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/*
* Copyright (c) 2010 Clemens Ladisch <clemens@ladisch.de>
*
* Permission to use, copy, modify, and/or distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
/*
* The functions in this file map the value ranges of ALSA mixer controls onto
* the interval 0..1.
*
* The mapping is designed so that the position in the interval is proportional
* to the volume as a human ear would perceive it (i.e., the position is the
* cubic root of the linear sample multiplication factor). For controls with
* a small range (24 dB or less), the mapping is linear in the dB values so
* that each step has the same size visually. Only for controls without dB
* information, a linear mapping of the hardware volume register values is used
* (this is the same algorithm as used in the old alsamixer).
*
* When setting the volume, 'dir' is the rounding direction:
* -1/0/1 = down/nearest/up.
*/
#define _ISOC99_SOURCE /* lrint() */
#include "aconfig.h"
#include <math.h>
#include <stdbool.h>
#include "volume_mapping.h"
#define MAX_LINEAR_DB_SCALE 24
static inline bool use_linear_dB_scale(long dBmin, long dBmax)
{
return dBmax - dBmin <= MAX_LINEAR_DB_SCALE * 100;
}
static long lrint_dir(double x, int dir)
{
if (dir > 0)
return lrint(ceil(x));
else if (dir < 0)
return lrint(floor(x));
else
return lrint(x);
}
enum ctl_dir { PLAYBACK, CAPTURE };
static int (* const get_dB_range[2])(snd_mixer_elem_t *, long *, long *) = {
snd_mixer_selem_get_playback_dB_range,
snd_mixer_selem_get_capture_dB_range,
};
static int (* const get_raw_range[2])(snd_mixer_elem_t *, long *, long *) = {
snd_mixer_selem_get_playback_volume_range,
snd_mixer_selem_get_capture_volume_range,
};
static int (* const get_dB[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long *) = {
snd_mixer_selem_get_playback_dB,
snd_mixer_selem_get_capture_dB,
};
static int (* const get_raw[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long *) = {
snd_mixer_selem_get_playback_volume,
snd_mixer_selem_get_capture_volume,
};
static int (* const set_dB[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long, int) = {
snd_mixer_selem_set_playback_dB,
snd_mixer_selem_set_capture_dB,
};
static int (* const set_raw[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long) = {
snd_mixer_selem_set_playback_volume,
snd_mixer_selem_set_capture_volume,
};
static double get_normalized_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel,
enum ctl_dir ctl_dir)
{
long min, max, value;
double normalized, min_norm;
int err;
err = get_dB_range[ctl_dir](elem, &min, &max);
if (err < 0 || min >= max) {
err = get_raw_range[ctl_dir](elem, &min, &max);
if (err < 0 || min == max)
return 0;
err = get_raw[ctl_dir](elem, channel, &value);
if (err < 0)
return 0;
return (value - min) / (double)(max - min);
}
err = get_dB[ctl_dir](elem, channel, &value);
if (err < 0)
return 0;
if (use_linear_dB_scale(min, max))
return (value - min) / (double)(max - min);
normalized = pow(10, (value - max) / 6000.0);
if (min != SND_CTL_TLV_DB_GAIN_MUTE) {
min_norm = pow(10, (min - max) / 6000.0);
normalized = (normalized - min_norm) / (1 - min_norm);
}
return normalized;
}
static int set_normalized_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel,
double volume,
int dir,
enum ctl_dir ctl_dir)
{
long min, max, value;
double min_norm;
int err;
err = get_dB_range[ctl_dir](elem, &min, &max);
if (err < 0 || min >= max) {
err = get_raw_range[ctl_dir](elem, &min, &max);
if (err < 0)
return err;
value = lrint_dir(volume * (max - min), dir) + min;
return set_raw[ctl_dir](elem, channel, value);
}
if (use_linear_dB_scale(min, max)) {
value = lrint_dir(volume * (max - min), dir) + min;
return set_dB[ctl_dir](elem, channel, value, dir);
}
if (min != SND_CTL_TLV_DB_GAIN_MUTE) {
min_norm = pow(10, (min - max) / 6000.0);
volume = volume * (1 - min_norm) + min_norm;
}
if (volume <= 0)
volume = 1e-36;
value = lrint_dir(6000.0 * log10(volume), dir) + max;
return set_dB[ctl_dir](elem, channel, value, dir);
}
double get_normalized_playback_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel)
{
return get_normalized_volume(elem, channel, PLAYBACK);
}
double get_normalized_capture_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel)
{
return get_normalized_volume(elem, channel, CAPTURE);
}
int set_normalized_playback_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel,
double volume,
int dir)
{
return set_normalized_volume(elem, channel, volume, dir, PLAYBACK);
}
int set_normalized_capture_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel,
double volume,
int dir)
{
return set_normalized_volume(elem, channel, volume, dir, CAPTURE);
}
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