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Diffstat (limited to 'gst/level/gstlevel.c')
| -rw-r--r-- | gst/level/gstlevel.c | 727 |
1 files changed, 727 insertions, 0 deletions
diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c new file mode 100644 index 0000000..6d89a68 --- /dev/null +++ b/gst/level/gstlevel.c @@ -0,0 +1,727 @@ +/* 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 + * + * 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> + * #GstValueList of #gdouble + * <classname>"peak"</classname>: + * the peak power level in dB for each channel + * </para> + * </listitem> + * <listitem> + * <para> + * #GstValueList 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 <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>"rms"</classname>: + * the Root Mean Square (or average power) level in dB for each channel + * </para> + * </listitem> + * </itemizedlist> + * + * <refsect2> + * <title>Example application</title> + * |[ + * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" /> + * ]| + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#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-int, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) { 8, 16, 32 }, " + "depth = (int) { 8, 16, 32 }, " + "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, 32 }, " + "depth = (int) { 8, 16, 32 }, " + "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_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_base_init (gpointer g_class) +{ + GstElementClass *element_class = g_class; + + 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_details_simple (element_class, "Level", + "Filter/Analyzer/Audio", + "RMS/Peak/Decaying Peak Level messager for audio/raw", + "Thomas Vander Stichele <thomas at apestaart dot org>"); +} + +static void +gst_level_class_init (GstLevelClass * klass) +{ + GObjectClass *gobject_class = G_OBJECT_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; + + g_object_class_install_property (gobject_class, PROP_SIGNAL_LEVEL, + g_param_spec_boolean ("message", "message", + "Post a level message for each passed interval", + TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, 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_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"); + + 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; + + 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_SIGNAL_LEVEL: + filter->message = g_value_get_boolean (value); + break; + case PROP_SIGNAL_INTERVAL: + filter->interval = g_value_get_uint64 (value); + if (filter->rate) { + filter->interval_frames = + GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate); + } + 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_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) (G_GINT64_CONSTANT(1) << (RESOLUTION * 2)); \ + \ + /* oil_squaresum_shifted_s16(&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 / normalizer; \ + *NPS = peaksquare / normalizer; \ +} + +DEFINE_INT_LEVEL_CALCULATOR (gint32, 31); +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 */ \ + \ + /* oil_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) +{ + oil_squaresum_f64(NCS,(gdouble *)data,num); + *NPS = 0.0; +} +*/ + + +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; + gint 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; + case 32: + filter->process = gst_level_calculate_gint32; + 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 (gdouble, filter->channels); + filter->peak = g_new (gdouble, filter->channels); + filter->last_peak = g_new (gdouble, filter->channels); + filter->decay_peak = g_new (gdouble, filter->channels); + filter->decay_peak_base = g_new (gdouble, 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_GUINT64_CONSTANT (0); + } + + filter->interval_frames = + GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate); + + 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 * level, GstClockTime timestamp, + GstClockTime duration) +{ + GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (level); + GstStructure *s; + GValue v = { 0, }; + GstClockTime endtime, running_time, stream_time; + + g_value_init (&v, GST_TYPE_LIST); + + 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); + /* 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 (level), 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; + gdouble CS; + guint i; + guint num_frames = 0; + guint num_int_samples = 0; /* number of interleaved samples + * ie. total count for all channels combined */ + GstClockTimeDiff falloff_time; + + filter = GST_LEVEL (trans); + + in_data = GST_BUFFER_DATA (in); + num_int_samples = GST_BUFFER_SIZE (in) / (filter->width / 8); + + 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 % filter->channels == 0, + GST_FLOW_ERROR); + + num_frames = num_int_samples / filter->channels; + + for (i = 0; i < filter->channels; ++i) { + if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) { + 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; + } else { + filter->peak[i] = 0.0; + } + in_data += (filter->width / 8); + + 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 */ + 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); + } + } + + if (G_UNLIKELY (!filter->num_frames)) { + /* remember start timestamp for message */ + filter->message_ts = GST_BUFFER_TIMESTAMP (in); + } + filter->num_frames += num_frames; + + /* do we need to message ? */ + if (filter->num_frames >= filter->interval_frames) { + if (filter->message) { + GstMessage *m; + GstClockTime duration = + GST_FRAMES_TO_CLOCK_TIME (filter->num_frames, filter->rate); + + m = gst_level_message_new (filter, filter->message_ts, duration); + + GST_LOG_OBJECT (filter, + "message: ts %" GST_TIME_FORMAT ", num_frames %d", + GST_TIME_ARGS (filter->message_ts), filter->num_frames); + + for (i = 0; i < filter->channels; ++i) { + gdouble RMS; + gdouble 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 + EPSILON); + /* peak values are square sums, ie. power, so 10 * log 10 */ + lastdB = 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, 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) +{ + /*oil_init (); */ + + 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); |