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virtual-surround-sink: Move normalization heuristic to its own function

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This patch also adds a description how the heuristic works and mentions that
there is a scaling factor that can be adjusted if there is audible clipping.
This commit is contained in:
Niels Ole Salscheider 2014-04-02 22:32:34 +02:00 • committed by Tanu Kaskinen
parent eca082a93f
commit 3064511df6
1 changed files with 40 additions and 17 deletions
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57
src/modules/module-virtual-surround-sink.c
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@ -519,6 +519,45 @@ static pa_channel_position_t mirror_channel(pa_channel_position_t channel) {
}
}
static void normalize_hrir(struct userdata *u) {
/* normalize hrir to avoid audible clipping
*
* The following heuristic tries to avoid audible clipping. It cannot avoid
* clipping in the worst case though, because the scaling factor would
* become too large resulting in a too quiet signal.
* The idea of the heuristic is to avoid clipping when a single click is
* played back on all channels. The scaling factor describes the additional
* factor that is necessary to avoid clipping for "normal" signals.
*
* This algorithm doesn't pretend to be perfect, it's just something that
* appears to work (not too quiet, no audible clipping) on the material that
* it has been tested on. If you find a real-world example where this
* algorithm results in audible clipping, please write a patch that adjusts
* the scaling factor constants or improves the algorithm (or if you can't
* write a patch, at least report the problem to the PulseAudio mailing list
* or bug tracker). */
const float scaling_factor = 2.5;
float hrir_sum, hrir_max;
unsigned i, j;
hrir_max = 0;
for (i = 0; i < u->hrir_samples; i++) {
hrir_sum = 0;
for (j = 0; j < u->hrir_channels; j++)
hrir_sum += fabs(u->hrir_data[i * u->hrir_channels + j]);
if (hrir_sum > hrir_max)
hrir_max = hrir_sum;
}
for (i = 0; i < u->hrir_samples; i++) {
for (j = 0; j < u->hrir_channels; j++)
u->hrir_data[i * u->hrir_channels + j] /= hrir_max * scaling_factor;
}
}
int pa__init(pa_module*m) {
struct userdata *u;
pa_sample_spec ss, sink_input_ss;
@ -533,7 +572,6 @@ int pa__init(pa_module*m) {
const char *hrir_file;
unsigned i, j, found_channel_left, found_channel_right;
float hrir_sum, hrir_max;
float *hrir_data;
pa_sample_spec hrir_ss;
@ -758,22 +796,7 @@ int pa__init(pa_module*m) {
goto fail;
}
/* normalize hrir to avoid clipping */
hrir_max = 0;
for (i = 0; i < u->hrir_samples; i++) {
hrir_sum = 0;
for (j = 0; j < u->hrir_channels; j++)
hrir_sum += fabs(u->hrir_data[i * u->hrir_channels + j]);
if (hrir_sum > hrir_max)
hrir_max = hrir_sum;
}
if (hrir_max > 1) {
for (i = 0; i < u->hrir_samples; i++) {
for (j = 0; j < u->hrir_channels; j++)
u->hrir_data[i * u->hrir_channels + j] /= hrir_max * 1.2;
}
}
normalize_hrir(u);
/* create mapping between hrir and input */
u->mapping_left = (unsigned *) pa_xnew0(unsigned, u->channels);