Mercurial > hg > octave-jordi
changeset 19556:182abae666ae
* audiodevinfo.cc: Explicitly use std::pow.
author | John W. Eaton <jwe@octave.org> |
---|---|
date | Fri, 09 Jan 2015 13:29:03 -0500 |
parents | d7057df2968e |
children | 0b068273340f |
files | libinterp/dldfcn/audiodevinfo.cc |
diffstat | 1 files changed, 26 insertions(+), 17 deletions(-) [+] |
line wrap: on
line diff
--- a/libinterp/dldfcn/audiodevinfo.cc +++ b/libinterp/dldfcn/audiodevinfo.cc @@ -564,7 +564,7 @@ { case 8: { - static double scale_factor = pow (2.0, 7) - 1.0; + static double scale_factor = std::pow (2.0, 7) - 1.0; int8_t *buffer = static_cast<int8_t *> (output); @@ -578,7 +578,7 @@ case 16: { - static double scale_factor = pow (2.0, 15) - 1.0; + static double scale_factor = std::pow (2.0, 15) - 1.0; int16_t *buffer = static_cast<int16_t *> (output); @@ -592,7 +592,7 @@ case 24: { - static double scale_factor = pow (2.0, 23) - 1.0; + static double scale_factor = std::pow (2.0, 23) - 1.0; static int big_endian = oct_mach_info::words_big_endian (); @@ -662,7 +662,7 @@ { case 8: { - static double scale_factor = pow (2.0, 7) - 1.0; + static double scale_factor = std::pow (2.0, 7) - 1.0; int8_t *buffer = static_cast<int8_t *> (output); @@ -685,7 +685,7 @@ case 16: { - static double scale_factor = pow (2.0, 15) - 1.0; + static double scale_factor = std::pow (2.0, 15) - 1.0; int16_t *buffer = static_cast<int16_t *> (output); @@ -708,7 +708,7 @@ case 24: { - static double scale_factor = pow (2.0, 23) - 1.0; + static double scale_factor = std::pow (2.0, 23) - 1.0; static int big_endian = oct_mach_info::words_big_endian (); @@ -1319,8 +1319,10 @@ const int8_t *input8 = static_cast<const int8_t *> (input); for (unsigned long i = 0; i < frames; i++) { - sample_l = input8[i * channels] / (pow (2.0, 7) - 1.0); - sample_r = input8[i * channels + (channels - 1)] / (pow (2.0, 7) - 1.0); + sample_l = input8[i * channels] / (std::pow (2.0, 7) - 1.0); + sample_r = input8[i * channels + (channels - 1)] + / (std::pow (2.0, 7) - 1.0); + sound(i, 0) = sample_l; sound(i, 1) = sample_r; } @@ -1330,8 +1332,10 @@ const int16_t *input16 = static_cast<const int16_t *> (input); for (unsigned long i = 0; i < frames; i++) { - sample_l = input16[i * channels] / (pow (2.0, 15) - 1.0); - sample_r = input16[i * channels + (channels - 1)] / (pow (2.0, 15) - 1.0); + sample_l = input16[i * channels] / (std::pow (2.0, 15) - 1.0); + sample_r = input16[i * channels + (channels - 1)] + / (std::pow (2.0, 15) - 1.0); + sound(i, 0) = sample_l; sound(i, 1) = sample_r; } @@ -1359,8 +1363,8 @@ if (sample_r32 & 0x00800000) sample_r32 |= 0xff000000; - sound(i, 0) = sample_l32 / pow (2.0, 23); - sound(i, 1) = sample_r32 / pow (2.0, 23); + sound(i, 0) = sample_l32 / std::pow (2.0, 23); + sound(i, 1) = sample_r32 / std::pow (2.0, 23); } } @@ -1384,8 +1388,10 @@ const int8_t *input8 = static_cast<const int8_t *> (input); for (unsigned long i = 0; i < frames; i++) { - sample_l = input8[i * channels] / (pow (2.0, 7) - 1.0); - sample_r = input8[i * channels + (channels - 1)] / (pow (2.0, 7) - 1.0); + sample_l = input8[i * channels] / (std::pow (2.0, 7) - 1.0); + sample_r = input8[i * channels + (channels - 1)] + / (std::pow (2.0, 7) - 1.0); + recorder->append (sample_l, sample_r); } } @@ -1394,8 +1400,10 @@ const int16_t *input16 = static_cast<const int16_t *> (input); for (unsigned long i = 0; i < frames; i++) { - sample_l = input16[i * channels] / (pow (2.0, 15) - 1.0); - sample_r = input16[i * channels + (channels - 1)] / (pow (2.0, 15) - 1.0); + sample_l = input16[i * channels] / (std::pow (2.0, 15) - 1.0); + sample_r = input16[i * channels + (channels - 1)] + / (std::pow (2.0, 15) - 1.0); + recorder->append (sample_l, sample_r); } } @@ -1422,7 +1430,8 @@ if (sample_r32 & 0x00800000) sample_r32 |= 0xff000000; - recorder->append (sample_l32 / pow (2.0, 23), sample_r32 / pow (2.0, 23)); + recorder->append (sample_l32 / std::pow (2.0, 23), + sample_r32 / std::pow (2.0, 23)); } }