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gstreamer/gst/level/gstlevel.c
Wim Taymans 80dc806b65 gst/level/gstlevel.*: Use basetransform segment so that it is correctly managed on flushes and start/stop. 
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
2007-09-13 17:31:16 +00:00

656 lines
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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2000,2001,2002,2003,2005
* Thomas Vander Stichele <thomas at apestaart dot org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-level
*
* <refsect2>
* <para>
* Level analyses incoming audio buffers and, if the
* <link linkend="GstLevel--message">message property</link> is #TRUE,
* generates an element message named
* <classname>&quot;level&quot;</classname>:
* after each interval of time given by the
* <link linkend="GstLevel--interval">interval property</link>.
* The message's structure contains four fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;endtime&quot;</classname>:
* the end time of the buffer that triggered the message
* </para>
* </listitem>
* <listitem>
* <para>
* #GstValueList of #gdouble
* <classname>&quot;peak&quot;</classname>:
* the peak power level in dB for each channel
* </para>
* </listitem>
* <listitem>
* <para>
* #GstValueList of #gdouble
* <classname>&quot;decay&quot;</classname>:
* the decaying peak power level in dB for each channel
* the decaying peak level follows the peak level, but starts dropping
* if no new peak is reached after the time given by
* the <link linkend="GstLevel--peak-ttl">the time to live</link>.
* When the decaying peak level drops, it does so at the decay rate
* as specified by the
* <link linkend="GstLevel--peak-falloff">the peak falloff rate</link>.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstValueList of #gdouble
* <classname>&quot;rms&quot;</classname>:
* the Root Mean Square (or average power) level in dB for each channel
* </para>
* </listitem>
* </itemizedlist>
* </para>
* <title>Example application</title>
* <para>
* <include xmlns="http://www.w3.org/2003/XInclude" href="element-level-example.xml" />
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstlevel.h"
#include "math.h"
GST_DEBUG_CATEGORY_STATIC (level_debug);
#define GST_CAT_DEFAULT level_debug
static const GstElementDetails level_details = GST_ELEMENT_DETAILS ("Level",
"Filter/Analyzer/Audio",
"RMS/Peak/Decaying Peak Level messager for audio/raw",
"Thomas Vander Stichele <thomas at apestaart dot org>");
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"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_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"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
{
PROP_0,
PROP_SIGNAL_LEVEL,
PROP_SIGNAL_INTERVAL,
PROP_PEAK_TTL,
PROP_PEAK_FALLOFF
};
GST_BOILERPLATE (GstLevel, gst_level, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM);
static void gst_level_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_level_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_level_dispose (GObject * obj);
static gboolean gst_level_set_caps (GstBaseTransform * trans, GstCaps * in,
GstCaps * out);
static gboolean gst_level_start (GstBaseTransform * trans);
static GstFlowReturn gst_level_transform_ip (GstBaseTransform * trans,
GstBuffer * in);
static void
gst_level_base_init (gpointer g_class)
{
GstElementClass *element_class = g_class;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template_factory));
gst_element_class_set_details (element_class, &level_details);
}
static void
gst_level_class_init (GstLevelClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
gobject_class->set_property = gst_level_set_property;
gobject_class->get_property = gst_level_get_property;
gobject_class->dispose = gst_level_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SIGNAL_LEVEL,
g_param_spec_boolean ("message", "mesage",
"Post a level message for each passed interval",
TRUE, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SIGNAL_INTERVAL,
g_param_spec_uint64 ("interval", "Interval",
"Interval of time between message posts (in nanoseconds)",
1, G_MAXUINT64, GST_SECOND / 10, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PEAK_TTL,
g_param_spec_uint64 ("peak-ttl", "Peak TTL",
"Time To Live of decay peak before it falls back (in nanoseconds)",
0, G_MAXUINT64, GST_SECOND / 10 * 3, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PEAK_FALLOFF,
g_param_spec_double ("peak-falloff", "Peak Falloff",
"Decay rate of decay peak after TTL (in dB/sec)",
0.0, G_MAXDOUBLE, 10.0, G_PARAM_READWRITE));
GST_DEBUG_CATEGORY_INIT (level_debug, "level", 0, "Level calculation");
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_level_set_caps);
trans_class->start = GST_DEBUG_FUNCPTR (gst_level_start);
trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_level_transform_ip);
trans_class->passthrough_on_same_caps = TRUE;
}
static void
gst_level_init (GstLevel * filter, GstLevelClass * g_class)
{
filter->CS = NULL;
filter->peak = NULL;
filter->rate = 0;
filter->width = 0;
filter->channels = 0;
filter->interval = GST_SECOND / 10;
filter->decay_peak_ttl = GST_SECOND / 10 * 3;
filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
filter->message = TRUE;
filter->process = NULL;
}
static void
gst_level_dispose (GObject * obj)
{
GstLevel *filter = GST_LEVEL (obj);
g_free (filter->CS);
g_free (filter->peak);
g_free (filter->last_peak);
g_free (filter->decay_peak);
g_free (filter->decay_peak_base);
g_free (filter->decay_peak_age);
filter->CS = NULL;
filter->peak = NULL;
filter->last_peak = NULL;
filter->decay_peak = NULL;
filter->decay_peak_base = NULL;
filter->decay_peak_age = NULL;
G_OBJECT_CLASS (parent_class)->dispose (obj);
}
static void
gst_level_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLevel *filter = GST_LEVEL (object);
switch (prop_id) {
case PROP_SIGNAL_LEVEL:
filter->message = g_value_get_boolean (value);
break;
case PROP_SIGNAL_INTERVAL:
filter->interval = g_value_get_uint64 (value);
break;
case PROP_PEAK_TTL:
filter->decay_peak_ttl =
gst_guint64_to_gdouble (g_value_get_uint64 (value));
break;
case PROP_PEAK_FALLOFF:
filter->decay_peak_falloff = g_value_get_double (value);
break;
default:
break;
}
}
static void
gst_level_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstLevel *filter = GST_LEVEL (object);
switch (prop_id) {
case PROP_SIGNAL_LEVEL:
g_value_set_boolean (value, filter->message);
break;
case PROP_SIGNAL_INTERVAL:
g_value_set_uint64 (value, filter->interval);
break;
case PROP_PEAK_TTL:
g_value_set_uint64 (value, filter->decay_peak_ttl);
break;
case PROP_PEAK_FALLOFF:
g_value_set_double (value, filter->decay_peak_falloff);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* 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)
{
gint ret;
if (!gst_structure_get_int (structure, field, &ret))
g_assert_not_reached ();
return ret;
}
static gboolean
gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
GstLevel *filter = GST_LEVEL (trans);
const gchar *mimetype;
GstStructure *structure;
int i;
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);
g_free (filter->peak);
g_free (filter->last_peak);
g_free (filter->decay_peak);
g_free (filter->decay_peak_base);
g_free (filter->decay_peak_age);
filter->CS = g_new (double, filter->channels);
filter->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_base = g_new (double, filter->channels);
filter->decay_peak_age = g_new (GstClockTime, filter->channels);
for (i = 0; i < filter->channels; ++i) {
filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
filter->decay_peak[i] = filter->decay_peak_base[i] = 0.0;
filter->decay_peak_age[i] = G_GINT64_CONSTANT (0);
}
return TRUE;
}
static gboolean
gst_level_start (GstBaseTransform * trans)
{
GstLevel *filter = GST_LEVEL (trans);
filter->num_frames = 0;
return TRUE;
}
static GstMessage *
gst_level_message_new (GstLevel * l, GstClockTime endtime)
{
GstStructure *s;
GValue v = { 0, };
g_value_init (&v, GST_TYPE_LIST);
s = gst_structure_new ("level", "endtime", GST_TYPE_CLOCK_TIME,
endtime, NULL);
/* will copy-by-value */
gst_structure_set_value (s, "rms", &v);
gst_structure_set_value (s, "peak", &v);
gst_structure_set_value (s, "decay", &v);
g_value_unset (&v);
return gst_message_new_element (GST_OBJECT (l), s);
}
static void
gst_level_message_append_channel (GstMessage * m, gdouble rms, gdouble peak,
gdouble decay)
{
GstStructure *s;
GValue v = { 0, };
GValue *l;
g_value_init (&v, G_TYPE_DOUBLE);
s = (GstStructure *) gst_message_get_structure (m);
l = (GValue *) gst_structure_get_value (s, "rms");
g_value_set_double (&v, rms);
gst_value_list_append_value (l, &v); /* copies by value */
l = (GValue *) gst_structure_get_value (s, "peak");
g_value_set_double (&v, peak);
gst_value_list_append_value (l, &v); /* copies by value */
l = (GValue *) gst_structure_get_value (s, "decay");
g_value_set_double (&v, decay);
gst_value_list_append_value (l, &v); /* copies by value */
g_value_unset (&v);
}
static GstFlowReturn
gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
{
GstLevel *filter;
guint8 *in_data;
double CS = 0.0;
guint num_frames = 0;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
guint i;
filter = GST_LEVEL (trans);
for (i = 0; i < filter->channels; ++i)
filter->peak[i] = 0.0;
in_data = GST_BUFFER_DATA (in);
num_int_samples = GST_BUFFER_SIZE (in) / (filter->width / 8);
g_return_val_if_fail (num_int_samples % filter->channels == 0,
GST_FLOW_ERROR);
num_frames = num_int_samples / filter->channels;
for (i = 0; i < filter->channels; ++i) {
CS = 0.0;
filter->process (in_data, 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);
filter->CS[i] += CS;
in_data += (filter->width / 8);
}
filter->num_frames += num_frames;
for (i = 0; i < filter->channels; ++i) {
filter->decay_peak_age[i] +=
GST_FRAMES_TO_CLOCK_TIME (num_frames, filter->rate);
GST_LOG_OBJECT (filter, "filter peak info [%d]: decay peak %f, age %"
GST_TIME_FORMAT, i,
filter->decay_peak[i], GST_TIME_ARGS (filter->decay_peak_age[i]));
/* update running peak */
if (filter->peak[i] > filter->last_peak[i])
filter->last_peak[i] = filter->peak[i];
/* make decay peak fall off if too old */
if (gst_guint64_to_gdouble (filter->decay_peak_age[i]) >
filter->decay_peak_ttl) {
double falloff_dB;
double falloff;
GstClockTimeDiff falloff_time;
double length; /* length of falloff time in seconds */
falloff_time = GST_CLOCK_DIFF (filter->decay_peak_ttl,
gst_guint64_to_gdouble (filter->decay_peak_age[i]));
length = (gdouble) falloff_time / GST_SECOND;
falloff_dB = filter->decay_peak_falloff * length;
falloff = pow (10, falloff_dB / -20.0);
GST_LOG_OBJECT (filter,
"falloff: current %f, base %f, interval %" GST_TIME_FORMAT
", dB falloff %f, factor %e",
filter->decay_peak[i], filter->decay_peak_base[i],
GST_TIME_ARGS (falloff_time), falloff_dB, falloff);
filter->decay_peak[i] = filter->decay_peak_base[i] * falloff;
GST_LOG_OBJECT (filter,
"peak is %" GST_TIME_FORMAT " old, decayed with factor %e to %f",
GST_TIME_ARGS (filter->decay_peak_age[i]), falloff,
filter->decay_peak[i]);
} else {
GST_LOG_OBJECT (filter, "peak not old enough, not decaying");
}
/* if the peak of this run is higher, the decay peak gets reset */
if (filter->peak[i] >= filter->decay_peak[i]) {
GST_LOG_OBJECT (filter, "new peak, %f", filter->peak[i]);
filter->decay_peak[i] = filter->peak[i];
filter->decay_peak_base[i] = filter->peak[i];
filter->decay_peak_age[i] = G_GINT64_CONSTANT (0);
}
}
/* do we need to message ? */
if (filter->num_frames >=
GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate)) {
if (filter->message) {
GstMessage *m;
GstClockTime endtime =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (in))
+ GST_FRAMES_TO_CLOCK_TIME (num_frames, filter->rate);
m = gst_level_message_new (filter, endtime);
GST_LOG_OBJECT (filter,
"message: end time %" GST_TIME_FORMAT ", num_frames %d",
GST_TIME_ARGS (endtime), filter->num_frames);
for (i = 0; i < filter->channels; ++i) {
double RMS;
double RMSdB, lastdB, decaydB;
RMS = sqrt (filter->CS[i] / filter->num_frames);
GST_LOG_OBJECT (filter,
"message: channel %d, CS %f, num_frames %d, RMS %f",
i, filter->CS[i], filter->num_frames, RMS);
GST_LOG_OBJECT (filter,
"message: last_peak: %f, decay_peak: %f",
filter->last_peak[i], filter->decay_peak[i]);
/* 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]);
if (filter->decay_peak[i] < filter->last_peak[i]) {
/* this can happen in certain cases, for example when
* the last peak is between decay_peak and decay_peak_base */
GST_DEBUG_OBJECT (filter,
"message: decay peak dB %f smaller than last peak dB %f, copying",
decaydB, lastdB);
filter->decay_peak[i] = filter->last_peak[i];
}
GST_LOG_OBJECT (filter,
"message: RMS %f dB, peak %f dB, decay %f dB",
RMSdB, lastdB, decaydB);
gst_level_message_append_channel (m, RMSdB, lastdB, decaydB);
/* reset cumulative and normal peak */
filter->CS[i] = 0.0;
filter->last_peak[i] = 0.0;
}
gst_element_post_message (GST_ELEMENT (filter), m);
}
filter->num_frames = 0;
}
return GST_FLOW_OK;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "level", GST_RANK_NONE, GST_TYPE_LEVEL);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"level",
"Audio level plugin",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
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