/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-level
*
* Level analyses incoming audio buffers and, if the #GstLevel:message property
* is %TRUE, generates an element message named
* <classname>"level"</classname>:
* after each interval of time given by the #GstLevel:interval property.
* The message's structure contains these fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>"timestamp"</classname>:
* the timestamp of the buffer that triggered the message.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>"stream-time"</classname>:
* the stream time of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>"running-time"</classname>:
* the running_time of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>"duration"</classname>:
* the duration of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>"endtime"</classname>:
* the end time of the buffer that triggered the message as stream time (this
* is deprecated, as it can be calculated from stream-time + duration)
* </para>
* </listitem>
* <listitem>
* <para>
* #GValueArray of #gdouble
* <classname>"peak"</classname>:
* the peak power level in dB for each channel
* </para>
* </listitem>
* <listitem>
* <para>
* #GValueArray of #gdouble
* <classname>"decay"</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 #GstLevel:peak-ttl.
* When the decaying peak level drops, it does so at the decay rate as
* specified by the #GstLevel:peak-falloff.
* </para>
* </listitem>
* <listitem>
* <para>
* #GValueArray of #gdouble
* <classname>"rms"</classname>:
* the Root Mean Square (or average power) level in dB for each channel
* </para>
* </listitem>
* </itemizedlist>
*
* <refsect2>
* <title>Example application</title>
* <informalexample><programlisting language="C">
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" />
* </programlisting></informalexample>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstlevel.h"
GST_DEBUG_CATEGORY_STATIC (level_debug);
#define GST_CAT_DEFAULT level_debug
#define EPSILON 1e-35f
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate src_template_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
enum
{
PROP_0,
PROP_POST_MESSAGES,
PROP_MESSAGE,
PROP_INTERVAL,
PROP_PEAK_TTL,
PROP_PEAK_FALLOFF
};
#define gst_level_parent_class parent_class
G_DEFINE_TYPE (GstLevel, gst_level, 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_finalize (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_post_message (GstLevel * filter);
static gboolean gst_level_sink_event (GstBaseTransform * trans,
GstEvent * event);
static void gst_level_recalc_interval_frames (GstLevel * level);
static void
gst_level_class_init (GstLevelClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
gobject_class->set_property = gst_level_set_property;
gobject_class->get_property = gst_level_get_property;
gobject_class->finalize = gst_level_finalize;
/**
* GstLevel:post-messages
*
* Post messages on the bus with level information.
*
* Since: 1.1.0
*/
g_object_class_install_property (gobject_class, PROP_POST_MESSAGES,
g_param_spec_boolean ("post-messages", "Post Messages",
"Whether to post a 'level' element message on the bus for each "
"passed interval", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/* FIXME(2.0): remove this property */
/**
* GstLevel:post-messages
*
* Post messages on the bus with level information.
*
* Deprecated: use the #GstLevel:post-messages property
*/
#ifndef GST_REMOVE_DEPRECATED
g_object_class_install_property (gobject_class, PROP_MESSAGE,
g_param_spec_boolean ("message", "message",
"Post a 'level' message for each passed interval "
"(deprecated, use the post-messages property instead)", TRUE,
G_PARAM_READWRITE | G_PARAM_DEPRECATED | G_PARAM_STATIC_STRINGS));
#endif
g_object_class_install_property (gobject_class, PROP_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_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, 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_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, 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 | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (level_debug, "level", 0, "Level calculation");
gst_element_class_add_static_pad_template (element_class,
&sink_template_factory);
gst_element_class_add_static_pad_template (element_class,
&src_template_factory);
gst_element_class_set_static_metadata (element_class, "Level",
"Filter/Analyzer/Audio",
"RMS/Peak/Decaying Peak Level messager for audio/raw",
"Thomas Vander Stichele <thomas at apestaart dot org>");
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->sink_event = GST_DEBUG_FUNCPTR (gst_level_sink_event);
trans_class->passthrough_on_same_caps = TRUE;
}
static void
gst_level_init (GstLevel * filter)
{
filter->CS = NULL;
filter->peak = NULL;
filter->last_peak = NULL;
filter->decay_peak = NULL;
filter->decay_peak_base = NULL;
filter->decay_peak_age = NULL;
gst_audio_info_init (&filter->info);
filter->interval = GST_SECOND / 10;
filter->decay_peak_ttl = GST_SECOND / 10 * 3;
filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
filter->post_messages = TRUE;
filter->process = NULL;
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
}
static void
gst_level_finalize (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)->finalize (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_POST_MESSAGES:
/* fall-through */
case PROP_MESSAGE:
filter->post_messages = g_value_get_boolean (value);
break;
case PROP_INTERVAL:
filter->interval = g_value_get_uint64 (value);
/* Not exactly thread-safe, but property does not advertise that it
* can be changed at runtime anyway */
if (GST_AUDIO_INFO_RATE (&filter->info)) {
gst_level_recalc_interval_frames (filter);
}
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:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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_POST_MESSAGES:
/* fall-through */
case PROP_MESSAGE:
g_value_set_boolean (value, filter->post_messages);
break;
case PROP_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 and is not modified
* this filter only accepts signed audio data, so mid level is always 0
*
* for integers, this code considers the non-existant positive max value to be
* full-scale; so max-1 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 input 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 Peak Square */ \
\
for (j = 0; j < num; j += channels) { \
square = ((gdouble) in[j]) * in[j]; \
if (square > peaksquare) peaksquare = square; \
squaresum += square; \
} \
\
normalizer = (gdouble) (G_GINT64_CONSTANT(1) << (RESOLUTION * 2)); \
*NCS = squaresum / normalizer; \
*NPS = peaksquare / normalizer; \
}
DEFINE_INT_LEVEL_CALCULATOR (gint32, 31);
DEFINE_INT_LEVEL_CALCULATOR (gint16, 15);
DEFINE_INT_LEVEL_CALCULATOR (gint8, 7);
/* FIXME: use orc to calculate squaresums? */
#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 input 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 Peak Square */ \
\
/* orc_level_squaresum_f64(&squaresum,in,num); */ \
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);
/* we would need stride to deinterleave also
static void inline
gst_level_calculate_gdouble (gpointer data, guint num, guint channels,
gdouble *NCS, gdouble *NPS)
{
orc_level_squaresum_f64(NCS,(gdouble *)data,num);
*NPS = 0.0;
}
*/
static void
gst_level_recalc_interval_frames (GstLevel * level)
{
GstClockTime interval = level->interval;
guint sample_rate = GST_AUDIO_INFO_RATE (&level->info);
guint interval_frames;
interval_frames = GST_CLOCK_TIME_TO_FRAMES (interval, sample_rate);
if (interval_frames == 0) {
GST_WARNING_OBJECT (level, "interval %" GST_TIME_FORMAT " is too small, "
"should be at least %" GST_TIME_FORMAT " for sample rate %u",
GST_TIME_ARGS (interval),
GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (1, sample_rate)), sample_rate);
interval_frames = 1;
}
level->interval_frames = interval_frames;
GST_INFO_OBJECT (level, "interval_frames now %u for interval "
"%" GST_TIME_FORMAT " and sample rate %u", interval_frames,
GST_TIME_ARGS (interval), sample_rate);
}
static gboolean
gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
GstLevel *filter = GST_LEVEL (trans);
GstAudioInfo info;
gint i, channels;
if (!gst_audio_info_from_caps (&info, in))
return FALSE;
switch (GST_AUDIO_INFO_FORMAT (&info)) {
case GST_AUDIO_FORMAT_S8:
filter->process = gst_level_calculate_gint8;
break;
case GST_AUDIO_FORMAT_S16:
filter->process = gst_level_calculate_gint16;
break;
case GST_AUDIO_FORMAT_S32:
filter->process = gst_level_calculate_gint32;
break;
case GST_AUDIO_FORMAT_F32:
filter->process = gst_level_calculate_gfloat;
break;
case GST_AUDIO_FORMAT_F64:
filter->process = gst_level_calculate_gdouble;
break;
default:
filter->process = NULL;
break;
}
filter->info = info;
channels = GST_AUDIO_INFO_CHANNELS (&info);
/* 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 (gdouble, channels);
filter->peak = g_new (gdouble, channels);
filter->last_peak = g_new (gdouble, channels);
filter->decay_peak = g_new (gdouble, channels);
filter->decay_peak_base = g_new (gdouble, channels);
filter->decay_peak_age = g_new (GstClockTime, channels);
for (i = 0; i < 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_GUINT64_CONSTANT (0);
}
gst_level_recalc_interval_frames (filter);
return TRUE;
}
static gboolean
gst_level_start (GstBaseTransform * trans)
{
GstLevel *filter = GST_LEVEL (trans);
filter->num_frames = 0;
filter->message_ts = GST_CLOCK_TIME_NONE;
return TRUE;
}
static GstMessage *
gst_level_message_new (GstLevel * level, GstClockTime timestamp,
GstClockTime duration)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (level);
GstStructure *s;
GValue v = { 0, };
GstClockTime endtime, running_time, stream_time;
running_time = gst_segment_to_running_time (&trans->segment, GST_FORMAT_TIME,
timestamp);
stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
timestamp);
/* endtime is for backwards compatibility */
endtime = stream_time + duration;
s = gst_structure_new ("level",
"endtime", GST_TYPE_CLOCK_TIME, endtime,
"timestamp", G_TYPE_UINT64, timestamp,
"stream-time", G_TYPE_UINT64, stream_time,
"running-time", G_TYPE_UINT64, running_time,
"duration", G_TYPE_UINT64, duration, NULL);
g_value_init (&v, G_TYPE_VALUE_ARRAY);
g_value_take_boxed (&v, g_value_array_new (0));
gst_structure_take_value (s, "rms", &v);
g_value_init (&v, G_TYPE_VALUE_ARRAY);
g_value_take_boxed (&v, g_value_array_new (0));
gst_structure_take_value (s, "peak", &v);
g_value_init (&v, G_TYPE_VALUE_ARRAY);
g_value_take_boxed (&v, g_value_array_new (0));
gst_structure_take_value (s, "decay", &v);
return gst_message_new_element (GST_OBJECT (level), s);
}
static void
gst_level_message_append_channel (GstMessage * m, gdouble rms, gdouble peak,
gdouble decay)
{
const GValue *array_val;
GstStructure *s;
GValueArray *arr;
GValue v = { 0, };
g_value_init (&v, G_TYPE_DOUBLE);
s = (GstStructure *) gst_message_get_structure (m);
array_val = gst_structure_get_value (s, "rms");
arr = (GValueArray *) g_value_get_boxed (array_val);
g_value_set_double (&v, rms);
g_value_array_append (arr, &v); /* copies by value */
array_val = gst_structure_get_value (s, "peak");
arr = (GValueArray *) g_value_get_boxed (array_val);
g_value_set_double (&v, peak);
g_value_array_append (arr, &v); /* copies by value */
array_val = gst_structure_get_value (s, "decay");
arr = (GValueArray *) g_value_get_boxed (array_val);
g_value_set_double (&v, decay);
g_value_array_append (arr, &v); /* copies by value */
g_value_unset (&v);
}
static GstFlowReturn
gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
{
GstLevel *filter;
GstMapInfo map;
guint8 *in_data;
gsize in_size;
gdouble CS;
guint i;
guint num_frames;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
guint block_size, block_int_size; /* we subdivide buffers to not skip message
* intervals */
GstClockTimeDiff falloff_time;
gint channels, rate, bps;
filter = GST_LEVEL (trans);
channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
bps = GST_AUDIO_INFO_BPS (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
gst_buffer_map (in, &map, GST_MAP_READ);
in_data = map.data;
in_size = map.size;
num_int_samples = in_size / bps;
GST_LOG_OBJECT (filter, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (in)));
g_return_val_if_fail (num_int_samples % channels == 0, GST_FLOW_ERROR);
if (GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_DISCONT)) {
filter->message_ts = GST_BUFFER_TIMESTAMP (in);
filter->num_frames = 0;
}
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (filter->message_ts))) {
filter->message_ts = GST_BUFFER_TIMESTAMP (in);
}
num_frames = num_int_samples / channels;
while (num_frames > 0) {
block_size = filter->interval_frames - filter->num_frames;
block_size = MIN (block_size, num_frames);
block_int_size = block_size * channels;
for (i = 0; i < channels; ++i) {
if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) {
filter->process (in_data + (bps * i), block_int_size, channels, &CS,
&filter->peak[i]);
GST_LOG_OBJECT (filter,
"[%d]: cumulative squares %lf, over %d samples/%d channels",
i, CS, block_int_size, channels);
filter->CS[i] += CS;
} else {
filter->peak[i] = 0.0;
}
filter->decay_peak_age[i] += GST_FRAMES_TO_CLOCK_TIME (num_frames, rate);
GST_LOG_OBJECT (filter,
"[%d]: peak %f, last peak %f, decay peak %f, age %" GST_TIME_FORMAT,
i, filter->peak[i], filter->last_peak[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 */
falloff_time =
GST_CLOCK_DIFF (gst_gdouble_to_guint64 (filter->decay_peak_ttl),
filter->decay_peak_age[i]);
if (falloff_time > 0) {
gdouble falloff_dB;
gdouble falloff;
gdouble length; /* length of falloff time in seconds */
length = (gdouble) falloff_time / (gdouble) 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);
}
}
in_data += block_size * bps * channels;
filter->num_frames += block_size;
num_frames -= block_size;
/* do we need to message ? */
if (filter->num_frames >= filter->interval_frames) {
gst_level_post_message (filter);
}
}
gst_buffer_unmap (in, &map);
return GST_FLOW_OK;
}
static void
gst_level_post_message (GstLevel * filter)
{
guint i;
gint channels, rate, frames = filter->num_frames;
GstClockTime duration;
channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
duration = GST_FRAMES_TO_CLOCK_TIME (frames, rate);
if (filter->post_messages) {
GstMessage *m =
gst_level_message_new (filter, filter->message_ts, duration);
GST_LOG_OBJECT (filter,
"message: ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
", num_frames %d", GST_TIME_ARGS (filter->message_ts),
GST_TIME_ARGS (duration), frames);
for (i = 0; i < channels; ++i) {
gdouble RMS;
gdouble RMSdB, peakdB, decaydB;
RMS = sqrt (filter->CS[i] / frames);
GST_LOG_OBJECT (filter,
"message: channel %d, CS %f, RMS %f", i, filter->CS[i], 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 + EPSILON);
/* peak values are square sums, ie. power, so 10 * log 10 */
peakdB = 10 * log10 (filter->last_peak[i] + EPSILON);
decaydB = 10 * log10 (filter->decay_peak[i] + EPSILON);
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, peakdB);
filter->decay_peak[i] = filter->last_peak[i];
}
GST_LOG_OBJECT (filter,
"message: RMS %f dB, peak %f dB, decay %f dB",
RMSdB, peakdB, decaydB);
gst_level_message_append_channel (m, RMSdB, peakdB, 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 -= frames;
filter->message_ts += duration;
}
static gboolean
gst_level_sink_event (GstBaseTransform * trans, GstEvent * event)
{
if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
GstLevel *filter = GST_LEVEL (trans);
gst_level_post_message (filter);
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
}
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);