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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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755 lines
27 KiB
C
755 lines
27 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2000,2001,2002,2003,2005
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* Thomas Vander Stichele <thomas at apestaart dot org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-level
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* @title: level
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*
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* Level analyses incoming audio buffers and, if the #GstLevel:message property
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* is %TRUE, generates an element message named
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* `level`: after each interval of time given by the #GstLevel:interval property.
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* The message's structure contains these fields:
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*
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* * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message.
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* * #GstClockTime `stream-time`: the stream time of the buffer.
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* * #GstClockTime `running-time`: the running_time of the buffer.
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* * #GstClockTime `duration`: the duration of the buffer.
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* * #GstClockTime `endtime`: the end time of the buffer that triggered the message as
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* stream time (this is deprecated, as it can be calculated from stream-time + duration)
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* * #GValueArray of #gdouble `peak`: the peak power level in dB for each channel
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* * #GValueArray of #gdouble `decay`: the decaying peak power level in dB for each channel
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* The decaying peak level follows the peak level, but starts dropping if no
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* new peak is reached after the time given by the #GstLevel:peak-ttl.
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* When the decaying peak level drops, it does so at the decay rate as
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* specified by the #GstLevel:peak-falloff.
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* * #GValueArray of #gdouble `rms`: the Root Mean Square (or average power) level in dB
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* for each channel
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*
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* ## Example application
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*
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* {{ tests/examples/level/level-example.c }}
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <string.h>
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#include <math.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "gstlevel.h"
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GST_DEBUG_CATEGORY_STATIC (level_debug);
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#define GST_CAT_DEFAULT level_debug
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#define EPSILON 1e-35f
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static GstStaticPadTemplate sink_template_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
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", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static GstStaticPadTemplate src_template_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
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", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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enum
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{
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PROP_0,
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PROP_POST_MESSAGES,
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PROP_MESSAGE,
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PROP_INTERVAL,
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PROP_PEAK_TTL,
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PROP_PEAK_FALLOFF
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};
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#define gst_level_parent_class parent_class
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G_DEFINE_TYPE (GstLevel, gst_level, GST_TYPE_BASE_TRANSFORM);
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static void gst_level_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_level_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_level_finalize (GObject * obj);
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static gboolean gst_level_set_caps (GstBaseTransform * trans, GstCaps * in,
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GstCaps * out);
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static gboolean gst_level_start (GstBaseTransform * trans);
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static GstFlowReturn gst_level_transform_ip (GstBaseTransform * trans,
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GstBuffer * in);
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static void gst_level_post_message (GstLevel * filter);
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static gboolean gst_level_sink_event (GstBaseTransform * trans,
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GstEvent * event);
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static void gst_level_recalc_interval_frames (GstLevel * level);
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static void
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gst_level_class_init (GstLevelClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
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gobject_class->set_property = gst_level_set_property;
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gobject_class->get_property = gst_level_get_property;
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gobject_class->finalize = gst_level_finalize;
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/**
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* GstLevel:post-messages
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*
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* Post messages on the bus with level information.
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*
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* Since: 1.1.0
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*/
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g_object_class_install_property (gobject_class, PROP_POST_MESSAGES,
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g_param_spec_boolean ("post-messages", "Post Messages",
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"Whether to post a 'level' element message on the bus for each "
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"passed interval", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/* FIXME(2.0): remove this property */
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/**
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* GstLevel:post-messages
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*
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* Post messages on the bus with level information.
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*
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* Deprecated: use the #GstLevel:post-messages property
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*/
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#ifndef GST_REMOVE_DEPRECATED
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g_object_class_install_property (gobject_class, PROP_MESSAGE,
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g_param_spec_boolean ("message", "message",
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"Post a 'level' message for each passed interval "
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"(deprecated, use the post-messages property instead)", TRUE,
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G_PARAM_READWRITE | G_PARAM_DEPRECATED | G_PARAM_STATIC_STRINGS));
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#endif
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g_object_class_install_property (gobject_class, PROP_INTERVAL,
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g_param_spec_uint64 ("interval", "Interval",
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"Interval of time between message posts (in nanoseconds)",
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1, G_MAXUINT64, GST_SECOND / 10,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PEAK_TTL,
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g_param_spec_uint64 ("peak-ttl", "Peak TTL",
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"Time To Live of decay peak before it falls back (in nanoseconds)",
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0, G_MAXUINT64, GST_SECOND / 10 * 3,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PEAK_FALLOFF,
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g_param_spec_double ("peak-falloff", "Peak Falloff",
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"Decay rate of decay peak after TTL (in dB/sec)",
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0.0, G_MAXDOUBLE, 10.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (level_debug, "level", 0, "Level calculation");
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gst_element_class_add_static_pad_template (element_class,
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&sink_template_factory);
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gst_element_class_add_static_pad_template (element_class,
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&src_template_factory);
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gst_element_class_set_static_metadata (element_class, "Level",
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"Filter/Analyzer/Audio",
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"RMS/Peak/Decaying Peak Level messager for audio/raw",
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"Thomas Vander Stichele <thomas at apestaart dot org>");
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trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_level_set_caps);
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trans_class->start = GST_DEBUG_FUNCPTR (gst_level_start);
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trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_level_transform_ip);
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trans_class->sink_event = GST_DEBUG_FUNCPTR (gst_level_sink_event);
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trans_class->passthrough_on_same_caps = TRUE;
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}
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static void
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gst_level_init (GstLevel * filter)
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{
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filter->CS = NULL;
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filter->peak = NULL;
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filter->last_peak = NULL;
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filter->decay_peak = NULL;
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filter->decay_peak_base = NULL;
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filter->decay_peak_age = NULL;
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gst_audio_info_init (&filter->info);
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filter->interval = GST_SECOND / 10;
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filter->decay_peak_ttl = GST_SECOND / 10 * 3;
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filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
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filter->post_messages = TRUE;
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filter->process = NULL;
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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}
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static void
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gst_level_finalize (GObject * obj)
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{
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GstLevel *filter = GST_LEVEL (obj);
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g_free (filter->CS);
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g_free (filter->peak);
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g_free (filter->last_peak);
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g_free (filter->decay_peak);
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g_free (filter->decay_peak_base);
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g_free (filter->decay_peak_age);
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filter->CS = NULL;
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filter->peak = NULL;
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filter->last_peak = NULL;
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filter->decay_peak = NULL;
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filter->decay_peak_base = NULL;
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filter->decay_peak_age = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_level_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstLevel *filter = GST_LEVEL (object);
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switch (prop_id) {
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case PROP_POST_MESSAGES:
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/* fall-through */
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case PROP_MESSAGE:
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filter->post_messages = g_value_get_boolean (value);
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break;
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case PROP_INTERVAL:
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filter->interval = g_value_get_uint64 (value);
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/* Not exactly thread-safe, but property does not advertise that it
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* can be changed at runtime anyway */
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if (GST_AUDIO_INFO_RATE (&filter->info)) {
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gst_level_recalc_interval_frames (filter);
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}
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break;
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case PROP_PEAK_TTL:
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filter->decay_peak_ttl =
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gst_guint64_to_gdouble (g_value_get_uint64 (value));
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break;
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case PROP_PEAK_FALLOFF:
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filter->decay_peak_falloff = g_value_get_double (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_level_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstLevel *filter = GST_LEVEL (object);
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switch (prop_id) {
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case PROP_POST_MESSAGES:
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/* fall-through */
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case PROP_MESSAGE:
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g_value_set_boolean (value, filter->post_messages);
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break;
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case PROP_INTERVAL:
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g_value_set_uint64 (value, filter->interval);
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break;
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case PROP_PEAK_TTL:
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g_value_set_uint64 (value, filter->decay_peak_ttl);
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break;
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case PROP_PEAK_FALLOFF:
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g_value_set_double (value, filter->decay_peak_falloff);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* process one (interleaved) channel of incoming samples
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* calculate square sum of samples
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* normalize and average over number of samples
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* returns a normalized cumulative square value, which can be averaged
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* to return the average power as a double between 0 and 1
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* also returns the normalized peak power (square of the highest amplitude)
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*
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* caller must assure num is a multiple of channels
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* samples for multiple channels are interleaved
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* input sample data enters in *in_data and is not modified
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* this filter only accepts signed audio data, so mid level is always 0
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*
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* for integers, this code considers the non-existent positive max value to be
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* full-scale; so max-1 will not map to 1.0
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*/
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#define DEFINE_INT_LEVEL_CALCULATOR(TYPE, RESOLUTION) \
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static void inline \
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gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
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gdouble *NCS, gdouble *NPS) \
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{ \
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TYPE * in = (TYPE *)data; \
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register guint j; \
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gdouble squaresum = 0.0; /* square sum of the input samples */ \
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register gdouble square = 0.0; /* Square */ \
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register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
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gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
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\
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/* *NCS = 0.0; Normalized Cumulative Square */ \
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/* *NPS = 0.0; Normalized Peak Square */ \
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\
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for (j = 0; j < num; j += channels) { \
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square = ((gdouble) in[j]) * in[j]; \
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if (square > peaksquare) peaksquare = square; \
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squaresum += square; \
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} \
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\
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normalizer = (gdouble) (G_GINT64_CONSTANT(1) << (RESOLUTION * 2)); \
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*NCS = squaresum / normalizer; \
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*NPS = peaksquare / normalizer; \
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}
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DEFINE_INT_LEVEL_CALCULATOR (gint32, 31);
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DEFINE_INT_LEVEL_CALCULATOR (gint16, 15);
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DEFINE_INT_LEVEL_CALCULATOR (gint8, 7);
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/* FIXME: use orc to calculate squaresums? */
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#define DEFINE_FLOAT_LEVEL_CALCULATOR(TYPE) \
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static void inline \
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gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
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gdouble *NCS, gdouble *NPS) \
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{ \
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TYPE * in = (TYPE *)data; \
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register guint j; \
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gdouble squaresum = 0.0; /* square sum of the input samples */ \
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register gdouble square = 0.0; /* Square */ \
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register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
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\
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/* *NCS = 0.0; Normalized Cumulative Square */ \
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/* *NPS = 0.0; Normalized Peak Square */ \
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\
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/* orc_level_squaresum_f64(&squaresum,in,num); */ \
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for (j = 0; j < num; j += channels) { \
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square = ((gdouble) in[j]) * in[j]; \
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if (square > peaksquare) peaksquare = square; \
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squaresum += square; \
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} \
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\
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*NCS = squaresum; \
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*NPS = peaksquare; \
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}
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DEFINE_FLOAT_LEVEL_CALCULATOR (gfloat);
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DEFINE_FLOAT_LEVEL_CALCULATOR (gdouble);
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/* we would need stride to deinterleave also
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static void inline
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gst_level_calculate_gdouble (gpointer data, guint num, guint channels,
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gdouble *NCS, gdouble *NPS)
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{
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orc_level_squaresum_f64(NCS,(gdouble *)data,num);
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*NPS = 0.0;
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}
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*/
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static void
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gst_level_recalc_interval_frames (GstLevel * level)
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{
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GstClockTime interval = level->interval;
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guint sample_rate = GST_AUDIO_INFO_RATE (&level->info);
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guint interval_frames;
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interval_frames = GST_CLOCK_TIME_TO_FRAMES (interval, sample_rate);
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if (interval_frames == 0) {
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GST_WARNING_OBJECT (level, "interval %" GST_TIME_FORMAT " is too small, "
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"should be at least %" GST_TIME_FORMAT " for sample rate %u",
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GST_TIME_ARGS (interval),
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GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (1, sample_rate)), sample_rate);
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interval_frames = 1;
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}
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level->interval_frames = interval_frames;
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GST_INFO_OBJECT (level, "interval_frames now %u for interval "
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"%" GST_TIME_FORMAT " and sample rate %u", interval_frames,
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GST_TIME_ARGS (interval), sample_rate);
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}
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static gboolean
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gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
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{
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GstLevel *filter = GST_LEVEL (trans);
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GstAudioInfo info;
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gint i, channels;
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if (!gst_audio_info_from_caps (&info, in))
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return FALSE;
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switch (GST_AUDIO_INFO_FORMAT (&info)) {
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case GST_AUDIO_FORMAT_S8:
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filter->process = gst_level_calculate_gint8;
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break;
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case GST_AUDIO_FORMAT_S16:
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filter->process = gst_level_calculate_gint16;
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break;
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case GST_AUDIO_FORMAT_S32:
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filter->process = gst_level_calculate_gint32;
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break;
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case GST_AUDIO_FORMAT_F32:
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filter->process = gst_level_calculate_gfloat;
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break;
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case GST_AUDIO_FORMAT_F64:
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filter->process = gst_level_calculate_gdouble;
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break;
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default:
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filter->process = NULL;
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break;
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}
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filter->info = info;
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channels = GST_AUDIO_INFO_CHANNELS (&info);
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/* allocate channel variable arrays */
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g_free (filter->CS);
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g_free (filter->peak);
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g_free (filter->last_peak);
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g_free (filter->decay_peak);
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g_free (filter->decay_peak_base);
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g_free (filter->decay_peak_age);
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filter->CS = g_new (gdouble, channels);
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filter->peak = g_new (gdouble, channels);
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filter->last_peak = g_new (gdouble, channels);
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filter->decay_peak = g_new (gdouble, channels);
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filter->decay_peak_base = g_new (gdouble, channels);
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filter->decay_peak_age = g_new (GstClockTime, channels);
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for (i = 0; i < channels; ++i) {
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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);
|