port fixes from 0.8 to level

Original commit message from CVS:
port fixes from 0.8 to level
This commit is contained in:
Thomas Vander Stichele 2005-08-10 10:54:02 +00:00
parent 1d185d4127
commit bd57e8657c
4 changed files with 87 additions and 68 deletions

View file

@ -1,3 +1,16 @@
2005-08-10 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform):
* gst/level/gstlevel.h:
remove unused MS struct member
don't reset the CS values for channels on every _chain, so that
level actually correctly calculates the RMS value. sigh.
calculate RMS values correctly for peak and decay peak sums;
before we were signalling them as if they already were amplitude
and not power values. sigh.
Remind me to not try and pretend I'm writing DSP code.
2005-08-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net> 2005-08-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps): * ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):

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@ -139,7 +139,6 @@ gst_level_init (GstLevel * filter)
{ {
filter->CS = NULL; filter->CS = NULL;
filter->peak = NULL; filter->peak = NULL;
filter->MS = NULL;
filter->RMS_dB = NULL; filter->RMS_dB = NULL;
filter->rate = 0; filter->rate = 0;
@ -233,70 +232,59 @@ gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
g_free (filter->last_peak); g_free (filter->last_peak);
g_free (filter->decay_peak); g_free (filter->decay_peak);
g_free (filter->decay_peak_age); g_free (filter->decay_peak_age);
g_free (filter->MS);
g_free (filter->RMS_dB); g_free (filter->RMS_dB);
filter->CS = g_new (double, filter->channels); filter->CS = g_new (double, filter->channels);
filter->peak = g_new (double, filter->channels); filter->peak = g_new (double, filter->channels);
filter->last_peak = g_new (double, filter->channels); filter->last_peak = g_new (double, filter->channels);
filter->decay_peak = g_new (double, filter->channels); filter->decay_peak = g_new (double, filter->channels);
filter->decay_peak_age = g_new (double, filter->channels); filter->decay_peak_age = g_new (double, filter->channels);
filter->MS = g_new (double, filter->channels);
filter->RMS_dB = g_new (double, filter->channels); filter->RMS_dB = g_new (double, filter->channels);
for (i = 0; i < filter->channels; ++i) { for (i = 0; i < filter->channels; ++i) {
filter->CS[i] = filter->peak[i] = filter->last_peak[i] = filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
filter->decay_peak[i] = filter->decay_peak_age[i] = filter->decay_peak[i] = filter->decay_peak_age[i] =
filter->MS[i] = filter->RMS_dB[i] = 0.0; filter->RMS_dB[i] = 0.0;
} }
return TRUE; return TRUE;
} }
#if 0
#define DEBUG(str,...) g_print (str, ...)
#else
#define DEBUG(str,...) /*nop */
#endif
/* process one (interleaved) channel of incoming samples /* process one (interleaved) channel of incoming samples
* calculate square sum of samples * calculate square sum of samples
* normalize and return normalized Cumulative Square * normalize and average over number of samples
* returns a normalized average power value as CS, as a double between 0 and 1
* also returns the normalized peak power (square of the highest amplitude)
*
* caller must assure num is a multiple of channels * caller must assure num is a multiple of channels
* samples for multiple channels are interleaved
* input sample data enters in *in_data as 8 or 16 bit data
* this filter only accepts signed audio data, so mid level is always 0 * this filter only accepts signed audio data, so mid level is always 0
*/ */
#define DEFINE_LEVEL_CALCULATOR(TYPE) \
static void inline \ #define DEFINE_LEVEL_CALCULATOR(TYPE) \
gst_level_calculate_##TYPE (TYPE * in, guint num, gint channels, \ static void inline \
gint resolution, double *CS, double *peak) \ gst_level_calculate_##TYPE (TYPE * in, guint num, gint channels, \
{ \ gint resolution, double *CS, double *peak) \
register int j; \ { \
double squaresum = 0.0; /* square sum of the integer samples */ \ register int j; \
register double square = 0.0; /* Square */ \ double squaresum = 0.0; /* square sum of the integer samples */ \
register double PSS = 0.0; /* Peak Square Sample */ \ register double square = 0.0; /* Square */ \
gdouble normalizer; \ register double PSS = 0.0; /* Peak Square Sample */ \
\ gdouble normalizer; /* divisor to get a [-1, - 1] range */ \
*CS = 0.0; /* Cumulative Square for this block */ \ \
\ *CS = 0.0; /* Cumulative Square for this block */ \
normalizer = (double) (1 << resolution); \ \
\ normalizer = (double) (1 << resolution); \
/* \ \
* process data here \ for (j = 0; j < num; j += channels) \
* input sample data enters in *in_data as 8 or 16 bit data \ { \
* samples for left and right channel are interleaved \ square = ((double) in[j]) * in[j]; \
* returns the Mean Square of the samples as a double between 0 and 1 \ if (square > PSS) PSS = square; \
*/ \ squaresum += square; \
\ } \
for (j = 0; j < num; j += channels) \ \
{ \ *CS = squaresum / (normalizer * normalizer); \
DEBUG ("ch %d -> smp %d\n", j, in[j]); \ *peak = PSS / (normalizer * normalizer); \
square = (double) (in[j] * in[j]); \
if (square > PSS) PSS = square; \
squaresum += square; \
} \
*peak = PSS / ((double) normalizer * (double) normalizer); \
\
/* return normalized cumulative square */ \
*CS = squaresum / ((double) normalizer * (double) normalizer); \
} }
DEFINE_LEVEL_CALCULATOR (gint16); DEFINE_LEVEL_CALCULATOR (gint16);
@ -350,49 +338,52 @@ gst_level_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
GstLevel *filter; GstLevel *filter;
gpointer in_data; gpointer in_data;
double CS = 0.0; double CS = 0.0;
gint num_samples = 0; gint num_int_samples = 0; /* number of samples for all channels combined */
gint i; gint i;
filter = GST_LEVEL (trans); filter = GST_LEVEL (trans);
for (i = 0; i < filter->channels; ++i) for (i = 0; i < filter->channels; ++i)
filter->CS[i] = filter->peak[i] = filter->MS[i] = filter->RMS_dB[i] = 0.0; filter->peak[i] = filter->RMS_dB[i] = 0.0;
in_data = GST_BUFFER_DATA (in); in_data = GST_BUFFER_DATA (in);
num_samples = GST_BUFFER_SIZE (in) / (filter->width / 8); num_int_samples = GST_BUFFER_SIZE (in) / (filter->width / 8);
g_return_val_if_fail (num_samples % filter->channels == 0, GST_FLOW_ERROR); g_return_val_if_fail (num_int_samples % filter->channels == 0,
GST_FLOW_ERROR);
for (i = 0; i < filter->channels; ++i) { for (i = 0; i < filter->channels; ++i) {
CS = 0.0;
switch (filter->width) { switch (filter->width) {
case 16: case 16:
gst_level_calculate_gint16 (in_data + i, num_samples, gst_level_calculate_gint16 (in_data + i, num_int_samples,
filter->channels, filter->width - 1, &CS, &filter->peak[i]); filter->channels, filter->width - 1, &CS, &filter->peak[i]);
break; break;
case 8: case 8:
gst_level_calculate_gint8 (((gint8 *) in_data) + i, num_samples, gst_level_calculate_gint8 (((gint8 *) in_data) + i, num_int_samples,
filter->channels, filter->width - 1, &CS, &filter->peak[i]); filter->channels, filter->width - 1, &CS, &filter->peak[i]);
break; break;
} }
GST_LOG_OBJECT (filter, "channel %d, cumulative sum %f, peak %f", i, CS, GST_LOG_OBJECT (filter,
filter->peak[i]); "channel %d, cumulative sum %f, peak %f, over %d channels/%d samples",
i, CS, filter->peak[i], num_int_samples, filter->channels);
filter->CS[i] += CS; filter->CS[i] += CS;
} }
filter->num_samples += num_samples; filter->num_samples += num_int_samples / filter->channels;
for (i = 0; i < filter->channels; ++i) { for (i = 0; i < filter->channels; ++i) {
filter->decay_peak_age[i] += num_samples; filter->decay_peak_age[i] += num_int_samples / filter->channels;
DEBUG ("filter peak info [%d]: peak %f, age %f\n", i, GST_LOG_OBJECT (filter, "filter peak info [%d]: peak %f, age %f\n", i,
filter->last_peak[i], filter->decay_peak_age[i]); filter->last_peak[i], filter->decay_peak_age[i]);
/* update running peak */ /* update running peak */
if (filter->peak[i] > filter->last_peak[i]) if (filter->peak[i] > filter->last_peak[i])
filter->last_peak[i] = filter->peak[i]; filter->last_peak[i] = filter->peak[i];
/* update decay peak */ /* update decay peak */
if (filter->peak[i] >= filter->decay_peak[i]) { if (filter->peak[i] >= filter->decay_peak[i]) {
DEBUG ("new peak, %f\n", filter->peak[i]); GST_LOG_OBJECT (filter, "new peak, %f\n", filter->peak[i]);
filter->decay_peak[i] = filter->peak[i]; filter->decay_peak[i] = filter->peak[i];
filter->decay_peak_age[i] = 0; filter->decay_peak_age[i] = 0;
} else { } else {
@ -403,15 +394,19 @@ gst_level_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
double length; /* length of buffer in seconds */ double length; /* length of buffer in seconds */
length = (double) num_samples / (filter->channels * filter->rate); length = (double) num_int_samples / (filter->channels * filter->rate);
falloff_dB = filter->decay_peak_falloff * length; falloff_dB = filter->decay_peak_falloff * length;
falloff = pow (10, falloff_dB / -20.0); falloff = pow (10, falloff_dB / -20.0);
DEBUG ("falloff: length %f, dB falloff %f, falloff factor %e\n", GST_LOG_OBJECT (filter,
"falloff: length %f, dB falloff %f, falloff factor %e\n",
length, falloff_dB, falloff); length, falloff_dB, falloff);
filter->decay_peak[i] *= falloff; filter->decay_peak[i] *= falloff;
DEBUG ("peak is %f samples old, decayed with factor %e to %f\n", GST_LOG_OBJECT (filter,
"peak is %f samples old, decayed with factor %e to %f\n",
filter->decay_peak_age[i], falloff, filter->decay_peak[i]); filter->decay_peak_age[i], falloff, filter->decay_peak[i]);
} else {
GST_LOG_OBJECT (filter, "peak not old enough, not decaying");
} }
} }
} }
@ -422,18 +417,30 @@ gst_level_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
if (filter->signal) { if (filter->signal) {
GstMessage *m; GstMessage *m;
double endtime, RMS; double endtime, RMS;
double RMSdB, lastdB, decaydB;
/* FIXME: convert to a GstClockTime instead */
endtime = (double) GST_BUFFER_TIMESTAMP (in) / GST_SECOND endtime = (double) GST_BUFFER_TIMESTAMP (in) / GST_SECOND
+ (double) num_samples / (double) filter->rate; + (double) num_int_samples / (filter->rate * filter->channels);
m = gst_level_message_new (filter, endtime); m = gst_level_message_new (filter, endtime);
for (i = 0; i < filter->channels; ++i) { for (i = 0; i < filter->channels; ++i) {
RMS = sqrt (filter->CS[i] / (filter->num_samples / filter->channels)); RMS = sqrt (filter->CS[i] / filter->num_samples);
GST_LOG_OBJECT (filter,
"CS: %f, num_samples %f, channel %d, RMS %f",
filter->CS[i], filter->num_samples, i, RMS);
/* RMS values are calculated in amplitude, so 20 * log 10 */
RMSdB = 20 * log10 (RMS);
/* peak values are square sums, ie. power, so 10 * log 10 */
lastdB = 10 * log10 (filter->last_peak[i]);
decaydB = 10 * log10 (filter->decay_peak[i]);
gst_level_message_append_channel (m, 20 * log10 (RMS), GST_LOG_OBJECT (filter,
20 * log10 (filter->last_peak[i]), "time %f, channel %d, RMS %f dB, peak %f dB, decay %f dB",
20 * log10 (filter->decay_peak[i])); endtime, i, RMSdB, lastdB, decaydB);
gst_level_message_append_channel (m, RMSdB, lastdB, decaydB);
/* reset cumulative and normal peak */ /* reset cumulative and normal peak */
filter->CS[i] = 0.0; filter->CS[i] = 0.0;

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@ -61,7 +61,7 @@ struct _GstLevel {
gdouble decay_peak_ttl; /* time to live for peak in seconds */ gdouble decay_peak_ttl; /* time to live for peak in seconds */
gdouble decay_peak_falloff; /* falloff in dB/sec */ gdouble decay_peak_falloff; /* falloff in dB/sec */
gdouble num_samples; /* cumulative sample count */ gdouble num_samples; /* one-channel sample count since last emit */
/* per-channel arrays for intermediate values */ /* per-channel arrays for intermediate values */
gdouble *CS; /* normalized Cumulative Square */ gdouble *CS; /* normalized Cumulative Square */
@ -70,7 +70,7 @@ struct _GstLevel {
gdouble *decay_peak; /* running decaying normalized Peak */ gdouble *decay_peak; /* running decaying normalized Peak */
gdouble *MS; /* normalized Mean Square of buffer */ gdouble *MS; /* normalized Mean Square of buffer */
gdouble *RMS_dB; /* RMS in dB to emit */ gdouble *RMS_dB; /* RMS in dB to emit */
gdouble *decay_peak_age; /* age of last peak */ gdouble *decay_peak_age; /* age of last peak in one-channel samples */
}; };
struct _GstLevelClass { struct _GstLevelClass {

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@ -1,4 +1,3 @@
plugin_LTLIBRARIES = libgstrtp.la plugin_LTLIBRARIES = libgstrtp.la
libgstrtp_la_SOURCES = gstrtp.c gstrtpdec.c libgstrtp_la_SOURCES = gstrtp.c gstrtpdec.c