diff options
Diffstat (limited to 'audio/filter')
| -rw-r--r-- | audio/filter/af.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_drc.c | 8 | ||||
| -rw-r--r-- | audio/filter/af_export.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_hrtf.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_ladspa.c | 2 |
5 files changed, 8 insertions, 8 deletions
diff --git a/audio/filter/af.c b/audio/filter/af.c index 5a686e813c..e67fc29203 100644 --- a/audio/filter/af.c +++ b/audio/filter/af.c @@ -646,7 +646,7 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label, return NULL; new->label = talloc_strdup(new, label); - // Reinitalize the filter list + // Reinitialize the filter list if (af_reinit(s) != AF_OK) { af_remove_by_label(s, label); return NULL; diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c index 4344766349..472758c4c7 100644 --- a/audio/filter/af_drc.c +++ b/audio/filter/af_drc.c @@ -131,7 +131,7 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c) data[i] = tmp; } - // Evaulation of newavg (not 100% accurate because of values clamping) + // Evaluation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; // Stores computed values for future smoothing @@ -168,7 +168,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c) for (i = 0; i < len; i++) data[i] *= s->mul; - // Evaulation of newavg (not 100% accurate because of values clamping) + // Evaluation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; // Stores computed values for future smoothing @@ -216,7 +216,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c) data[i] = tmp; } - // Evaulation of newavg (not 100% accurate because of values clamping) + // Evaluation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; // Stores computed values for future smoothing @@ -262,7 +262,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c) for (i = 0; i < len; i++) data[i] *= s->mul; - // Evaulation of newavg (not 100% accurate because of values clamping) + // Evaluation of newavg (not 100% accurate because of values clamping) newavg = s->mul * curavg; // Stores computed values for future smoothing diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index f2613530e5..6020d9d98e 100644 --- a/audio/filter/af_export.c +++ b/audio/filter/af_export.c @@ -167,7 +167,7 @@ static int filter(struct af_instance *af, struct mp_audio *data) return 0; struct mp_audio* c = data; // Current working data af_export_t* s = af->priv; // Setup for this instance - int16_t* a = c->planes[0]; // Incomming sound + int16_t* a = c->planes[0]; // Incoming sound int nch = c->nch; // Number of channels int len = c->samples*c->nch; // Number of sample in data chunk int sz = s->sz; // buffer size (in samples) diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index 94a1599cd0..3c8a89c665 100644 --- a/audio/filter/af_hrtf.c +++ b/audio/filter/af_hrtf.c @@ -206,7 +206,7 @@ static inline void matrix_decode(short *in, const int k, const int il, information about Lt, Rt correlation. This effectively reshapes the front and rear "cones" to concentrate Lt + Rt to C and introduce Lt - Rt in L, R. */ - /* 0.67677 is the emprical lower bound for lpr_gain. */ + /* 0.67677 is the empirical lower bound for lpr_gain. */ c_gain = 8 * (*adapt_lpr_gain - 0.67677); c_gain = c_gain > 0 ? c_gain : 0; /* c_gain should not be too high, not even reaching full diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c index bd54dbb267..edde6a68b1 100644 --- a/audio/filter/af_ladspa.c +++ b/audio/filter/af_ladspa.c @@ -144,7 +144,7 @@ static int af_ladspa_parse_plugin(struct af_instance *af) { LADSPA_PortRangeHint hint; if (!setup->libhandle) - return AF_ERROR; /* only call parse after a succesful load */ + return AF_ERROR; /* only call parse after a successful load */ if (!setup->plugin_descriptor) return AF_ERROR; /* same as above */ |