osmocom-analog/src/libsample/sample.c

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/* Sample definition
*
* (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
* All Rights Reserved
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdint.h>
#include "sample.h"
/*
* A regular voice conversation takes place at this factor below the full range
* of 16 bits signed value:
*/
static double int_16_speech_level = SPEECH_LEVEL * 0.7079; /* 16 dBm below dBm0, which is about 3dBm below full 16 bit range */
static double int_16_1mw_level = 0.7079; /* dBm0, 3dBm below full 16 bit range */
/* A sample_t is a value that has virtually infinite precision but will also
* support high numbers. 'double' or 'float' types are sufficient.
*
* When using sample_t inside signal processing of each base station, the
* level of +- 1 is relative to the normal speech evenlope.
*
* When converting sample_t to int16_t, the level of +- 1 is reduced by factor.
* This way the speech may be louder before clipping happens.
*
* When using sample_t to modulate (SDR or sound card), the level is changed,
* so it represents the frequency deviation in Hz. The deviation of speech
* envelope is network dependent.
*/
/* sample conversion relative to SPEECH level */
void samples_to_int16_speech(int16_t *spl, sample_t *samples, int length)
{
int32_t value;
while (length--) {
value = *samples++ * int_16_speech_level * 32768.0;
if (value > 32767.0)
*spl++ = 32767;
else if (value < -32767.0)
*spl++ = -32767;
else
*spl++ = (uint16_t)value;
}
}
void int16_to_samples_speech(sample_t *samples, int16_t *spl, int length)
{
while (length--) {
*samples++ = (double)(*spl++) / 32767.0 / int_16_speech_level;
}
}
/* sample conversion relative to 1mW level */
void samples_to_int16_1mw(int16_t *spl, sample_t *samples, int length)
{
int32_t value;
while (length--) {
value = *samples++ * int_16_1mw_level * 32768.0;
if (value > 32767.0)
*spl++ = 32767;
else if (value < -32767.0)
*spl++ = -32767;
else
*spl++ = (uint16_t)value;
}
}
void int16_to_samples_1mw(sample_t *samples, int16_t *spl, int length)
{
while (length--) {
*samples++ = (double)(*spl++) / 32767.0 / int_16_1mw_level;
}
}