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/* audioopmodule - Module to detect peak values in arrays */
#define PY_SSIZE_T_CLEAN
#include "Python.h"
#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 PyModuleDef audioopmodule;
typedef struct {
PyObject *AudioopError;
} _audioopstate;
#define _audioopstate(o) ((_audioopstate *)PyModule_GetState(o))
static int
audioop_check_size(PyObject *module, int size)
{
if (size < 1 || size > 4) {
PyErr_SetString(_audioopstate(module)->AudioopError, "Size should be 1, 2, 3 or 4");
return 0;
}
else
return 1;
}
static int
audioop_check_parameters(PyObject *module, Py_ssize_t len, int size)
{
if (!audioop_check_size(module, size))
return 0;
if (len % size != 0) {
PyErr_SetString(_audioopstate(module)->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(module, fragment->len, width))
return NULL;
if (index < 0 || index >= fragment->len/width) {
PyErr_SetString(_audioopstate(module)->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(module, 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(module, 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(module, 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(module, 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(_audioopstate(module)->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(_audioopstate(module)->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(_audioopstate(module)->AudioopError, "Strings should be even-sized");
return NULL;
}
if (fragment->len != reference->len) {
PyErr_SetString(_audioopstate(module)->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(_audioopstate(module)->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(_audioopstate(module)->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(module, 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(module, 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(module, 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(module, 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(module, len, width))
return NULL;
if (((len / width) & 1) != 0) {
PyErr_SetString(_audioopstate(module)->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(module, 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(module, fragment1->len, width))
return NULL;
if (fragment1->len != fragment2->len) {
PyErr_SetString(_audioopstate(module)->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(module, 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(module, 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(module, 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(module, fragment->len, width))
return NULL;
if (!audioop_check_size(module, 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(module, width))
return NULL;
if (nchannels < 1) {
PyErr_SetString(_audioopstate(module)->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(_audioopstate(module)->AudioopError,
"weightA should be >= 1, weightB should be >= 0");
return NULL;
}
assert(fragment->len >= 0);
if (fragment->len % bytes_per_frame != 0) {
PyErr_SetString(_audioopstate(module)->AudioopError, "not a whole number of frames");
return NULL;
}
if (inrate <= 0 || outrate <= 0) {
PyErr_SetString(_audioopstate(module)->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!;ratecv(): illegal state argument",
&d, &PyTuple_Type, &samps))
goto exit;
if (PyTuple_Size(samps) != nchannels) {
PyErr_SetString(_audioopstate(module)->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(): illegal state argument",
&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(module, 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(module, 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(module, 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(module, 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(module, 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;lin2adpcm(): illegal state argument",
&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(module, 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;adpcm2lin(): illegal state argument",
&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 "clinic/audioop.c.h"
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 int
audioop_traverse(PyObject *m, visitproc visit, void *arg) {
_audioopstate *state = _audioopstate(m);
if (state != NULL)
Py_VISIT(state->AudioopError);
return 0;
}
static int
audioop_clear(PyObject *m) {
_audioopstate *state = _audioopstate(m);
if (state != NULL)
Py_CLEAR(state->AudioopError);
return 0;
}
static void
audioop_free(void *m) {
audioop_clear((PyObject *)m);
}
static struct PyModuleDef audioopmodule = {
PyModuleDef_HEAD_INIT,
"audioop",
NULL,
sizeof(_audioopstate),
audioop_methods,
NULL,
audioop_traverse,
audioop_clear,
audioop_free
};
PyMODINIT_FUNC
PyInit_audioop(void)
{
PyObject *m = PyModule_Create(&audioopmodule);
if (m == NULL)
return NULL;
PyObject *AudioopError = PyErr_NewException("audioop.error", NULL, NULL);
if (AudioopError == NULL)
return NULL;
Py_INCREF(AudioopError);
PyModule_AddObject(m, "error", AudioopError);
_audioopstate(m)->AudioopError = AudioopError;
return m;
}
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