mirror of
https://github.com/python/cpython
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1987 lines
59 KiB
C
1987 lines
59 KiB
C
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/* audioopmodule - Module to detect peak values in arrays */
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#define PY_SSIZE_T_CLEAN
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#include "Python.h"
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static const int maxvals[] = {0, 0x7F, 0x7FFF, 0x7FFFFF, 0x7FFFFFFF};
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/* -1 trick is needed on Windows to support -0x80000000 without a warning */
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static const int minvals[] = {0, -0x80, -0x8000, -0x800000, -0x7FFFFFFF-1};
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static const unsigned int masks[] = {0, 0xFF, 0xFFFF, 0xFFFFFF, 0xFFFFFFFF};
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static int
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fbound(double val, double minval, double maxval)
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{
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if (val > maxval) {
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val = maxval;
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}
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else if (val < minval + 1.0) {
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val = minval;
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}
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/* Round towards minus infinity (-inf) */
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val = floor(val);
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/* Cast double to integer: round towards zero */
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return (int)val;
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}
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/* Code shamelessly stolen from sox, 12.17.7, g711.c
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** (c) Craig Reese, Joe Campbell and Jeff Poskanzer 1989 */
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/* From g711.c:
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*
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* December 30, 1994:
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* Functions linear2alaw, linear2ulaw have been updated to correctly
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* convert unquantized 16 bit values.
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* Tables for direct u- to A-law and A- to u-law conversions have been
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* corrected.
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* Borge Lindberg, Center for PersonKommunikation, Aalborg University.
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* bli@cpk.auc.dk
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*
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*/
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#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
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#define CLIP 32635
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#define SIGN_BIT (0x80) /* Sign bit for an A-law byte. */
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#define QUANT_MASK (0xf) /* Quantization field mask. */
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#define SEG_SHIFT (4) /* Left shift for segment number. */
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#define SEG_MASK (0x70) /* Segment field mask. */
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static const int16_t seg_aend[8] = {
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0x1F, 0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF
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};
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static const int16_t seg_uend[8] = {
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0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF
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};
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static int16_t
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search(int16_t val, const int16_t *table, int size)
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{
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int i;
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for (i = 0; i < size; i++) {
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if (val <= *table++)
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return (i);
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}
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return (size);
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}
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#define st_ulaw2linear16(uc) (_st_ulaw2linear16[uc])
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#define st_alaw2linear16(uc) (_st_alaw2linear16[uc])
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static const int16_t _st_ulaw2linear16[256] = {
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-32124, -31100, -30076, -29052, -28028, -27004, -25980,
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-24956, -23932, -22908, -21884, -20860, -19836, -18812,
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-17788, -16764, -15996, -15484, -14972, -14460, -13948,
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-13436, -12924, -12412, -11900, -11388, -10876, -10364,
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-9852, -9340, -8828, -8316, -7932, -7676, -7420,
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-7164, -6908, -6652, -6396, -6140, -5884, -5628,
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-5372, -5116, -4860, -4604, -4348, -4092, -3900,
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-3772, -3644, -3516, -3388, -3260, -3132, -3004,
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-2876, -2748, -2620, -2492, -2364, -2236, -2108,
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-1980, -1884, -1820, -1756, -1692, -1628, -1564,
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-1500, -1436, -1372, -1308, -1244, -1180, -1116,
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-1052, -988, -924, -876, -844, -812, -780,
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-748, -716, -684, -652, -620, -588, -556,
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-524, -492, -460, -428, -396, -372, -356,
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-340, -324, -308, -292, -276, -260, -244,
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-228, -212, -196, -180, -164, -148, -132,
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-120, -112, -104, -96, -88, -80, -72,
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-64, -56, -48, -40, -32, -24, -16,
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-8, 0, 32124, 31100, 30076, 29052, 28028,
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27004, 25980, 24956, 23932, 22908, 21884, 20860,
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19836, 18812, 17788, 16764, 15996, 15484, 14972,
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14460, 13948, 13436, 12924, 12412, 11900, 11388,
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10876, 10364, 9852, 9340, 8828, 8316, 7932,
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7676, 7420, 7164, 6908, 6652, 6396, 6140,
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5884, 5628, 5372, 5116, 4860, 4604, 4348,
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4092, 3900, 3772, 3644, 3516, 3388, 3260,
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3132, 3004, 2876, 2748, 2620, 2492, 2364,
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2236, 2108, 1980, 1884, 1820, 1756, 1692,
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1628, 1564, 1500, 1436, 1372, 1308, 1244,
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1180, 1116, 1052, 988, 924, 876, 844,
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812, 780, 748, 716, 684, 652, 620,
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588, 556, 524, 492, 460, 428, 396,
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372, 356, 340, 324, 308, 292, 276,
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260, 244, 228, 212, 196, 180, 164,
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148, 132, 120, 112, 104, 96, 88,
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80, 72, 64, 56, 48, 40, 32,
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24, 16, 8, 0
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};
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/*
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* linear2ulaw() accepts a 14-bit signed integer and encodes it as u-law data
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* stored in an unsigned char. This function should only be called with
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* the data shifted such that it only contains information in the lower
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* 14-bits.
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*
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* In order to simplify the encoding process, the original linear magnitude
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* is biased by adding 33 which shifts the encoding range from (0 - 8158) to
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* (33 - 8191). The result can be seen in the following encoding table:
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*
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* Biased Linear Input Code Compressed Code
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* ------------------------ ---------------
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* 00000001wxyza 000wxyz
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* 0000001wxyzab 001wxyz
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* 000001wxyzabc 010wxyz
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* 00001wxyzabcd 011wxyz
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* 0001wxyzabcde 100wxyz
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* 001wxyzabcdef 101wxyz
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* 01wxyzabcdefg 110wxyz
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* 1wxyzabcdefgh 111wxyz
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*
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* Each biased linear code has a leading 1 which identifies the segment
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* number. The value of the segment number is equal to 7 minus the number
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* of leading 0's. The quantization interval is directly available as the
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* four bits wxyz. * The trailing bits (a - h) are ignored.
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*
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* Ordinarily the complement of the resulting code word is used for
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* transmission, and so the code word is complemented before it is returned.
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*
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* For further information see John C. Bellamy's Digital Telephony, 1982,
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* John Wiley & Sons, pps 98-111 and 472-476.
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*/
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static unsigned char
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st_14linear2ulaw(int16_t pcm_val) /* 2's complement (14-bit range) */
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{
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int16_t mask;
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int16_t seg;
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unsigned char uval;
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/* u-law inverts all bits */
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/* Get the sign and the magnitude of the value. */
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if (pcm_val < 0) {
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pcm_val = -pcm_val;
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mask = 0x7F;
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} else {
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mask = 0xFF;
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}
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if ( pcm_val > CLIP ) pcm_val = CLIP; /* clip the magnitude */
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pcm_val += (BIAS >> 2);
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/* Convert the scaled magnitude to segment number. */
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seg = search(pcm_val, seg_uend, 8);
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/*
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* Combine the sign, segment, quantization bits;
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* and complement the code word.
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*/
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if (seg >= 8) /* out of range, return maximum value. */
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return (unsigned char) (0x7F ^ mask);
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else {
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uval = (unsigned char) (seg << 4) | ((pcm_val >> (seg + 1)) & 0xF);
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return (uval ^ mask);
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}
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}
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static const int16_t _st_alaw2linear16[256] = {
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-5504, -5248, -6016, -5760, -4480, -4224, -4992,
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-4736, -7552, -7296, -8064, -7808, -6528, -6272,
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-7040, -6784, -2752, -2624, -3008, -2880, -2240,
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-2112, -2496, -2368, -3776, -3648, -4032, -3904,
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-3264, -3136, -3520, -3392, -22016, -20992, -24064,
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-23040, -17920, -16896, -19968, -18944, -30208, -29184,
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-32256, -31232, -26112, -25088, -28160, -27136, -11008,
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-10496, -12032, -11520, -8960, -8448, -9984, -9472,
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-15104, -14592, -16128, -15616, -13056, -12544, -14080,
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-13568, -344, -328, -376, -360, -280, -264,
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-312, -296, -472, -456, -504, -488, -408,
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-392, -440, -424, -88, -72, -120, -104,
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-24, -8, -56, -40, -216, -200, -248,
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-232, -152, -136, -184, -168, -1376, -1312,
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-1504, -1440, -1120, -1056, -1248, -1184, -1888,
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-1824, -2016, -1952, -1632, -1568, -1760, -1696,
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-688, -656, -752, -720, -560, -528, -624,
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-592, -944, -912, -1008, -976, -816, -784,
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-880, -848, 5504, 5248, 6016, 5760, 4480,
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4224, 4992, 4736, 7552, 7296, 8064, 7808,
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6528, 6272, 7040, 6784, 2752, 2624, 3008,
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2880, 2240, 2112, 2496, 2368, 3776, 3648,
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4032, 3904, 3264, 3136, 3520, 3392, 22016,
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20992, 24064, 23040, 17920, 16896, 19968, 18944,
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30208, 29184, 32256, 31232, 26112, 25088, 28160,
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27136, 11008, 10496, 12032, 11520, 8960, 8448,
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9984, 9472, 15104, 14592, 16128, 15616, 13056,
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12544, 14080, 13568, 344, 328, 376, 360,
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280, 264, 312, 296, 472, 456, 504,
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488, 408, 392, 440, 424, 88, 72,
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120, 104, 24, 8, 56, 40, 216,
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200, 248, 232, 152, 136, 184, 168,
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1376, 1312, 1504, 1440, 1120, 1056, 1248,
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1184, 1888, 1824, 2016, 1952, 1632, 1568,
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1760, 1696, 688, 656, 752, 720, 560,
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528, 624, 592, 944, 912, 1008, 976,
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816, 784, 880, 848
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};
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/*
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* linear2alaw() accepts a 13-bit signed integer and encodes it as A-law data
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* stored in an unsigned char. This function should only be called with
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* the data shifted such that it only contains information in the lower
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* 13-bits.
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*
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* Linear Input Code Compressed Code
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* ------------------------ ---------------
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* 0000000wxyza 000wxyz
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* 0000001wxyza 001wxyz
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* 000001wxyzab 010wxyz
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* 00001wxyzabc 011wxyz
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* 0001wxyzabcd 100wxyz
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* 001wxyzabcde 101wxyz
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* 01wxyzabcdef 110wxyz
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* 1wxyzabcdefg 111wxyz
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*
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* For further information see John C. Bellamy's Digital Telephony, 1982,
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* John Wiley & Sons, pps 98-111 and 472-476.
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*/
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static unsigned char
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st_linear2alaw(int16_t pcm_val) /* 2's complement (13-bit range) */
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{
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int16_t mask;
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int16_t seg;
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unsigned char aval;
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/* A-law using even bit inversion */
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if (pcm_val >= 0) {
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mask = 0xD5; /* sign (7th) bit = 1 */
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} else {
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mask = 0x55; /* sign bit = 0 */
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pcm_val = -pcm_val - 1;
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}
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/* Convert the scaled magnitude to segment number. */
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seg = search(pcm_val, seg_aend, 8);
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/* Combine the sign, segment, and quantization bits. */
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if (seg >= 8) /* out of range, return maximum value. */
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return (unsigned char) (0x7F ^ mask);
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else {
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aval = (unsigned char) seg << SEG_SHIFT;
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if (seg < 2)
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aval |= (pcm_val >> 1) & QUANT_MASK;
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else
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aval |= (pcm_val >> seg) & QUANT_MASK;
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return (aval ^ mask);
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}
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}
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/* End of code taken from sox */
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/* Intel ADPCM step variation table */
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static const int indexTable[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8,
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};
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static const int stepsizeTable[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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#define GETINTX(T, cp, i) (*(T *)((unsigned char *)(cp) + (i)))
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#define SETINTX(T, cp, i, val) do { \
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*(T *)((unsigned char *)(cp) + (i)) = (T)(val); \
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} while (0)
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#define GETINT8(cp, i) GETINTX(signed char, (cp), (i))
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#define GETINT16(cp, i) GETINTX(int16_t, (cp), (i))
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#define GETINT32(cp, i) GETINTX(int32_t, (cp), (i))
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#ifdef WORDS_BIGENDIAN
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#define GETINT24(cp, i) ( \
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((unsigned char *)(cp) + (i))[2] + \
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(((unsigned char *)(cp) + (i))[1] << 8) + \
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(((signed char *)(cp) + (i))[0] << 16) )
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#else
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#define GETINT24(cp, i) ( \
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((unsigned char *)(cp) + (i))[0] + \
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(((unsigned char *)(cp) + (i))[1] << 8) + \
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(((signed char *)(cp) + (i))[2] << 16) )
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#endif
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#define SETINT8(cp, i, val) SETINTX(signed char, (cp), (i), (val))
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#define SETINT16(cp, i, val) SETINTX(int16_t, (cp), (i), (val))
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#define SETINT32(cp, i, val) SETINTX(int32_t, (cp), (i), (val))
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#ifdef WORDS_BIGENDIAN
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#define SETINT24(cp, i, val) do { \
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((unsigned char *)(cp) + (i))[2] = (int)(val); \
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((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \
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((signed char *)(cp) + (i))[0] = (int)(val) >> 16; \
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} while (0)
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#else
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#define SETINT24(cp, i, val) do { \
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((unsigned char *)(cp) + (i))[0] = (int)(val); \
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((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \
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((signed char *)(cp) + (i))[2] = (int)(val) >> 16; \
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} while (0)
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#endif
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#define GETRAWSAMPLE(size, cp, i) ( \
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(size == 1) ? (int)GETINT8((cp), (i)) : \
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(size == 2) ? (int)GETINT16((cp), (i)) : \
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(size == 3) ? (int)GETINT24((cp), (i)) : \
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(int)GETINT32((cp), (i)))
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#define SETRAWSAMPLE(size, cp, i, val) do { \
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if (size == 1) \
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SETINT8((cp), (i), (val)); \
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else if (size == 2) \
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SETINT16((cp), (i), (val)); \
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else if (size == 3) \
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SETINT24((cp), (i), (val)); \
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else \
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SETINT32((cp), (i), (val)); \
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} while(0)
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#define GETSAMPLE32(size, cp, i) ( \
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(size == 1) ? (int)GETINT8((cp), (i)) << 24 : \
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(size == 2) ? (int)GETINT16((cp), (i)) << 16 : \
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(size == 3) ? (int)GETINT24((cp), (i)) << 8 : \
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(int)GETINT32((cp), (i)))
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#define SETSAMPLE32(size, cp, i, val) do { \
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if (size == 1) \
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SETINT8((cp), (i), (val) >> 24); \
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else if (size == 2) \
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SETINT16((cp), (i), (val) >> 16); \
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else if (size == 3) \
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SETINT24((cp), (i), (val) >> 8); \
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else \
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SETINT32((cp), (i), (val)); \
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} while(0)
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static PyModuleDef audioopmodule;
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typedef struct {
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PyObject *AudioopError;
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} audioop_state;
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static inline audioop_state *
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get_audioop_state(PyObject *module)
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{
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void *state = PyModule_GetState(module);
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assert(state != NULL);
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return (audioop_state *)state;
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}
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static int
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audioop_check_size(PyObject *module, int size)
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{
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if (size < 1 || size > 4) {
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PyErr_SetString(get_audioop_state(module)->AudioopError,
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"Size should be 1, 2, 3 or 4");
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return 0;
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}
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else
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return 1;
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}
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static int
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audioop_check_parameters(PyObject *module, Py_ssize_t len, int size)
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{
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if (!audioop_check_size(module, size))
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return 0;
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if (len % size != 0) {
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PyErr_SetString(get_audioop_state(module)->AudioopError,
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"not a whole number of frames");
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return 0;
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}
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return 1;
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}
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/*[clinic input]
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module audioop
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[clinic start generated code]*/
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/*[clinic end generated code: output=da39a3ee5e6b4b0d input=8fa8f6611be3591a]*/
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/*[clinic input]
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audioop.getsample
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fragment: Py_buffer
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width: int
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index: Py_ssize_t
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/
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Return the value of sample index from the fragment.
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|
[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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(module)->AudioopError,
|
|
"Strings should be even-sized");
|
|
return NULL;
|
|
}
|
|
if (fragment->len != reference->len) {
|
|
PyErr_SetString(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(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(get_audioop_state(module)->AudioopError,
|
|
"not a whole number of frames");
|
|
return NULL;
|
|
}
|
|
if (inrate <= 0 || outrate <= 0) {
|
|
PyErr_SetString(get_audioop_state(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(get_audioop_state(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 *module, visitproc visit, void *arg)
|
|
{
|
|
audioop_state *state = get_audioop_state(module);
|
|
Py_VISIT(state->AudioopError);
|
|
return 0;
|
|
}
|
|
|
|
static int
|
|
audioop_clear(PyObject *module)
|
|
{
|
|
audioop_state *state = get_audioop_state(module);
|
|
Py_CLEAR(state->AudioopError);
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
audioop_free(void *module) {
|
|
audioop_clear((PyObject *)module);
|
|
}
|
|
|
|
static int
|
|
audioop_exec(PyObject* module)
|
|
{
|
|
audioop_state *state = get_audioop_state(module);
|
|
|
|
state->AudioopError = PyErr_NewException("audioop.error", NULL, NULL);
|
|
if (state->AudioopError == NULL) {
|
|
return -1;
|
|
}
|
|
|
|
Py_INCREF(state->AudioopError);
|
|
if (PyModule_AddObject(module, "error", state->AudioopError) < 0) {
|
|
Py_DECREF(state->AudioopError);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static PyModuleDef_Slot audioop_slots[] = {
|
|
{Py_mod_exec, audioop_exec},
|
|
{0, NULL}
|
|
};
|
|
|
|
static struct PyModuleDef audioopmodule = {
|
|
PyModuleDef_HEAD_INIT,
|
|
"audioop",
|
|
NULL,
|
|
sizeof(audioop_state),
|
|
audioop_methods,
|
|
audioop_slots,
|
|
audioop_traverse,
|
|
audioop_clear,
|
|
audioop_free
|
|
};
|
|
|
|
PyMODINIT_FUNC
|
|
PyInit_audioop(void)
|
|
{
|
|
if (PyErr_WarnEx(PyExc_DeprecationWarning,
|
|
"'audioop' is deprecated and slated for removal in "
|
|
"Python 3.13",
|
|
7)) {
|
|
return NULL;
|
|
}
|
|
|
|
return PyModuleDef_Init(&audioopmodule);
|
|
}
|