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gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio.

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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
This commit is contained in:
Stefan Kost 2007-02-21 10:18:12 +00:00
parent 296687a398
commit 6e44a9c618
3 changed files with 140 additions and 67 deletions
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8
ChangeLog
View file

@ -1,3 +1,11 @@
2007-02-21 Stefan Kost <ensonic@users.sf.net>
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
2007-02-19 Stefan Kost <ensonic@users.sf.net>
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),

191
gst/level/gstlevel.c
View file

@ -78,6 +78,7 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstlevel.h"
@ -92,7 +93,7 @@ static const GstElementDetails level_details = GST_ELEMENT_DETAILS ("Level",
"Thomas Vander Stichele <thomas at apestaart dot org>");
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
@ -100,11 +101,16 @@ GST_STATIC_PAD_TEMPLATE ("sink",
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, " "signed = (boolean) true")
"depth = (int) { 8, 16 }, "
"signed = (boolean) true; "
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) {32, 64} ")
);
static GstStaticPadTemplate src_template_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
@ -112,7 +118,12 @@ GST_STATIC_PAD_TEMPLATE ("src",
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, " "signed = (boolean) true")
"depth = (int) { 8, 16 }, "
"signed = (boolean) true; "
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) {32, 64} ")
);
enum
@ -203,6 +214,8 @@ gst_level_init (GstLevel * filter, GstLevelClass * g_class)
filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
filter->message = TRUE;
filter->process = NULL;
}
static void
@ -277,6 +290,82 @@ gst_level_get_property (GObject * object, guint prop_id,
}
}
/* process one (interleaved) channel of incoming samples
* calculate square sum of samples
* normalize and average over number of samples
* returns a normalized cumulative square value, which can be averaged
* to return the average power 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
* 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
*
* for 16 bit, this code considers the non-existant 32768 value to be
* full-scale; so 32767 will not map to 1.0
*/
#define DEFINE_INT_LEVEL_CALCULATOR(TYPE, RESOLUTION) \
static void inline \
gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
gdouble *NCS, gdouble *NPS) \
{ \
TYPE * in = (TYPE *)data; \
register guint j; \
gdouble squaresum = 0.0; /* square sum of the integer samples */ \
register gdouble square = 0.0; /* Square */ \
register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
\
*NCS = 0.0; /* Normalized Cumulative Square */ \
*NPS = 0.0; /* Normalized Peask Square */ \
\
normalizer = (gdouble) (1 << (RESOLUTION * 2)); \
\
for (j = 0; j < num; j += channels) \
{ \
square = ((gdouble) in[j]) * in[j]; \
if (square > peaksquare) peaksquare = square; \
squaresum += square; \
} \
\
*NCS = squaresum / normalizer; \
*NPS = peaksquare / normalizer; \
}
DEFINE_INT_LEVEL_CALCULATOR (gint16, 15);
DEFINE_INT_LEVEL_CALCULATOR (gint8, 7);
#define DEFINE_FLOAT_LEVEL_CALCULATOR(TYPE) \
static void inline \
gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
gdouble *NCS, gdouble *NPS) \
{ \
TYPE * in = (TYPE *)data; \
register guint j; \
gdouble squaresum = 0.0; /* square sum of the integer samples */ \
register gdouble square = 0.0; /* Square */ \
register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
\
*NCS = 0.0; /* Normalized Cumulative Square */ \
*NPS = 0.0; /* Normalized Peask Square */ \
\
for (j = 0; j < num; j += channels) \
{ \
square = ((gdouble) in[j]) * in[j]; \
if (square > peaksquare) peaksquare = square; \
squaresum += square; \
} \
\
*NCS = squaresum; \
*NPS = peaksquare; \
}
DEFINE_FLOAT_LEVEL_CALCULATOR (gfloat);
DEFINE_FLOAT_LEVEL_CALCULATOR (gdouble);
static gint
structure_get_int (GstStructure * structure, const gchar * field)
{
@ -291,18 +380,42 @@ structure_get_int (GstStructure * structure, const gchar * field)
static gboolean
gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
GstLevel *filter;
GstLevel *filter = GST_LEVEL (trans);
const gchar *mimetype;
GstStructure *structure;
int i;
filter = GST_LEVEL (trans);
filter->num_frames = 0;
structure = gst_caps_get_structure (in, 0);
filter->rate = structure_get_int (structure, "rate");
filter->width = structure_get_int (structure, "width");
filter->channels = structure_get_int (structure, "channels");
mimetype = gst_structure_get_name (structure);
/* FIXME: set calculator func depending on caps */
filter->process = NULL;
if (strcmp (mimetype, "audio/x-raw-int") == 0) {
GST_DEBUG_OBJECT (filter, "use int: %u", filter->width);
switch (filter->width) {
case 8:
filter->process = gst_level_calculate_gint8;
break;
case 16:
filter->process = gst_level_calculate_gint16;
break;
}
} else if (strcmp (mimetype, "audio/x-raw-float") == 0) {
GST_DEBUG_OBJECT (filter, "use float, %u", filter->width);
switch (filter->width) {
case 32:
filter->process = gst_level_calculate_gfloat;
break;
case 64:
filter->process = gst_level_calculate_gdouble;
break;
}
}
/* allocate channel variable arrays */
g_free (filter->CS);
@ -328,52 +441,6 @@ gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
return TRUE;
}
/* process one (interleaved) channel of incoming samples
* calculate square sum of samples
* normalize and average over number of samples
* returns a normalized cumulative square value, which can be averaged
* to return the average power 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
* 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
*
* for 16 bit, this code considers the non-existant 32768 value to be
* full-scale; so 32767 will not map to 1.0
*/
#define DEFINE_LEVEL_CALCULATOR(TYPE, RESOLUTION) \
static void inline \
gst_level_calculate_##TYPE (TYPE * in, guint num, gint channels, \
double *NCS, double *NPS) \
{ \
register int j; \
double squaresum = 0.0; /* square sum of the integer samples */ \
register double square = 0.0; /* Square */ \
register double peaksquare = 0.0; /* Peak Square Sample */ \
gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
\
*NCS = 0.0; /* Normalized Cumulative Square */ \
*NPS = 0.0; /* Normalized Peask Square */ \
\
normalizer = (double) (1 << (RESOLUTION * 2)); \
\
for (j = 0; j < num; j += channels) \
{ \
square = ((double) in[j]) * in[j]; \
if (square > peaksquare) peaksquare = square; \
squaresum += square; \
} \
\
*NCS = squaresum / normalizer; \
*NPS = peaksquare / normalizer; \
}
DEFINE_LEVEL_CALCULATOR (gint16, 15);
DEFINE_LEVEL_CALCULATOR (gint8, 7);
static GstMessage *
gst_level_message_new (GstLevel * l, GstClockTime endtime)
{
@ -427,10 +494,10 @@ gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
GstLevel *filter;
gpointer in_data;
double CS = 0.0;
gint num_frames = 0;
gint num_int_samples = 0; /* number of interleaved samples
guint num_frames = 0;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
gint i;
guint i;
filter = GST_LEVEL (trans);
@ -447,16 +514,8 @@ gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
for (i = 0; i < filter->channels; ++i) {
CS = 0.0;
switch (filter->width) {
case 16:
gst_level_calculate_gint16 (((gint16 *) in_data) + i, num_int_samples,
filter->channels, &CS, &filter->peak[i]);
break;
case 8:
gst_level_calculate_gint8 (((gint8 *) in_data) + i, num_int_samples,
filter->channels, &CS, &filter->peak[i]);
break;
}
filter->process (in_data + i, num_int_samples, filter->channels, &CS,
&filter->peak[i]);
GST_LOG_OBJECT (filter,
"channel %d, cumulative sum %f, peak %f, over %d samples/%d channels",
i, CS, filter->peak[i], num_int_samples, filter->channels);

8
gst/level/gstlevel.h
View file

@ -48,7 +48,11 @@ G_BEGIN_DECLS
typedef struct _GstLevel GstLevel;
typedef struct _GstLevelClass GstLevelClass;
/**
* GstLevel:
*
* Opaque data structure.
*/
struct _GstLevel {
GstBaseTransform element;
@ -73,6 +77,8 @@ struct _GstLevel {
gdouble *MS; /* normalized Mean Square of buffer */
gdouble *RMS_dB; /* RMS in dB to emit */
GstClockTime *decay_peak_age; /* age of last peak */
void (*process)(gpointer, guint, guint, gdouble*, gdouble*);
};
struct _GstLevelClass {