8316, -7932, -7676, -7420,
-7164, -6908, -6652, -6396, -6140, -5884, -5628,
-5372, -5116, -4860, -4604, -4348, -4092, -3900,
-3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108,
-1980, -1884, -1820, -1756, -1692, -1628, -1564,
-1500, -1436, -1372, -1308, -1244, -1180, -1116,
-1052, -988, -924, -876, -844, -812, -780,
-748, -716, -684, -652, -620, -588, -556,
-524, -492, -460, -428, -396, -372, -356,
-340, -324, -308, -292, -276, -260, -244,
-228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72,
-64, -56, -48, -40, -32, -24, -16,
-8, 0, 32124, 31100, 30076, 29052, 28028,
27004, 25980, 24956, 23932, 22908, 21884, 20860,
19836, 18812, 17788, 16764, 15996, 15484, 14972,
14460, 13948, 13436, 12924, 12412, 11900, 11388,
10876, 10364, 9852, 9340, 8828, 8316, 7932,
7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348,
4092, 3900, 3772, 3644, 3516, 3388, 3260,
3132, 3004, 2876, 2748, 2620, 2492, 2364,
2236, 2108, 1980, 1884, 1820, 1756, 1692,
1628, 1564, 1500, 1436, 1372, 1308, 1244,
1180, 1116, 1052, 988, 924, 876, 844,
812, 780, 748, 716, 684, 652, 620,
588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276,
260, 244, 228, 212, 196, 180, 164,
148, 132, 120, 112, 104, 96, 88,
80, 72, 64, 56, 48, 40, 32,
24, 16, 8, 0
};
/*
* linear2ulaw() accepts a 14-bit signed integer and encodes it as u-law data
* stored in an unsigned char. This function should only be called with
* the data shifted such that it only contains information in the lower
* 14-bits.
*
* In order to simplify the encoding process, the original linear magnitude
* is biased by adding 33 which shifts the encoding range from (0 - 8158) to
* (33 - 8191). The result can be seen in the following encoding table:
*
* Biased Linear Input Code Compressed Code
* ------------------------ ---------------
* 00000001wxyza 000wxyz
* 0000001wxyzab 001wxyz
* 000001wxyzabc 010wxyz
* 00001wxyzabcd 011wxyz
* 0001wxyzabcde 100wxyz
* 001wxyzabcdef 101wxyz
* 01wxyzabcdefg 110wxyz
* 1wxyzabcdefgh 111wxyz
*
* Each biased linear code has a leading 1 which identifies the segment
* number. The value of the segment number is equal to 7 minus the number
* of leading 0's. The quantization interval is directly available as the
* four bits wxyz. * The trailing bits (a - h) are ignored.
*
* Ordinarily the complement of the resulting code word is used for
* transmission, and so the code word is complemented before it is returned.
*
* For further information see John C. Bellamy's Digital Telephony, 1982,
* John Wiley & Sons, pps 98-111 and 472-476.
*/
static unsigned char
st_14linear2ulaw(int16_t pcm_val) /* 2's complement (14-bit range) */
{
int16_t mask;
int16_t seg;
unsigned char uval;
/* u-law inverts all bits */
/* Get the sign and the magnitude of the value. */
if (pcm_val < 0) {
pcm_val = -pcm_val;
mask = 0x7F;
} else {
mask = 0xFF;
}
if ( pcm_val > CLIP ) pcm_val = CLIP; /* clip the magnitude */
pcm_val += (BIAS >> 2);
/* Convert the scaled magnitude to segment number. */
seg = search(pcm_val, seg_uend, 8);
/*
* Combine the sign, segment, quantization bits;
* and complement the code word.
*/
if (seg >= 8) /* out of range, return maximum value. */
return (unsigned char) (0x7F ^ mask);
else {
uval = (unsigned char) (seg << 4) | ((pcm_val >> (seg + 1)) & 0xF);
return (uval ^ mask);
}
}
static const int16_t _st_alaw2linear16[256] = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992,
-4736, -7552, -7296, -8064, -7808, -6528, -6272,
-7040, -6784, -2752, -2624, -3008, -2880, -2240,
-2112, -2496, -2368, -3776, -3648, -4032, -3904,
-3264, -3136, -3520, -3392, -22016, -20992, -24064,
-23040, -17920, -16896, -19968, -18944, -30208, -29184,
-32256, -31232, -26112, -25088, -28160, -27136, -11008,
-10496, -12032, -11520, -8960, -8448, -9984, -9472,
-15104, -14592, -16128, -15616, -13056, -12544, -14080,
-13568, -344, -328, -376, -360, -280, -264,
-312, -296, -472, -456, -504, -488, -408,
-392, -440, -424, -88, -72, -120, -104,
-24, -8, -56, -40, -216, -200, -248,
-232, -152, -136, -184, -168, -1376, -1312,
-1504, -1440, -1120, -1056, -1248, -1184, -1888,
-1824, -2016, -1952, -1632, -1568, -1760, -1696,
-688, -656, -752, -720, -560, -528, -624,
-592, -944, -912, -1008, -976, -816, -784,
-880, -848, 5504, 5248, 6016, 5760, 4480,
4224, 4992, 4736, 7552, 7296, 8064, 7808,
6528, 6272, 7040, 6784, 2752, 2624, 3008,
2880, 2240, 2112, 2496, 2368, 3776, 3648,
4032, 3904, 3264, 3136, 3520, 3392, 22016,
20992, 24064, 23040, 17920, 16896, 19968, 18944,
30208, 29184, 32256, 31232, 26112, 25088, 28160,
27136, 11008, 10496, 12032, 11520, 8960, 8448,
9984, 9472, 15104, 14592, 16128, 15616, 13056,
12544, 14080, 13568, 344, 328, 376, 360,
280, 264, 312, 296, 472, 456, 504,
488, 408, 392, 440, 424, 88, 72,
120, 104, 24, 8, 56, 40, 216,
200, 248, 232, 152, 136, 184, 168,
1376, 1312, 1504, 1440, 1120, 1056, 1248,
1184, 1888, 1824, 2016, 1952, 1632, 1568,
1760, 1696, 688, 656, 752, 720, 560,
528, 624, 592, 944, 912, 1008, 976,
816, 784, 880, 848
};
/*
* linear2alaw() accepts a 13-bit signed integer and encodes it as A-law data
* stored in an unsigned char. This function should only be called with
* the data shifted such that it only contains information in the lower
* 13-bits.
*
* Linear Input Code Compressed Code
* ------------------------ ---------------
* 0000000wxyza 000wxyz
* 0000001wxyza 001wxyz
* 000001wxyzab 010wxyz
* 00001wxyzabc 011wxyz
* 0001wxyzabcd 100wxyz
* 001wxyzabcde 101wxyz
* 01wxyzabcdef 110wxyz
* 1wxyzabcdefg 111wxyz
*
* For further information see John C. Bellamy's Digital Telephony, 1982,
* John Wiley & Sons, pps 98-111 and 472-476.
*/
static unsigned char
st_linear2alaw(int16_t pcm_val) /* 2's complement (13-bit range) */
{
int16_t mask;
int16_t seg;
unsigned char aval;
/* A-law using even bit inversion */
if (pcm_val >= 0) {
mask = 0xD5; /* sign (7th) bit = 1 */
} else {
mask = 0x55; /* sign bit = 0 */
pcm_val = -pcm_val - 1;
}
/* Convert the scaled magnitude to segment number. */
seg = search(pcm_val, seg_aend, 8);
/* Combine the sign, segment, and quantization bits. */
if (seg >= 8) /* out of range, return maximum value. */
return (unsigned char) (0x7F ^ mask);
else {
aval = (unsigned char) seg << SEG_SHIFT;
if (seg < 2)
aval |= (pcm_val >> 1) & QUANT_MASK;
else
aval |= (pcm_val >> seg) & QUANT_MASK;
return (aval ^ mask);
}
}
/* End of code taken from sox */
/* Intel ADPCM step variation table */
static const int indexTable[16] = {
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8,
};
static const int stepsizeTable[89] = {
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 GETINTX(T, cp, i) (*(T *)((unsigned char *)(cp) + (i)))
#define SETINTX(T, cp, i, val) do { \
*(T *)((unsigned char *)(cp) + (i)) = (T)(val); \
} while (0)
#define GETINT8(cp, i) GETINTX(signed char, (cp), (i))
#define GETINT16(cp, i) GETINTX(int16_t, (cp), (i))
#define GETINT32(cp, i) GETINTX(int32_t, (cp), (i))
#if WORDS_BIGENDIAN
#define GETINT24(cp, i) ( \
((unsigned char *)(cp) + (i))[2] + \
(((unsigned char *)(cp) + (i))[1] << 8) + \
(((signed char *)(cp) + (i))[0] << 16) )
#else
#define GETINT24(cp, i) ( \
((unsigned char *)(cp) + (i))[0] + \
(((unsigned char *)(cp) + (i))[1] << 8) + \
(((signed char *)(cp) + (i))[2] << 16) )
#endif
#define SETINT8(cp, i, val) SETINTX(signed char, (cp), (i), (val))
#define SETINT16(cp, i, val) SETINTX(int16_t, (cp), (i), (val))
#define SETINT32(cp, i, val) SETINTX(int32_t, (cp), (i), (val))
#if WORDS_BIGENDIAN
#define SETINT24(cp, i, val) do { \
((unsigned char *)(cp) + (i))[2] = (int)(val); \
((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \
((signed char *)(cp) + (i))[0] = (int)(val) >> 16; \
} while (0)
#else
#define SETINT24(cp, i, val) do { \
((unsigned char *)(cp) + (i))[0] = (int)(val); \
((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \
((signed char *)(cp) + (i))[2] = (int)(val) >> 16; \
} while (0)
#endif
#define GETRAWSAMPLE(size, cp, i) ( \
(size == 1) ? (int)GETINT8((cp), (i)) : \
(size == 2) ? (int)GETINT16((cp), (i)) : \
(size == 3) ? (int)GETINT24((cp), (i)) : \
(int)GETINT32((cp), (i)))
#define SETRAWSAMPLE(size, cp, i, val) do { \
if (size == 1) \
SETINT8((cp), (i), (val)); \
else if (size == 2) \
SETINT16((cp), (i), (val)); \
else if (size == 3) \
SETINT24((cp), (i), (val)); \
else \
SETINT32((cp), (i), (val)); \
} while(0)
#define GETSAMPLE32(size, cp, i) ( \
(size == 1) ? (int)GETINT8((cp), (i)) << 24 : \
(size == 2) ? (int)GETINT16((cp), (i)) << 16 : \
(size == 3) ? (int)GETINT24((cp), (i)) << 8 : \
(int)GETINT32((cp), (i)))
#define SETSAMPLE32(size, cp, i, val) do { \
if (size == 1) \
SETINT8((cp), (i), (val) >> 24); \
else if (size == 2) \
SETINT16((cp), (i), (val) >> 16); \
else if (size == 3) \
SETINT24((cp), (i), (val) >> 8); \
else \
SETINT32((cp), (i), (val)); \
} while(0)
static PyObject *AudioopError;
static int
audioop_check_size(int size)
{
if (size < 1 || size > 4) {
PyErr_SetString(AudioopError, "Size should be 1, 2, 3 or 4");
return 0;
}
else
return 1;
}
static int
audioop_check_parameters(Py_ssize_t len, int size)
{
if (!audioop_check_size(size))
return 0;
if (len % size != 0) {
PyErr_SetString(AudioopError, "not a whole number of frames");
return 0;
}
return 1;
}
/*[clinic input]
module audioop
[clinic start generated code]*/
/*[clinic end generated code: output=da39a3ee5e6b4b0d input=8fa8f6611be3591a]*/
/*[clinic input]
audioop.getsample
fragment: Py_buffer
width: int
index: Py_ssize_t
/
Return the value of sample index from the fragment.
[clinic start generated code]*/
static PyObject *
audioop_getsample_impl(PyObject *module, Py_buffer *fragment, int width,
Py_ssize_t index)
/*[clinic end generated code: output=8fe1b1775134f39a input=88edbe2871393549]*/
{
int val;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
if (index < 0 || index >= fragment->len/width) {
PyErr_SetString(AudioopError, "Index out of range");
return NULL;
}
val = GETRAWSAMPLE(width, fragment->buf, index*width);
return PyLong_FromLong(val);
}
/*[clinic input]
audioop.max
fragment: Py_buffer
width: int
/
Return the maximum of the absolute value of all samples in a fragment.
[clinic start generated code]*/
static PyObject *
audioop_max_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=e6c5952714f1c3f0 input=32bea5ea0ac8c223]*/
{
Py_ssize_t i;
unsigned int absval, max = 0;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
for (i = 0; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i);
/* Cast to unsigned before negating. Unsigned overflow is well-
defined, but signed overflow is not. */
if (val < 0) absval = (unsigned int)-(int64_t)val;
else absval = val;
if (absval > max) max = absval;
}
return PyLong_FromUnsignedLong(max);
}
/*[clinic input]
audioop.minmax
fragment: Py_buffer
width: int
/
Return the minimum and maximum values of all samples in the sound fragment.
[clinic start generated code]*/
static PyObject *
audioop_minmax_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=473fda66b15c836e input=89848e9b927a0696]*/
{
Py_ssize_t i;
/* -1 trick below is needed on Windows to support -0x80000000 without
a warning */
int min = 0x7fffffff, max = -0x7FFFFFFF-1;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
for (i = 0; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i);
if (val > max) max = val;
if (val < min) min = val;
}
return Py_BuildValue("(ii)", min, max);
}
/*[clinic input]
audioop.avg
fragment: Py_buffer
width: int
/
Return the average over all samples in the fragment.
[clinic start generated code]*/
static PyObject *
audioop_avg_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=4410a4c12c3586e6 input=1114493c7611334d]*/
{
Py_ssize_t i;
int avg;
double sum = 0.0;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
for (i = 0; i < fragment->len; i += width)
sum += GETRAWSAMPLE(width, fragment->buf, i);
if (fragment->len == 0)
avg = 0;
else
avg = (int)floor(sum / (double)(fragment->len/width));
return PyLong_FromLong(avg);
}
/*[clinic input]
audioop.rms
fragment: Py_buffer
width: int
/
Return the root-mean-square of the fragment, i.e. sqrt(sum(S_i^2)/n).
[clinic start generated code]*/
static PyObject *
audioop_rms_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=1e7871c826445698 input=4cc57c6c94219d78]*/
{
Py_ssize_t i;
unsigned int res;
double sum_squares = 0.0;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
for (i = 0; i < fragment->len; i += width) {
double val = GETRAWSAMPLE(width, fragment->buf, i);
sum_squares += val*val;
}
if (fragment->len == 0)
res = 0;
else
res = (unsigned int)sqrt(sum_squares / (double)(fragment->len/width));
return PyLong_FromUnsignedLong(res);
}
static double _sum2(const int16_t *a, const int16_t *b, Py_ssize_t len)
{
Py_ssize_t i;
double sum = 0.0;
for( i=0; i<len; i++) {
sum = sum + (double)a[i]*(double)b[i];
}
return sum;
}
/*
** Findfit tries to locate a sample within another sample. Its main use
** is in echo-cancellation (to find the feedback of the output signal in
** the input signal).
** The method used is as follows:
**
** let R be the reference signal (length n) and A the input signal (length N)
** with N > n, and let all sums be over i from 0 to n-1.
**
** Now, for each j in {0..N-n} we compute a factor fj so that -fj*R matches A
** as good as possible, i.e. sum( (A[j+i]+fj*R[i])^2 ) is minimal. This
** equation gives fj = sum( A[j+i]R[i] ) / sum(R[i]^2).
**
** Next, we compute the relative distance between the original signal and
** the modified signal and minimize that over j:
** vj = sum( (A[j+i]-fj*R[i])^2 ) / sum( A[j+i]^2 ) =>
** vj = ( sum(A[j+i]^2)*sum(R[i]^2) - sum(A[j+i]R[i])^2 ) / sum( A[j+i]^2 )
**
** In the code variables correspond as follows:
** cp1 A
** cp2 R
** len1 N
** len2 n
** aj_m1 A[j-1]
** aj_lm1 A[j+n-1]
** sum_ri_2 sum(R[i]^2)
** sum_aij_2 sum(A[i+j]^2)
** sum_aij_ri sum(A[i+j]R[i])
**
** sum_ri is calculated once, sum_aij_2 is updated each step and sum_aij_ri
** is completely recalculated each step.
*/
/*[clinic input]
audioop.findfit
fragment: Py_buffer
reference: Py_buffer
/
Try to match reference as well as possible to a portion of fragment.
[clinic start generated code]*/
static PyObject *
audioop_findfit_impl(PyObject *module, Py_buffer *fragment,
Py_buffer *reference)
/*[clinic end generated code: output=5752306d83cbbada input=62c305605e183c9a]*/
{
const int16_t *cp1, *cp2;
Py_ssize_t len1, len2;
Py_ssize_t j, best_j;
double aj_m1, aj_lm1;
double sum_ri_2, sum_aij_2, sum_aij_ri, result, best_result, factor;
if (fragment->len & 1 || reference->len & 1) {
PyErr_SetString(AudioopError, "Strings should be even-sized");
return NULL;
}
cp1 = (const int16_t *)fragment->buf;
len1 = fragment->len >> 1;
cp2 = (const int16_t *)reference->buf;
len2 = reference->len >> 1;
if (len1 < len2) {
PyErr_SetString(AudioopError, "First sample should be longer");
return NULL;
}
sum_ri_2 = _sum2(cp2, cp2, len2);
sum_aij_2 = _sum2(cp1, cp1, len2);
sum_aij_ri = _sum2(cp1, cp2, len2);
result = (sum_ri_2*sum_aij_2 - sum_aij_ri*sum_aij_ri) / sum_aij_2;
best_result = result;
best_j = 0;
for ( j=1; j<=len1-len2; j++) {
aj_m1 = (double)cp1[j-1];
aj_lm1 = (double)cp1[j+len2-1];
sum_aij_2 = sum_aij_2 + aj_lm1*aj_lm1 - aj_m1*aj_m1;
sum_aij_ri = _sum2(cp1+j, cp2, len2);
result = (sum_ri_2*sum_aij_2 - sum_aij_ri*sum_aij_ri)
/ sum_aij_2;
if ( result < best_result ) {
best_result = result;
best_j = j;
}
}
factor = _sum2(cp1+best_j, cp2, len2) / sum_ri_2;
return Py_BuildValue("(nf)", best_j, factor);
}
/*
** findfactor finds a factor f so that the energy in A-fB is minimal.
** See the comment for findfit for details.
*/
/*[clinic input]
audioop.findfactor
fragment: Py_buffer
reference: Py_buffer
/
Return a factor F such that rms(add(fragment, mul(reference, -F))) is minimal.
[clinic start generated code]*/
static PyObject *
audioop_findfactor_impl(PyObject *module, Py_buffer *fragment,
Py_buffer *reference)
/*[clinic end generated code: output=14ea95652c1afcf8 input=816680301d012b21]*/
{
const int16_t *cp1, *cp2;
Py_ssize_t len;
double sum_ri_2, sum_aij_ri, result;
if (fragment->len & 1 || reference->len & 1) {
PyErr_SetString(AudioopError, "Strings should be even-sized");
return NULL;
}
if (fragment->len != reference->len) {
PyErr_SetString(AudioopError, "Samples should be same size");
return NULL;
}
cp1 = (const int16_t *)fragment->buf;
cp2 = (const int16_t *)reference->buf;
len = fragment->len >> 1;
sum_ri_2 = _sum2(cp2, cp2, len);
sum_aij_ri = _sum2(cp1, cp2, len);
result = sum_aij_ri / sum_ri_2;
return PyFloat_FromDouble(result);
}
/*
** findmax returns the index of the n-sized segment of the input sample
** that contains the most energy.
*/
/*[clinic input]
audioop.findmax
fragment: Py_buffer
length: Py_ssize_t
/
Search fragment for a slice of specified number of samples with maximum energy.
[clinic start generated code]*/
static PyObject *
audioop_findmax_impl(PyObject *module, Py_buffer *fragment,
Py_ssize_t length)
/*[clinic end generated code: output=f008128233523040 input=2f304801ed42383c]*/
{
const int16_t *cp1;
Py_ssize_t len1;
Py_ssize_t j, best_j;
double aj_m1, aj_lm1;
double result, best_result;
if (fragment->len & 1) {
PyErr_SetString(AudioopError, "Strings should be even-sized");
return NULL;
}
cp1 = (const int16_t *)fragment->buf;
len1 = fragment->len >> 1;
if (length < 0 || len1 < length) {
PyErr_SetString(AudioopError, "Input sample should be longer");
return NULL;
}
result = _sum2(cp1, cp1, length);
best_result = result;
best_j = 0;
for ( j=1; j<=len1-length; j++) {
aj_m1 = (double)cp1[j-1];
aj_lm1 = (double)cp1[j+length-1];
result = result + aj_lm1*aj_lm1 - aj_m1*aj_m1;
if ( result > best_result ) {
best_result = result;
best_j = j;
}
}
return PyLong_FromSsize_t(best_j);
}
/*[clinic input]
audioop.avgpp
fragment: Py_buffer
width: int
/
Return the average peak-peak value over all samples in the fragment.
[clinic start generated code]*/
static PyObject *
audioop_avgpp_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=269596b0d5ae0b2b input=0b3cceeae420a7d9]*/
{
Py_ssize_t i;
int prevval, prevextremevalid = 0, prevextreme = 0;
double sum = 0.0;
unsigned int avg;
int diff, prevdiff, nextreme = 0;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
if (fragment->len <= width)
return PyLong_FromLong(0);
prevval = GETRAWSAMPLE(width, fragment->buf, 0);
prevdiff = 17; /* Anything != 0, 1 */
for (i = width; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i);
if (val != prevval) {
diff = val < prevval;
if (prevdiff == !diff) {
/* Derivative changed sign. Compute difference to last
** extreme value and remember.
*/
if (prevextremevalid) {
if (prevval < prevextreme)
sum += (double)((unsigned int)prevextreme -
(unsigned int)prevval);
else
sum += (double)((unsigned int)prevval -
(unsigned int)prevextreme);
nextreme++;
}
prevextremevalid = 1;
prevextreme = prevval;
}
prevval = val;
prevdiff = diff;
}
}
if ( nextreme == 0 )
avg = 0;
else
avg = (unsigned int)(sum / (double)nextreme);
return PyLong_FromUnsignedLong(avg);
}
/*[clinic input]
audioop.maxpp
fragment: Py_buffer
width: int
/
Return the maximum peak-peak value in the sound fragment.
[clinic start generated code]*/
static PyObject *
audioop_maxpp_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=5b918ed5dbbdb978 input=671a13e1518f80a1]*/
{
Py_ssize_t i;
int prevval, prevextremevalid = 0, prevextreme = 0;
unsigned int max = 0, extremediff;
int diff, prevdiff;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
if (fragment->len <= width)
return PyLong_FromLong(0);
prevval = GETRAWSAMPLE(width, fragment->buf, 0);
prevdiff = 17; /* Anything != 0, 1 */
for (i = width; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i);
if (val != prevval) {
diff = val < prevval;
if (prevdiff == !diff) {
/* Derivative changed sign. Compute difference to
** last extreme value and remember.
*/
if (prevextremevalid) {
if (prevval < prevextreme)
extremediff = (unsigned int)prevextreme -
(unsigned int)prevval;
else
extremediff = (unsigned int)prevval -
(unsigned int)prevextreme;
if ( extremediff > max )
max = extremediff;
}
prevextremevalid = 1;
prevextreme = prevval;
}
prevval = val;
prevdiff = diff;
}
}
return PyLong_FromUnsignedLong(max);
}
/*[clinic input]
audioop.cross
fragment: Py_buffer
width: int
/
Return the number of zero crossings in the fragment passed as an argument.
[clinic start generated code]*/
static PyObject *
audioop_cross_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=5938dcdd74a1f431 input=b1b3f15b83f6b41a]*/
{
Py_ssize_t i;
int prevval;
Py_ssize_t ncross;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
ncross = -1;
prevval = 17; /* Anything <> 0,1 */
for (i = 0; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i) < 0;
if (val != prevval) ncross++;
prevval = val;
}
return PyLong_FromSsize_t(ncross);
}
/*[clinic input]
audioop.mul
fragment: Py_buffer
width: int
factor: double
/
Return a fragment that has all samples in the original fragment multiplied by the floating-point value factor.
[clinic start generated code]*/
static PyObject *
audioop_mul_impl(PyObject *module, Py_buffer *fragment, int width,
double factor)
/*[clinic end generated code: output=6cd48fe796da0ea4 input=c726667baa157d3c]*/
{
signed char *ncp;
Py_ssize_t i;
double maxval, minval;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
maxval = (double) maxvals[width];
minval = (double) minvals[width];
rv = PyBytes_FromStringAndSize(NULL, fragment->len);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
double val = GETRAWSAMPLE(width, fragment->buf, i);
int ival = fbound(val * factor, minval, maxval);
SETRAWSAMPLE(width, ncp, i, ival);
}
return rv;
}
/*[clinic input]
audioop.tomono
fragment: Py_buffer
width: int
lfactor: double
rfactor: double
/
Convert a stereo fragment to a mono fragment.
[clinic start generated code]*/
static PyObject *
audioop_tomono_impl(PyObject *module, Py_buffer *fragment, int width,
double lfactor, double rfactor)
/*[clinic end generated code: output=235c8277216d4e4e input=c4ec949b3f4dddfa]*/
{
signed char *cp, *ncp;
Py_ssize_t len, i;
double maxval, minval;
PyObject *rv;
cp = fragment->buf;
len = fragment->len;
if (!audioop_check_parameters(len, width))
return NULL;
if (((len / width) & 1) != 0) {
PyErr_SetString(AudioopError, "not a whole number of frames");
return NULL;
}
maxval = (double) maxvals[width];
minval = (double) minvals[width];
rv = PyBytes_FromStringAndSize(NULL, len/2);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
for (i = 0; i < len; i += width*2) {
double val1 = GETRAWSAMPLE(width, cp, i);
double val2 = GETRAWSAMPLE(width, cp, i + width);
double val = val1 * lfactor + val2 * rfactor;
int ival = fbound(val, minval, maxval);
SETRAWSAMPLE(width, ncp, i/2, ival);
}
return rv;
}
/*[clinic input]
audioop.tostereo
fragment: Py_buffer
width: int
lfactor: double
rfactor: double
/
Generate a stereo fragment from a mono fragment.
[clinic start generated code]*/
static PyObject *
audioop_tostereo_impl(PyObject *module, Py_buffer *fragment, int width,
double lfactor, double rfactor)
/*[clinic end generated code: output=046f13defa5f1595 input=27b6395ebfdff37a]*/
{
signed char *ncp;
Py_ssize_t i;
double maxval, minval;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
maxval = (double) maxvals[width];
minval = (double) minvals[width];
if (fragment->len > PY_SSIZE_T_MAX/2) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
rv = PyBytes_FromStringAndSize(NULL, fragment->len*2);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
double val = GETRAWSAMPLE(width, fragment->buf, i);
int val1 = fbound(val * lfactor, minval, maxval);
int val2 = fbound(val * rfactor, minval, maxval);
SETRAWSAMPLE(width, ncp, i*2, val1);
SETRAWSAMPLE(width, ncp, i*2 + width, val2);
}
return rv;
}
/*[clinic input]
audioop.add
fragment1: Py_buffer
fragment2: Py_buffer
width: int
/
Return a fragment which is the addition of the two samples passed as parameters.
[clinic start generated code]*/
static PyObject *
audioop_add_impl(PyObject *module, Py_buffer *fragment1,
Py_buffer *fragment2, int width)
/*[clinic end generated code: output=60140af4d1aab6f2 input=4a8d4bae4c1605c7]*/
{
signed char *ncp;
Py_ssize_t i;
int minval, maxval, newval;
PyObject *rv;
if (!audioop_check_parameters(fragment1->len, width))
return NULL;
if (fragment1->len != fragment2->len) {
PyErr_SetString(AudioopError, "Lengths should be the same");
return NULL;
}
maxval = maxvals[width];
minval = minvals[width];
rv = PyBytes_FromStringAndSize(NULL, fragment1->len);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
for (i = 0; i < fragment1->len; i += width) {
int val1 = GETRAWSAMPLE(width, fragment1->buf, i);
int val2 = GETRAWSAMPLE(width, fragment2->buf, i);
if (width < 4) {
newval = val1 + val2;
/* truncate in case of overflow */
if (newval > maxval)
newval = maxval;
else if (newval < minval)
newval = minval;
}
else {
double fval = (double)val1 + (double)val2;
/* truncate in case of overflow */
newval = fbound(fval, minval, maxval);
}
SETRAWSAMPLE(width, ncp, i, newval);
}
return rv;
}
/*[clinic input]
audioop.bias
fragment: Py_buffer
width: int
bias: int
/
Return a fragment that is the original fragment with a bias added to each sample.
[clinic start generated code]*/
static PyObject *
audioop_bias_impl(PyObject *module, Py_buffer *fragment, int width, int bias)
/*[clinic end generated code: output=6e0aa8f68f045093 input=2b5cce5c3bb4838c]*/
{
signed char *ncp;
Py_ssize_t i;
unsigned int val = 0, mask;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
rv = PyBytes_FromStringAndSize(NULL, fragment->len);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
mask = masks[width];
for (i = 0; i < fragment->len; i += width) {
if (width == 1)
val = GETINTX(unsigned char, fragment->buf, i);
else if (width == 2)
val = GETINTX(uint16_t, fragment->buf, i);
else if (width == 3)
val = ((unsigned int)GETINT24(fragment->buf, i)) & 0xffffffu;
else {
assert(width == 4);
val = GETINTX(uint32_t, fragment->buf, i);
}
val += (unsigned int)bias;
/* wrap around in case of overflow */
val &= mask;
if (width == 1)
SETINTX(unsigned char, ncp, i, val);
else if (width == 2)
SETINTX(uint16_t, ncp, i, val);
else if (width == 3)
SETINT24(ncp, i, (int)val);
else {
assert(width == 4);
SETINTX(uint32_t, ncp, i, val);
}
}
return rv;
}
/*[clinic input]
audioop.reverse
fragment: Py_buffer
width: int
/
Reverse the samples in a fragment and returns the modified fragment.
[clinic start generated code]*/
static PyObject *
audioop_reverse_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=b44135698418da14 input=668f890cf9f9d225]*/
{
unsigned char *ncp;
Py_ssize_t i;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
rv = PyBytes_FromStringAndSize(NULL, fragment->len);
if (rv == NULL)
return NULL;
ncp = (unsigned char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
int val = GETRAWSAMPLE(width, fragment->buf, i);
SETRAWSAMPLE(width, ncp, fragment->len - i - width, val);
}
return rv;
}
/*[clinic input]
audioop.byteswap
fragment: Py_buffer
width: int
/
Convert big-endian samples to little-endian and vice versa.
[clinic start generated code]*/
static PyObject *
audioop_byteswap_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=50838a9e4b87cd4d input=fae7611ceffa5c82]*/
{
unsigned char *ncp;
Py_ssize_t i;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
rv = PyBytes_FromStringAndSize(NULL, fragment->len);
if (rv == NULL)
return NULL;
ncp = (unsigned char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
int j;
for (j = 0; j < width; j++)
ncp[i + width - 1 - j] = ((unsigned char *)fragment->buf)[i + j];
}
return rv;
}
/*[clinic input]
audioop.lin2lin
fragment: Py_buffer
width: int
newwidth: int
/
Convert samples between 1-, 2-, 3- and 4-byte formats.
[clinic start generated code]*/
static PyObject *
audioop_lin2lin_impl(PyObject *module, Py_buffer *fragment, int width,
int newwidth)
/*[clinic end generated code: output=17b14109248f1d99 input=5ce08c8aa2f24d96]*/
{
unsigned char *ncp;
Py_ssize_t i, j;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
if (!audioop_check_size(newwidth))
return NULL;
if (fragment->len/width > PY_SSIZE_T_MAX/newwidth) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
rv = PyBytes_FromStringAndSize(NULL, (fragment->len/width)*newwidth);
if (rv == NULL)
return NULL;
ncp = (unsigned char *)PyBytes_AsString(rv);
for (i = j = 0; i < fragment->len; i += width, j += newwidth) {
int val = GETSAMPLE32(width, fragment->buf, i);
SETSAMPLE32(newwidth, ncp, j, val);
}
return rv;
}
static int
gcd(int a, int b)
{
while (b > 0) {
int tmp = a % b;
a = b;
b = tmp;
}
return a;
}
/*[clinic input]
audioop.ratecv
fragment: Py_buffer
width: int
nchannels: int
inrate: int
outrate: int
state: object
weightA: int = 1
weightB: int = 0
/
Convert the frame rate of the input fragment.
[clinic start generated code]*/
static PyObject *
audioop_ratecv_impl(PyObject *module, Py_buffer *fragment, int width,
int nchannels, int inrate, int outrate, PyObject *state,
int weightA, int weightB)
/*[clinic end generated code: output=624038e843243139 input=aff3acdc94476191]*/
{
char *cp, *ncp;
Py_ssize_t len;
int chan, d, *prev_i, *cur_i, cur_o;
PyObject *samps, *str, *rv = NULL, *channel;
int bytes_per_frame;
if (!audioop_check_size(width))
return NULL;
if (nchannels < 1) {
PyErr_SetString(AudioopError, "# of channels should be >= 1");
return NULL;
}
if (width > INT_MAX / nchannels) {
/* This overflow test is rigorously correct because
both multiplicands are >= 1. Use the argument names
from the docs for the error msg. */
PyErr_SetString(PyExc_OverflowError,
"width * nchannels too big for a C int");
return NULL;
}
bytes_per_frame = width * nchannels;
if (weightA < 1 || weightB < 0) {
PyErr_SetString(AudioopError,
"weightA should be >= 1, weightB should be >= 0");
return NULL;
}
assert(fragment->len >= 0);
if (fragment->len % bytes_per_frame != 0) {
PyErr_SetString(AudioopError, "not a whole number of frames");
return NULL;
}
if (inrate <= 0 || outrate <= 0) {
PyErr_SetString(AudioopError, "sampling rate not > 0");
return NULL;
}
/* divide inrate and outrate by their greatest common divisor */
d = gcd(inrate, outrate);
inrate /= d;
outrate /= d;
/* divide weightA and weightB by their greatest common divisor */
d = gcd(weightA, weightB);
weightA /= d;
weightB /= d;
if ((size_t)nchannels > SIZE_MAX/sizeof(int)) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
prev_i = (int *) PyMem_Malloc(nchannels * sizeof(int));
cur_i = (int *) PyMem_Malloc(nchannels * sizeof(int));
if (prev_i == NULL || cur_i == NULL) {
(void) PyErr_NoMemory();
goto exit;
}
len = fragment->len / bytes_per_frame; /* # of frames */
if (state == Py_None) {
d = -outrate;
for (chan = 0; chan < nchannels; chan++)
prev_i[chan] = cur_i[chan] = 0;
}
else {
if (!PyTuple_Check(state)) {
PyErr_SetString(PyExc_TypeError, "state must be a tuple or None");
goto exit;
}
if (!PyArg_ParseTuple(state,
"iO!;audioop.ratecv: illegal state argument",
&d, &PyTuple_Type, &samps))
goto exit;
if (PyTuple_Size(samps) != nchannels) {
PyErr_SetString(AudioopError,
"illegal state argument");
goto exit;
}
for (chan = 0; chan < nchannels; chan++) {
channel = PyTuple_GetItem(samps, chan);
if (!PyTuple_Check(channel)) {
PyErr_SetString(PyExc_TypeError,
"ratecv(): illegal state argument");
goto exit;
}
if (!PyArg_ParseTuple(channel,
"ii:ratecv", &prev_i[chan],
&cur_i[chan]))
goto exit;
}
}
/* str <- Space for the output buffer. */
if (len == 0)
str = PyBytes_FromStringAndSize(NULL, 0);
else {
/* There are len input frames, so we need (mathematically)
ceiling(len*outrate/inrate) output frames, and each frame
requires bytes_per_frame bytes. Computing this
without spurious overflow is the challenge; we can
settle for a reasonable upper bound, though, in this
case ceiling(len/inrate) * outrate. */
/* compute ceiling(len/inrate) without overflow */
Py_ssize_t q = 1 + (len - 1) / inrate;
if (outrate > PY_SSIZE_T_MAX / q / bytes_per_frame)
str = NULL;
else
str = PyBytes_FromStringAndSize(NULL,
q * outrate * bytes_per_frame);
}
if (str == NULL) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
goto exit;
}
ncp = PyBytes_AsString(str);
cp = fragment->buf;
for (;;) {
while (d < 0) {
if (len == 0) {
samps = PyTuple_New(nchannels);
if (samps == NULL)
goto exit;
for (chan = 0; chan < nchannels; chan++)
PyTuple_SetItem(samps, chan,
Py_BuildValue("(ii)",
prev_i[chan],
cur_i[chan]));
if (PyErr_Occurred())
goto exit;
/* We have checked before that the length
* of the string fits into int. */
len = (Py_ssize_t)(ncp - PyBytes_AsString(str));
rv = PyBytes_FromStringAndSize
(PyBytes_AsString(str), len);
Py_DECREF(str);
str = rv;
if (str == NULL)
goto exit;
rv = Py_BuildValue("(O(iO))", str, d, samps);
Py_DECREF(samps);
Py_DECREF(str);
goto exit; /* return rv */
}
for (chan = 0; chan < nchannels; chan++) {
prev_i[chan] = cur_i[chan];
cur_i[chan] = GETSAMPLE32(width, cp, 0);
cp += width;
/* implements a simple digital filter */
cur_i[chan] = (int)(
((double)weightA * (double)cur_i[chan] +
(double)weightB * (double)prev_i[chan]) /
((double)weightA + (double)weightB));
}
len--;
d += outrate;
}
while (d >= 0) {
for (chan = 0; chan < nchannels; chan++) {
cur_o = (int)(((double)prev_i[chan] * (double)d +
(double)cur_i[chan] * (double)(outrate - d)) /
(double)outrate);
SETSAMPLE32(width, ncp, 0, cur_o);
ncp += width;
}
d -= inrate;
}
}
exit:
PyMem_Free(prev_i);
PyMem_Free(cur_i);
return rv;
}
/*[clinic input]
audioop.lin2ulaw
fragment: Py_buffer
width: int
/
Convert samples in the audio fragment to u-LAW encoding.
[clinic start generated code]*/
static PyObject *
audioop_lin2ulaw_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=14fb62b16fe8ea8e input=2450d1b870b6bac2]*/
{
unsigned char *ncp;
Py_ssize_t i;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
rv = PyBytes_FromStringAndSize(NULL, fragment->len/width);
if (rv == NULL)
return NULL;
ncp = (unsigned char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
int val = GETSAMPLE32(width, fragment->buf, i);
*ncp++ = st_14linear2ulaw(val >> 18);
}
return rv;
}
/*[clinic input]
audioop.ulaw2lin
fragment: Py_buffer
width: int
/
Convert sound fragments in u-LAW encoding to linearly encoded sound fragments.
[clinic start generated code]*/
static PyObject *
audioop_ulaw2lin_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=378356b047521ba2 input=45d53ddce5be7d06]*/
{
unsigned char *cp;
signed char *ncp;
Py_ssize_t i;
PyObject *rv;
if (!audioop_check_size(width))
return NULL;
if (fragment->len > PY_SSIZE_T_MAX/width) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
rv = PyBytes_FromStringAndSize(NULL, fragment->len*width);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
cp = fragment->buf;
for (i = 0; i < fragment->len*width; i += width) {
int val = st_ulaw2linear16(*cp++) << 16;
SETSAMPLE32(width, ncp, i, val);
}
return rv;
}
/*[clinic input]
audioop.lin2alaw
fragment: Py_buffer
width: int
/
Convert samples in the audio fragment to a-LAW encoding.
[clinic start generated code]*/
static PyObject *
audioop_lin2alaw_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=d076f130121a82f0 input=ffb1ef8bb39da945]*/
{
unsigned char *ncp;
Py_ssize_t i;
PyObject *rv;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
rv = PyBytes_FromStringAndSize(NULL, fragment->len/width);
if (rv == NULL)
return NULL;
ncp = (unsigned char *)PyBytes_AsString(rv);
for (i = 0; i < fragment->len; i += width) {
int val = GETSAMPLE32(width, fragment->buf, i);
*ncp++ = st_linear2alaw(val >> 19);
}
return rv;
}
/*[clinic input]
audioop.alaw2lin
fragment: Py_buffer
width: int
/
Convert sound fragments in a-LAW encoding to linearly encoded sound fragments.
[clinic start generated code]*/
static PyObject *
audioop_alaw2lin_impl(PyObject *module, Py_buffer *fragment, int width)
/*[clinic end generated code: output=85c365ec559df647 input=4140626046cd1772]*/
{
unsigned char *cp;
signed char *ncp;
Py_ssize_t i;
int val;
PyObject *rv;
if (!audioop_check_size(width))
return NULL;
if (fragment->len > PY_SSIZE_T_MAX/width) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
rv = PyBytes_FromStringAndSize(NULL, fragment->len*width);
if (rv == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(rv);
cp = fragment->buf;
for (i = 0; i < fragment->len*width; i += width) {
val = st_alaw2linear16(*cp++) << 16;
SETSAMPLE32(width, ncp, i, val);
}
return rv;
}
/*[clinic input]
audioop.lin2adpcm
fragment: Py_buffer
width: int
state: object
/
Convert samples to 4 bit Intel/DVI ADPCM encoding.
[clinic start generated code]*/
static PyObject *
audioop_lin2adpcm_impl(PyObject *module, Py_buffer *fragment, int width,
PyObject *state)
/*[clinic end generated code: output=cc19f159f16c6793 input=12919d549b90c90a]*/
{
signed char *ncp;
Py_ssize_t i;
int step, valpred, delta,
index, sign, vpdiff, diff;
PyObject *rv = NULL, *str;
int outputbuffer = 0, bufferstep;
if (!audioop_check_parameters(fragment->len, width))
return NULL;
/* Decode state, should have (value, step) */
if ( state == Py_None ) {
/* First time, it seems. Set defaults */
valpred = 0;
index = 0;
}
else if (!PyTuple_Check(state)) {
PyErr_SetString(PyExc_TypeError, "state must be a tuple or None");
return NULL;
}
else if (!PyArg_ParseTuple(state, "ii", &valpred, &index)) {
return NULL;
}
else if (valpred >= 0x8000 || valpred < -0x8000 ||
(size_t)index >= Py_ARRAY_LENGTH(stepsizeTable)) {
PyErr_SetString(PyExc_ValueError, "bad state");
return NULL;
}
str = PyBytes_FromStringAndSize(NULL, fragment->len/(width*2));
if (str == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(str);
step = stepsizeTable[index];
bufferstep = 1;
for (i = 0; i < fragment->len; i += width) {
int val = GETSAMPLE32(width, fragment->buf, i) >> 16;
/* Step 1 - compute difference with previous value */
if (val < valpred) {
diff = valpred - val;
sign = 8;
}
else {
diff = val - valpred;
sign = 0;
}
/* Step 2 - Divide and clamp */
/* Note:
** This code *approximately* computes:
** delta = diff*4/step;
** vpdiff = (delta+0.5)*step/4;
** but in shift step bits are dropped. The net result of this
** is that even if you have fast mul/div hardware you cannot
** put it to good use since the fixup would be too expensive.
*/
delta = 0;
vpdiff = (step >> 3);
if ( diff >= step ) {
delta = 4;
diff -= step;
vpdiff += step;
}
step >>= 1;
if ( diff >= step ) {
delta |= 2;
diff -= step;
vpdiff += step;
}
step >>= 1;
if ( diff >= step ) {
delta |= 1;
vpdiff += step;
}
/* Step 3 - Update previous value */
if ( sign )
valpred -= vpdiff;
else
valpred += vpdiff;
/* Step 4 - Clamp previous value to 16 bits */
if ( valpred > 32767 )
valpred = 32767;
else if ( valpred < -32768 )
valpred = -32768;
/* Step 5 - Assemble value, update index and step values */
delta |= sign;
index += indexTable[delta];
if ( index < 0 ) index = 0;
if ( index > 88 ) index = 88;
step = stepsizeTable[index];
/* Step 6 - Output value */
if ( bufferstep ) {
outputbuffer = (delta << 4) & 0xf0;
} else {
*ncp++ = (delta & 0x0f) | outputbuffer;
}
bufferstep = !bufferstep;
}
rv = Py_BuildValue("(O(ii))", str, valpred, index);
Py_DECREF(str);
return rv;
}
/*[clinic input]
audioop.adpcm2lin
fragment: Py_buffer
width: int
state: object
/
Decode an Intel/DVI ADPCM coded fragment to a linear fragment.
[clinic start generated code]*/
static PyObject *
audioop_adpcm2lin_impl(PyObject *module, Py_buffer *fragment, int width,
PyObject *state)
/*[clinic end generated code: output=3440ea105acb3456 input=f5221144f5ca9ef0]*/
{
signed char *cp;
signed char *ncp;
Py_ssize_t i, outlen;
int valpred, step, delta, index, sign, vpdiff;
PyObject *rv, *str;
int inputbuffer = 0, bufferstep;
if (!audioop_check_size(width))
return NULL;
/* Decode state, should have (value, step) */
if ( state == Py_None ) {
/* First time, it seems. Set defaults */
valpred = 0;
index = 0;
}
else if (!PyTuple_Check(state)) {
PyErr_SetString(PyExc_TypeError, "state must be a tuple or None");
return NULL;
}
else if (!PyArg_ParseTuple(state, "ii", &valpred, &index)) {
return NULL;
}
else if (valpred >= 0x8000 || valpred < -0x8000 ||
(size_t)index >= Py_ARRAY_LENGTH(stepsizeTable)) {
PyErr_SetString(PyExc_ValueError, "bad state");
return NULL;
}
if (fragment->len > (PY_SSIZE_T_MAX/2)/width) {
PyErr_SetString(PyExc_MemoryError,
"not enough memory for output buffer");
return NULL;
}
outlen = fragment->len*width*2;
str = PyBytes_FromStringAndSize(NULL, outlen);
if (str == NULL)
return NULL;
ncp = (signed char *)PyBytes_AsString(str);
cp = fragment->buf;
step = stepsizeTable[index];
bufferstep = 0;
for (i = 0; i < outlen; i += width) {
/* Step 1 - get the delta value and compute next index */
if ( bufferstep ) {
delta = inputbuffer & 0xf;
} else {
inputbuffer = *cp++;
delta = (inputbuffer >> 4) & 0xf;
}
bufferstep = !bufferstep;
/* Step 2 - Find new index value (for later) */
index += indexTable[delta];
if ( index < 0 ) index = 0;
if ( index > 88 ) index = 88;
/* Step 3 - Separate sign and magnitude */
sign = delta & 8;
delta = delta & 7;
/* Step 4 - Compute difference and new predicted value */
/*
** Computes 'vpdiff = (delta+0.5)*step/4', but see comment
** in adpcm_coder.
*/
vpdiff = step >> 3;
if ( delta & 4 ) vpdiff += step;
if ( delta & 2 ) vpdiff += step>>1;
if ( delta & 1 ) vpdiff += step>>2;
if ( sign )
valpred -= vpdiff;
else
valpred += vpdiff;
/* Step 5 - clamp output value */
if ( valpred > 32767 )
valpred = 32767;
else if ( valpred < -32768 )
valpred = -32768;
/* Step 6 - Update step value */
step = stepsizeTable[index];
/* Step 6 - Output value */
SETSAMPLE32(width, ncp, i, valpred << 16);
}
rv = Py_BuildValue("(O(ii))", str, valpred, index);
Py_DECREF(str);
return rv;
}
#include "third_party/python/Modules/clinic/audioop.inc"
static PyMethodDef audioop_methods[] = {
AUDIOOP_MAX_METHODDEF
AUDIOOP_MINMAX_METHODDEF
AUDIOOP_AVG_METHODDEF
AUDIOOP_MAXPP_METHODDEF
AUDIOOP_AVGPP_METHODDEF
AUDIOOP_RMS_METHODDEF
AUDIOOP_FINDFIT_METHODDEF
AUDIOOP_FINDMAX_METHODDEF
AUDIOOP_FINDFACTOR_METHODDEF
AUDIOOP_CROSS_METHODDEF
AUDIOOP_MUL_METHODDEF
AUDIOOP_ADD_METHODDEF
AUDIOOP_BIAS_METHODDEF
AUDIOOP_ULAW2LIN_METHODDEF
AUDIOOP_LIN2ULAW_METHODDEF
AUDIOOP_ALAW2LIN_METHODDEF
AUDIOOP_LIN2ALAW_METHODDEF
AUDIOOP_LIN2LIN_METHODDEF
AUDIOOP_ADPCM2LIN_METHODDEF
AUDIOOP_LIN2ADPCM_METHODDEF
AUDIOOP_TOMONO_METHODDEF
AUDIOOP_TOSTEREO_METHODDEF
AUDIOOP_GETSAMPLE_METHODDEF
AUDIOOP_REVERSE_METHODDEF
AUDIOOP_BYTESWAP_METHODDEF
AUDIOOP_RATECV_METHODDEF
{ 0, 0 }
};
static struct PyModuleDef audioopmodule = {
PyModuleDef_HEAD_INIT,
"audioop",
NULL,
-1,
audioop_methods,
NULL,
NULL,
NULL,
NULL
};
PyMODINIT_FUNC
PyInit_audioop(void)
{
PyObject *m, *d;
m = PyModule_Create(&audioopmodule);
if (m == NULL)
return NULL;
d = PyModule_GetDict(m);
if (d == NULL)
return NULL;
AudioopError = PyErr_NewException("audioop.error", NULL, NULL);
if (AudioopError != NULL)
PyDict_SetItemString(d,"error",AudioopError);
return m;
}
Reference in a new issue View git blame Copy permalink