Mercurial > sdl-ios-xcode

comparison src/audio/SDL_audiocvt.c @ 2760:02aa80d7905f

Find changesets by keywords (author, files, the commit message), revision number or hash, or revset expression.
Updated Visual C++ build
author Sam Lantinga <slouken@libsdl.org>
date Mon, 15 Sep 2008 07:34:36 +0000
parents 79c1bd651f04
children f55c87ae336b
comparison
equal deleted inserted replaced
2759:95fccd9bf262 2760:02aa80d7905f
26 26
27 /* Functions for audio drivers to perform runtime conversion of audio format */ 27 /* Functions for audio drivers to perform runtime conversion of audio format */
28 28
29 #include "SDL_audio.h" 29 #include "SDL_audio.h"
30 #include "SDL_audio_c.h" 30 #include "SDL_audio_c.h"
31
32 #include "../libm/math.h"
31 33
32 //#define DEBUG_CONVERT 34 //#define DEBUG_CONVERT
33 35
34 /* These are fractional multiplication routines. That is, their inputs 36 /* These are fractional multiplication routines. That is, their inputs
35 are two numbers in the range [-1, 1) and the result falls in that 37 are two numbers in the range [-1, 1) and the result falls in that
1350 type of dst_format. Also assume the buffer is allocated. 1352 type of dst_format. Also assume the buffer is allocated.
1351 Note the buffer needs to be 6 units long. 1353 Note the buffer needs to be 6 units long.
1352 For now, use RBJ's cookbook coefficients. It might be more 1354 For now, use RBJ's cookbook coefficients. It might be more
1353 optimal to create a Butterworth filter, but this is more difficult. 1355 optimal to create a Butterworth filter, but this is more difficult.
1354 */ 1356 */
1357 #if 0
1355 int 1358 int
1356 SDL_BuildIIRLowpass(SDL_AudioCVT * cvt, SDL_AudioFormat format) 1359 SDL_BuildIIRLowpass(SDL_AudioCVT * cvt, SDL_AudioFormat format)
1357 { 1360 {
1358 float fc; /* cutoff frequency */ 1361 float fc; /* cutoff frequency */
1359 float coeff[6]; /* floating point iir coefficients b0, b1, b2, a0, a1, a2 */ 1362 float coeff[6]; /* floating point iir coefficients b0, b1, b2, a0, a1, a2 */
1441 1444
1442 /* Initialize the state buffer to all zeroes, and set initial position */ 1445 /* Initialize the state buffer to all zeroes, and set initial position */
1443 SDL_memset(cvt->state_buf, 0, 4 * SDL_AUDIO_BITSIZE(format) / 4); 1446 SDL_memset(cvt->state_buf, 0, 4 * SDL_AUDIO_BITSIZE(format) / 4);
1444 cvt->state_pos = 0; 1447 cvt->state_pos = 0;
1445 #undef convert_fixed 1448 #undef convert_fixed
1446 } 1449
1450 return 0;
1451 }
1452 #endif
1447 1453
1448 /* Apply the lowpass IIR filter to the given SDL_AudioCVT struct */ 1454 /* Apply the lowpass IIR filter to the given SDL_AudioCVT struct */
1449 /* This was implemented because it would be much faster than the fir filter, 1455 /* This was implemented because it would be much faster than the fir filter,
1450 but it doesn't seem to have a steep enough cutoff so we'd need several 1456 but it doesn't seem to have a steep enough cutoff so we'd need several
1451 cascaded biquads, which probably isn't a great idea. Therefore, this 1457 cascaded biquads, which probably isn't a great idea. Therefore, this
1665 /* Apply blackman window */ 1671 /* Apply blackman window */
1666 fSinc[i] *= 1672 fSinc[i] *=
1667 0.42f - 0.5f * cosf(two_pi_over_m * (float) i) + 1673 0.42f - 0.5f * cosf(two_pi_over_m * (float) i) +
1668 0.08f * cosf(four_pi_over_m * (float) i); 1674 0.08f * cosf(four_pi_over_m * (float) i);
1669 } 1675 }
1670 norm_sum += fabs(fSinc[i]); 1676 norm_sum += fSinc[i] < 0 ? -fSinc[i] : fSinc[i]; /* fabs(fSinc[i]); */
1671 } 1677 }
1672 1678
1673 norm_fact = 1.0f / norm_sum; 1679 norm_fact = 1.0f / norm_sum;
1674 1680
1675 #define convert_fixed(type, fix) { \ 1681 #define convert_fixed(type, fix) { \