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authorarcnmx <arcnmx@users.noreply.github.com>2018-02-21 13:25:08 -0500
committerOrestis <orestisf1993@gmail.com>2018-10-11 00:21:03 +0300
commitddadc8e4d728923cf5bc7c8a894b103cf061c47d (patch)
treef278c0a2e647b589d2f3892881ef1785b105705f
parent494efd49a26ed96d7b0d3d4f69099ccd83e2ccba (diff)
Provide a more natural volume percentage with ALSA.
The rationale of the code is explained in the header: http://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=alsamixer/volume_mapping.c;h=1c0d7c45e6686239464e1b0bbc8983ea57f3914f;hb=HEAD > 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). and the commit message: http://git.alsa-project.org/?p=alsa-utils.git;a=commit;h=34bb514b5fd1d6f91ba9a7b3a70b0ea0c6014250 > use a mapping where the bar height is proportional to the audible > volume, i.e., where the amplitude is the cube of the bar height. and further explanation can be found in the pull request: https://github.com/i3/i3status/pull/268#pullrequestreview-147429763
-rw-r--r--src/print_volume.c35
1 files changed, 28 insertions, 7 deletions
diff --git a/src/print_volume.c b/src/print_volume.c
index 4c0fbde..c3180fe 100644
--- a/src/print_volume.c
+++ b/src/print_volume.c
@@ -11,6 +11,7 @@
#ifdef LINUX
#include <alsa/asoundlib.h>
#include <alloca.h>
+#include <math.h>
#endif
#if defined(__FreeBSD__) || defined(__DragonFly__)
@@ -111,11 +112,13 @@ void print_volume(yajl_gen json_gen, char *buffer, const char *fmt, const char *
#endif
#ifdef LINUX
+ const long MAX_LINEAR_DB_SCALE = 24;
int err;
snd_mixer_t *m;
snd_mixer_selem_id_t *sid;
snd_mixer_elem_t *elem;
long min, max, val;
+ bool force_linear = false;
int avg;
if ((err = snd_mixer_open(&m, 0)) < 0) {
@@ -161,16 +164,34 @@ void print_volume(yajl_gen json_gen, char *buffer, const char *fmt, const char *
}
/* Get the volume range to convert the volume later */
- snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
-
snd_mixer_handle_events(m);
- snd_mixer_selem_get_playback_volume(elem, 0, &val);
- if (max != 100) {
- float avgf = ((float)val / max) * 100;
+ err = snd_mixer_selem_get_playback_dB_range(elem, &min, &max) ||
+ snd_mixer_selem_get_playback_dB(elem, 0, &val);
+ if (err != 0 || min >= max) {
+ err = snd_mixer_selem_get_playback_volume_range(elem, &min, &max) ||
+ snd_mixer_selem_get_playback_volume(elem, 0, &val);
+ force_linear = true;
+ }
+
+ if (err != 0) {
+ fprintf(stderr, "i3status: ALSA: Cannot get playback volume.\n");
+ goto out;
+ }
+
+ /* Use linear mapping for raw register values or small ranges of 24 dB */
+ if (force_linear || max - min <= MAX_LINEAR_DB_SCALE * 100) {
+ float avgf = ((float)(val - min) / (max - min)) * 100;
avg = (int)avgf;
avg = (avgf - avg < 0.5 ? avg : (avg + 1));
- } else
- avg = (int)val;
+ } else {
+ /* mapped volume to be more natural for the human ear */
+ double normalized = exp10((val - max) / 6000.0);
+ if (min != SND_CTL_TLV_DB_GAIN_MUTE) {
+ double min_norm = exp10((min - max) / 6000.0);
+ normalized = (normalized - min_norm) / (1 - min_norm);
+ }
+ avg = lround(normalized * 100);
+ }
/* Check for mute */
if (snd_mixer_selem_has_playback_switch(elem)) {