diff options
author | paul <paul@0c269be4-1314-0410-8aa9-9f06e86f4224> | 2008-10-24 16:04:42 +0000 |
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committer | paul <paul@0c269be4-1314-0410-8aa9-9f06e86f4224> | 2008-10-24 16:04:42 +0000 |
commit | 898f36b3171dcda7341a1acc767a7c62d3d88253 (patch) | |
tree | ee51a75415d3f797cd13e903e059554d54507e2e /drivers/alsa | |
parent | b5b34c44033fee168711e11bbfdc4bb9230b9923 (diff) | |
download | jack1-898f36b3171dcda7341a1acc767a7c62d3d88253.tar.gz |
next iteration of a cleanup of sample conversions. SUBJECT TO REVIEW. TEST. DO NOT USE FOR IMPORTANT INSTALLATIONS
git-svn-id: svn+ssh://jackaudio.org/trunk/jack@3045 0c269be4-1314-0410-8aa9-9f06e86f4224
Diffstat (limited to 'drivers/alsa')
-rw-r--r-- | drivers/alsa/memops.c | 662 |
1 files changed, 112 insertions, 550 deletions
diff --git a/drivers/alsa/memops.c b/drivers/alsa/memops.c index 024bd12..a363d32 100644 --- a/drivers/alsa/memops.c +++ b/drivers/alsa/memops.c @@ -34,19 +34,64 @@ #include <jack/memops.h> -#define SAMPLE_MAX_24BIT 8388608.0f -#define SAMPLE_MAX_16BIT 32768.0f +/* Notes about these *_SCALING values. -/* define these in case we end up on a platform that is missing + the MAX_<N>BIT values are floating point. when multiplied by + a full-scale normalized floating point sample value (-1.0..+1.0) + they should give the maxium value representable with an integer + sample type of N bits. Note that this is asymmetric. Sample ranges + for signed integer, 2's complement values are -(2^(N-1) to +(2^(N-1)-1) + + Complications + ------------- + If we use +2^(N-1) for the scaling factors, we run into a problem: + + if we start with a normalized float value of -1.0, scaling + to 24 bits would give -8388608 (-2^23), which is ideal. + But with +1.0, we get +8388608, which is technically out of range. + + We never multiply a full range normalized value by this constant, + but we could multiply it by a positive value that is close enough to +1.0 + to produce a value > +(2^(N-1)-1. + + There is no way around this paradox without wasting CPU cycles to determine + which scaling factor to use (i.e. determine if its negative or not, + use the right factor). + + So, for now (October 2008) we use 2^(N-1)-1 as the scaling factor. +*/ + +#define SAMPLE_24BIT_SCALING 8388607.0f +#define SAMPLE_16BIT_SCALING 32767.0f + +/* these are just values to use if the floating point value was out of range + */ + +#define SAMPLE_24BIT_MAX 8388607 // 2^23 - 1 +#define SAMPLE_24BIT_MIN -8388608 // - (2^23) + +#define SAMPLE_16BIT_MAX 32767 // 2^15-1 +#define SAMPLE_16BIT_MIN -32768 // - (2^15) + +/* these mark the outer edges of the range considered "within" range + for a floating point sample value. values outside (and on the boundaries) + of this range will be clipped before conversion; values within this + range will be scaled to appropriate values for the target sample + type. +*/ + +#define NORMALIZED_FLOAT_MIN -1.0f +#define NORMALIZED_FLOAT_MAX 1.0f + +/* define this in case we end up on a platform that is missing the real lrintf functions */ #define f_round(f) lrintf(f) -#define f_roundl(f) llrintf(f) /* Linear Congruential noise generator. From the music-dsp list - * less random than rand(), but good enough and 10x faster */ -inline unsigned int fast_rand(); + * less random than rand(), but good enough and 10x faster + */ inline unsigned int fast_rand() { static unsigned int seed = 22222; @@ -101,17 +146,15 @@ void sample_move_dS_floatLE (char *dst, jack_default_audio_sample_t *src, unsign void sample_move_d32u24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { - int64_t y; - int z; + int32_t z; while (nsamples--) { - y = f_roundl(*src * SAMPLE_MAX_24BIT) << 8; - if (y > INT_MAX) { - z = INT_MAX; - } else if (y < INT_MIN) { - z = INT_MIN; + if (*src <= NORMALIZED_FLOAT_MIN) { + z = SAMPLE_24BIT_MIN; + } else if (*src >= NORMALIZED_FLOAT_MAX) { + z = SAMPLE_24BIT_MAX; } else { - z = (int)y; + z = f_round (*src * SAMPLE_24BIT_SCALING) << 8; } #if __BYTE_ORDER == __LITTLE_ENDIAN dst[0]=(char)(z>>24); @@ -131,16 +174,13 @@ void sample_move_d32u24_sSs (char *dst, jack_default_audio_sample_t *src, unsign void sample_move_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { - int64_t y; - while (nsamples--) { - y = f_roundl(*src * SAMPLE_MAX_24BIT) << 8; - if (y > INT_MAX) { - *((int *) dst) = INT_MAX; - } else if (y < INT_MIN) { - *((int *) dst) = INT_MIN; + if (*src <= NORMALIZED_FLOAT_MIN) { + *((int32_t*) dst) = SAMPLE_24BIT_MIN; + } else if (*src >= NORMALIZED_FLOAT_MAX) { + *((int32_t*) dst) = SAMPLE_24BIT_MAX; } else { - *((int *) dst) = (int)y; + *((int32_t*) dst) = f_round (*src * SAMPLE_24BIT_SCALING) << 8; } dst += dst_skip; src++; @@ -170,7 +210,7 @@ void sample_move_dS_s32u24s (jack_default_audio_sample_t *dst, char *src, unsign x <<= 8; x |= (unsigned char)(src[0]); #endif - *dst = (x >> 8) / SAMPLE_MAX_24BIT; + *dst = (x >> 8) / SAMPLE_24BIT_SCALING; dst++; src += src_skip; } @@ -181,252 +221,19 @@ void sample_move_dS_s32u24 (jack_default_audio_sample_t *dst, char *src, unsigne /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - *dst = (*((int *) src) >> 8) / SAMPLE_MAX_24BIT; + *dst = (*((int *) src) >> 8) / SAMPLE_24BIT_SCALING; dst++; src += src_skip; } } -void sample_move_dither_rect_d32u24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - /* ALERT: signed sign-extension portability !!! */ - jack_default_audio_sample_t x; - int64_t y; - int z; - - while (nsamples--) { - x = *src * SAMPLE_MAX_16BIT; - x -= (float)fast_rand() / (float)INT_MAX; - y = f_roundl(x); - y <<= 16; - if (y > INT_MAX) { - z = INT_MAX; - } else if (y < INT_MIN) { - z = INT_MIN; - } else { - z = (int) y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>24); - dst[1]=(char)(z>>16); - dst[2]=(char)(z>>8); - dst[3]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); - dst[3]=(char)(z>>24); -#endif - dst += dst_skip; - src++; - } -} - -void sample_move_dither_rect_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - /* ALERT: signed sign-extension portability !!! */ - jack_default_audio_sample_t x; - int64_t y; - - while (nsamples--) { - x = *src * SAMPLE_MAX_16BIT; - x -= (float)fast_rand() / (float)INT_MAX; - y = f_roundl(x); - y <<= 16; - if (y > INT_MAX) { - *((int *) dst) = INT_MAX; - } else if (y < INT_MIN) { - *((int *) dst) = INT_MIN; - } else { - *((int *) dst) = (int) y; - } - dst += dst_skip; - src++; - } -} - -void sample_move_dither_tri_d32u24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - float r; - float rm1 = state->rm1; - int64_t y; - int z; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - x += r - rm1; - rm1 = r; - y = f_roundl(x); - y <<= 16; - - if (y > INT_MAX) { - z = INT_MAX; - } else if (y < INT_MIN) { - z = INT_MIN; - } else { - z = (int)y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>24); - dst[1]=(char)(z>>16); - dst[2]=(char)(z>>8); - dst[3]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); - dst[3]=(char)(z>>24); -#endif - dst += dst_skip; - src++; - } - state->rm1 = rm1; -} - -void sample_move_dither_tri_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - float r; - float rm1 = state->rm1; - int64_t y; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - x += r - rm1; - rm1 = r; - y = f_roundl(x); - y <<= 16; - - if (y > INT_MAX) { - *((int *) dst) = INT_MAX; - } else if (y < INT_MIN) { - *((int *) dst) = INT_MIN; - } else { - *((int *) dst) = (int)y; - } - - dst += dst_skip; - src++; - } - state->rm1 = rm1; -} - -void sample_move_dither_shaped_d32u24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - jack_default_audio_sample_t xe; /* the innput sample - filtered error */ - jack_default_audio_sample_t xp; /* x' */ - float r; - float rm1 = state->rm1; - unsigned int idx = state->idx; - int64_t y; - int z; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - /* Filter the error with Lipshitz's minimally audible FIR: - [2.033 -2.165 1.959 -1.590 0.6149] */ - xe = x - - state->e[idx] * 2.033f - + state->e[(idx - 1) & DITHER_BUF_MASK] * 2.165f - - state->e[(idx - 2) & DITHER_BUF_MASK] * 1.959f - + state->e[(idx - 3) & DITHER_BUF_MASK] * 1.590f - - state->e[(idx - 4) & DITHER_BUF_MASK] * 0.6149f; - xp = xe + r - rm1; - rm1 = r; - - /* This could be some inline asm on x86 */ - y = f_roundl(xp); - - /* Intrinsic z^-1 delay */ - idx = (idx + 1) & DITHER_BUF_MASK; - state->e[idx] = y - xe; - - y <<= 16; - - if (y > INT_MAX) { - z = INT_MAX; - } else if (y < INT_MIN) { - z = INT_MIN; - } else { - z = (int)y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>24); - dst[1]=(char)(z>>16); - dst[2]=(char)(z>>8); - dst[3]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); - dst[3]=(char)(z>>24); -#endif - dst += dst_skip; - src++; - } - state->rm1 = rm1; - state->idx = idx; -} - -void sample_move_dither_shaped_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - jack_default_audio_sample_t xe; /* the innput sample - filtered error */ - jack_default_audio_sample_t xp; /* x' */ - float r; - float rm1 = state->rm1; - unsigned int idx = state->idx; - int64_t y; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - /* Filter the error with Lipshitz's minimally audible FIR: - [2.033 -2.165 1.959 -1.590 0.6149] */ - xe = x - - state->e[idx] * 2.033f - + state->e[(idx - 1) & DITHER_BUF_MASK] * 2.165f - - state->e[(idx - 2) & DITHER_BUF_MASK] * 1.959f - + state->e[(idx - 3) & DITHER_BUF_MASK] * 1.590f - - state->e[(idx - 4) & DITHER_BUF_MASK] * 0.6149f; - xp = xe + r - rm1; - rm1 = r; - - /* This could be some inline asm on x86 */ - y = f_roundl(xp); - - /* Intrinsic z^-1 delay */ - idx = (idx + 1) & DITHER_BUF_MASK; - state->e[idx] = y - xe; - - y <<= 16; - - if (y > INT_MAX) { - *((int *) dst) = INT_MAX; - } else if (y < INT_MIN) { - *((int *) dst) = INT_MIN; - } else { - *((int *) dst) = y; - } - dst += dst_skip; - src++; - } - state->rm1 = rm1; - state->idx = idx; -} - void sample_move_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { int64_t y; int z; while (nsamples--) { - y = (int64_t)(*src * SAMPLE_MAX_24BIT); + y = (int64_t)(*src * SAMPLE_24BIT_SCALING); if (y > (INT_MAX >> 8 )) { z = (INT_MAX >> 8); @@ -454,7 +261,7 @@ void sample_move_d24_sS (char *dst, jack_default_audio_sample_t *src, unsigned l int64_t y; while (nsamples--) { - y = (int64_t)(*src * SAMPLE_MAX_24BIT); + y = (int64_t)(*src * SAMPLE_24BIT_SCALING); if (y > (INT_MAX >> 8 )) { y = (INT_MAX >> 8); @@ -498,7 +305,7 @@ void sample_move_dS_s24s (jack_default_audio_sample_t *dst, char *src, unsigned x |= 0xff << 24; } #endif - *dst = x / SAMPLE_MAX_24BIT; + *dst = x / SAMPLE_24BIT_SCALING; dst++; src += src_skip; } @@ -516,268 +323,26 @@ void sample_move_dS_s24 (jack_default_audio_sample_t *dst, char *src, unsigned l memcpy(&x, src, 3); #endif x >>= 8; - *dst = x / SAMPLE_MAX_24BIT; + *dst = x / SAMPLE_24BIT_SCALING; dst++; src += src_skip; } } -void sample_move_dither_rect_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - /* ALERT: signed sign-extension portability !!! */ - jack_default_audio_sample_t x; - int64_t y; - int z; - - while (nsamples--) { - x = *src * SAMPLE_MAX_16BIT; - x -= (float)fast_rand() / (float)INT_MAX; - y = f_roundl(x); - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - z = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - z = (INT_MIN >> 8); - } else { - z = (int)y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>16); - dst[1]=(char)(z>>8); - dst[2]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); -#endif - dst += dst_skip; - src++; - } -} - -void sample_move_dither_rect_d24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - /* ALERT: signed sign-extension portability !!! */ - jack_default_audio_sample_t x; - int64_t y; - - while (nsamples--) { - x = *src * SAMPLE_MAX_16BIT; - x -= (float)fast_rand() / (float)INT_MAX; - y = f_roundl(x); - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - y = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - y = (INT_MIN >> 8); - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - memcpy (dst, &y, 3); -#elif __BYTE_ORDER == __BIG_ENDIAN - memcpy (dst, (char *)&y + 5, 3); -#endif - - dst += dst_skip; - src++; - } -} - -void sample_move_dither_tri_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - float r; - float rm1 = state->rm1; - int64_t y; - int z; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - x += r - rm1; - rm1 = r; - y = f_roundl(x); - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - z = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - z = (INT_MIN >> 8); - } else { - z = (int)y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>16); - dst[1]=(char)(z>>8); - dst[2]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); -#endif - dst += dst_skip; - src++; - } - state->rm1 = rm1; -} - -void sample_move_dither_tri_d24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - float r; - float rm1 = state->rm1; - int64_t y; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - x += r - rm1; - rm1 = r; - y = f_roundl(x); - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - y = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - y = (INT_MIN >> 8); - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - memcpy (dst, &y, 3); -#elif __BYTE_ORDER == __BIG_ENDIAN - memcpy (dst, (char *)&y + 5, 3); -#endif - - dst += dst_skip; - src++; - } - state->rm1 = rm1; -} - -void sample_move_dither_shaped_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - jack_default_audio_sample_t xe; /* the innput sample - filtered error */ - jack_default_audio_sample_t xp; /* x' */ - float r; - float rm1 = state->rm1; - unsigned int idx = state->idx; - int64_t y; - int z; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - /* Filter the error with Lipshitz's minimally audible FIR: - [2.033 -2.165 1.959 -1.590 0.6149] */ - xe = x - - state->e[idx] * 2.033f - + state->e[(idx - 1) & DITHER_BUF_MASK] * 2.165f - - state->e[(idx - 2) & DITHER_BUF_MASK] * 1.959f - + state->e[(idx - 3) & DITHER_BUF_MASK] * 1.590f - - state->e[(idx - 4) & DITHER_BUF_MASK] * 0.6149f; - xp = xe + r - rm1; - rm1 = r; - - /* This could be some inline asm on x86 */ - y = f_roundl(xp); - - /* Intrinsic z^-1 delay */ - idx = (idx + 1) & DITHER_BUF_MASK; - state->e[idx] = y - xe; - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - z = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - z = (INT_MIN >> 8); - } else { - z = (int)y; - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - dst[0]=(char)(z>>16); - dst[1]=(char)(z>>8); - dst[2]=(char)(z); -#elif __BYTE_ORDER == __BIG_ENDIAN - dst[0]=(char)(z); - dst[1]=(char)(z>>8); - dst[2]=(char)(z>>16); -#endif - dst += dst_skip; - src++; - } - state->rm1 = rm1; - state->idx = idx; -} - -void sample_move_dither_shaped_d24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) -{ - jack_default_audio_sample_t x; - jack_default_audio_sample_t xe; /* the innput sample - filtered error */ - jack_default_audio_sample_t xp; /* x' */ - float r; - float rm1 = state->rm1; - unsigned int idx = state->idx; - int64_t y; - - while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; - r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; - /* Filter the error with Lipshitz's minimally audible FIR: - [2.033 -2.165 1.959 -1.590 0.6149] */ - xe = x - - state->e[idx] * 2.033f - + state->e[(idx - 1) & DITHER_BUF_MASK] * 2.165f - - state->e[(idx - 2) & DITHER_BUF_MASK] * 1.959f - + state->e[(idx - 3) & DITHER_BUF_MASK] * 1.590f - - state->e[(idx - 4) & DITHER_BUF_MASK] * 0.6149f; - xp = xe + r - rm1; - rm1 = r; - - /* This could be some inline asm on x86 */ - y = f_roundl(xp); - - /* Intrinsic z^-1 delay */ - idx = (idx + 1) & DITHER_BUF_MASK; - state->e[idx] = y - xe; - - y <<= 8; - - if (y > (INT_MAX >> 8)) { - y = (INT_MAX >> 8); - } else if (y < (INT_MIN >> 8)) { - y = (INT_MIN >> 8); - } -#if __BYTE_ORDER == __LITTLE_ENDIAN - memcpy (dst, &y, 3); -#elif __BYTE_ORDER == __BIG_ENDIAN - memcpy (dst, (char *)&y + 5, 3); -#endif - - dst += dst_skip; - src++; - } - state->rm1 = rm1; - state->idx = idx; -} void sample_move_d16_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { - int32_t tmp; + int16_t tmp; /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - tmp = f_round(*src * SAMPLE_MAX_16BIT); - if (tmp > SHRT_MAX) { - tmp = SHRT_MAX; - } else if (tmp < SHRT_MIN) { - tmp = SHRT_MIN; + if (*src <= NORMALIZED_FLOAT_MIN) { + tmp = SAMPLE_16BIT_MIN; + } else if (*src >= NORMALIZED_FLOAT_MAX) { + tmp = SAMPLE_16BIT_MAX; + } else { + tmp = (int16_t) f_round (*src * SAMPLE_16BIT_SCALING); } #if __BYTE_ORDER == __LITTLE_ENDIAN dst[0]=(char)(tmp>>8); @@ -793,18 +358,15 @@ void sample_move_d16_sSs (char *dst, jack_default_audio_sample_t *src, unsigned void sample_move_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { - int32_t tmp; - /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - tmp = f_round(*src * SAMPLE_MAX_16BIT); - if (tmp > SHRT_MAX) { - *((short *)dst) = SHRT_MAX; - } else if (tmp < SHRT_MIN) { - *((short *)dst) = SHRT_MIN; + if (*src <= NORMALIZED_FLOAT_MIN) { + *((int16_t*) dst) = SAMPLE_16BIT_MIN; + } else if (*src >= NORMALIZED_FLOAT_MAX) { + *((int16_t*) dst) = SAMPLE_16BIT_MAX; } else { - *((short *) dst) = (short) tmp; + *((int16_t*) dst) = (int16_t) f_round (*src * SAMPLE_16BIT_SCALING); } dst += dst_skip; src++; @@ -814,16 +376,16 @@ void sample_move_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned void sample_move_dither_rect_d16_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { jack_default_audio_sample_t val; - int tmp; + int16_t tmp; while (nsamples--) { - val = *src * (float)SAMPLE_MAX_16BIT; - val -= (float)fast_rand() / (float)INT_MAX; - tmp = f_round(val); - if (tmp > SHRT_MAX) { - tmp = SHRT_MAX; - } else if (tmp < SHRT_MIN) { - tmp = SHRT_MIN; + val = *src - (fast_rand()/((float)INT_MAX-1.0f)); + if (val <= NORMALIZED_FLOAT_MIN) { + tmp = SAMPLE_16BIT_MIN; + } else if (val >= NORMALIZED_FLOAT_MAX) { + tmp = SAMPLE_16BIT_MAX; + } else { + tmp = (int16_t) f_round(*src * SAMPLE_16BIT_SCALING); } #if __BYTE_ORDER == __LITTLE_ENDIAN dst[0]=(char)(tmp>>8); @@ -840,18 +402,16 @@ void sample_move_dither_rect_d16_sSs (char *dst, jack_default_audio_sample_t *s void sample_move_dither_rect_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state) { jack_default_audio_sample_t val; - int tmp; + int16_t tmp; while (nsamples--) { - val = *src * (float)SAMPLE_MAX_16BIT; - val -= (float)fast_rand() / (float)INT_MAX; - tmp = f_round(val); - if (tmp > SHRT_MAX) { - *((short *)dst) = SHRT_MAX; - } else if (tmp < SHRT_MIN) { - *((short *)dst) = SHRT_MIN; + val = *src - (fast_rand()/((float)INT_MAX - 1.0f)); + if (val <= NORMALIZED_FLOAT_MIN) { + *((int16_t*) dst) = SAMPLE_16BIT_MIN; + } else if (val >= NORMALIZED_FLOAT_MAX) { + *((int16_t*) dst) = SAMPLE_16BIT_MAX; } else { - *((short *) dst) = (short)tmp; + *((int16_t*) dst) = (int16_t) f_round(*src * SAMPLE_16BIT_SCALING); } dst += dst_skip; src++; @@ -863,20 +423,22 @@ void sample_move_dither_tri_d16_sSs (char *dst, jack_default_audio_sample_t *sr jack_default_audio_sample_t x; float r; float rm1 = state->rm1; - int y; + int16_t y; while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; + x = *src * SAMPLE_16BIT_SCALING; r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; x += r - rm1; rm1 = r; - y = f_round(x); - - if (y > SHRT_MAX) { - y = SHRT_MAX; - } else if (y < SHRT_MIN) { - y = SHRT_MIN; + + if (x <= NORMALIZED_FLOAT_MIN) { + y = SAMPLE_16BIT_MIN; + } else if (y >= NORMALIZED_FLOAT_MAX) { + y = SAMPLE_16BIT_MAX; + } else { + y = f_round(x); } + #if __BYTE_ORDER == __LITTLE_ENDIAN dst[0]=(char)(y>>8); dst[1]=(char)(y); @@ -898,7 +460,7 @@ void sample_move_dither_tri_d16_sS (char *dst, jack_default_audio_sample_t *src int y; while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; + x = *src * SAMPLE_16BIT_SCALING; r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; x += r - rm1; rm1 = r; @@ -929,7 +491,7 @@ void sample_move_dither_shaped_d16_sSs (char *dst, jack_default_audio_sample_t int y; while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; + x = *src * SAMPLE_16BIT_SCALING; r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; /* Filter the error with Lipshitz's minimally audible FIR: [2.033 -2.165 1.959 -1.590 0.6149] */ @@ -979,7 +541,7 @@ void sample_move_dither_shaped_d16_sS (char *dst, jack_default_audio_sample_t * int y; while (nsamples--) { - x = *src * (float)SAMPLE_MAX_16BIT; + x = *src * SAMPLE_16BIT_SCALING; r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f; /* Filter the error with Lipshitz's minimally audible FIR: [2.033 -2.165 1.959 -1.590 0.6149] */ @@ -1028,7 +590,7 @@ void sample_move_dS_s16s (jack_default_audio_sample_t *dst, char *src, unsigned z <<= 8; z |= (unsigned char)(src[0]); #endif - *dst = z / SAMPLE_MAX_16BIT; + *dst = z / SAMPLE_16BIT_SCALING; dst++; src += src_skip; } @@ -1039,7 +601,7 @@ void sample_move_dS_s16 (jack_default_audio_sample_t *dst, char *src, unsigned l { /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - *dst = (*((short *) src)) / SAMPLE_MAX_16BIT; + *dst = (*((short *) src)) / SAMPLE_16BIT_SCALING; dst++; src += src_skip; } @@ -1052,7 +614,7 @@ void sample_merge_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - val = (short) (*src * SAMPLE_MAX_16BIT); + val = (short) (*src * SAMPLE_16BIT_SCALING); if (val > SHRT_MAX - *((short *) dst)) { *((short *)dst) = SHRT_MAX; @@ -1071,7 +633,7 @@ void sample_merge_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsign /* ALERT: signed sign-extension portability !!! */ while (nsamples--) { - *((int *) dst) += (((int) (*src * SAMPLE_MAX_24BIT)) << 8); + *((int *) dst) += (((int) (*src * SAMPLE_24BIT_SCALING)) << 8); dst += dst_skip; src++; } |