2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) 2000,2001,2002,2003,2005
4 * Thomas Vander Stichele <thomas at apestaart dot org>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-level
25 * Level analyses incoming audio buffers and, if the #GstLevel:message property
26 * is #TRUE, generates an element message named
27 * <classname>"level"</classname>:
28 * after each interval of time given by the #GstLevel:interval property.
29 * The message's structure contains these fields:
34 * <classname>"timestamp"</classname>:
35 * the timestamp of the buffer that triggered the message.
41 * <classname>"stream-time"</classname>:
42 * the stream time of the buffer.
48 * <classname>"running-time"</classname>:
49 * the running_time of the buffer.
55 * <classname>"duration"</classname>:
56 * the duration of the buffer.
62 * <classname>"endtime"</classname>:
63 * the end time of the buffer that triggered the message as stream time (this
64 * is deprecated, as it can be calculated from stream-time + duration)
69 * #GValueArray of #gdouble
70 * <classname>"peak"</classname>:
71 * the peak power level in dB for each channel
76 * #GValueArray of #gdouble
77 * <classname>"decay"</classname>:
78 * the decaying peak power level in dB for each channel
79 * the decaying peak level follows the peak level, but starts dropping
80 * if no new peak is reached after the time given by
81 * the <link linkend="GstLevel--peak-ttl">the time to live</link>.
82 * When the decaying peak level drops, it does so at the decay rate
84 * <link linkend="GstLevel--peak-falloff">the peak falloff rate</link>.
89 * #GValueArray of #gdouble
90 * <classname>"rms"</classname>:
91 * the Root Mean Square (or average power) level in dB for each channel
97 * <title>Example application</title>
99 * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" />
108 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
109 * with newer GLib versions (>= 2.31.0) */
110 #define GLIB_DISABLE_DEPRECATION_WARNINGS
115 #include <gst/audio/audio.h>
117 #include "gstlevel.h"
119 GST_DEBUG_CATEGORY_STATIC (level_debug);
120 #define GST_CAT_DEFAULT level_debug
122 #define EPSILON 1e-35f
124 static GstStaticPadTemplate sink_template_factory =
125 GST_STATIC_PAD_TEMPLATE ("sink",
128 GST_STATIC_CAPS ("audio/x-raw, "
129 "format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
130 ", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
131 "layout = (string) interleaved, "
132 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
135 static GstStaticPadTemplate src_template_factory =
136 GST_STATIC_PAD_TEMPLATE ("src",
139 GST_STATIC_CAPS ("audio/x-raw, "
140 "format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
141 ", " GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
142 "layout = (string) interleaved, "
143 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
156 #define gst_level_parent_class parent_class
157 G_DEFINE_TYPE (GstLevel, gst_level, GST_TYPE_BASE_TRANSFORM);
159 static void gst_level_set_property (GObject * object, guint prop_id,
160 const GValue * value, GParamSpec * pspec);
161 static void gst_level_get_property (GObject * object, guint prop_id,
162 GValue * value, GParamSpec * pspec);
163 static void gst_level_finalize (GObject * obj);
165 static gboolean gst_level_set_caps (GstBaseTransform * trans, GstCaps * in,
167 static gboolean gst_level_start (GstBaseTransform * trans);
168 static GstFlowReturn gst_level_transform_ip (GstBaseTransform * trans,
170 static void gst_level_post_message (GstLevel * filter);
171 static gboolean gst_level_sink_event (GstBaseTransform * trans,
176 gst_level_class_init (GstLevelClass * klass)
178 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
179 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
180 GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
182 gobject_class->set_property = gst_level_set_property;
183 gobject_class->get_property = gst_level_get_property;
184 gobject_class->finalize = gst_level_finalize;
187 * GstLevel:post-messages
189 * Post messages on the bus with level information.
193 g_object_class_install_property (gobject_class, PROP_POST_MESSAGES,
194 g_param_spec_boolean ("post-messages", "Post Messages",
195 "Whether to post a 'level' element message on the bus for each "
196 "passed interval", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
197 /* FIXME(2.0): remove this property */
199 * GstLevel:post-messages
201 * Post messages on the bus with level information.
203 * Deprecated: use the #GstLevel:post-messages property
205 g_object_class_install_property (gobject_class, PROP_MESSAGE,
206 g_param_spec_boolean ("message", "message",
207 "Post a 'level' message for each passed interval (deprecated)",
208 TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
209 g_object_class_install_property (gobject_class, PROP_INTERVAL,
210 g_param_spec_uint64 ("interval", "Interval",
211 "Interval of time between message posts (in nanoseconds)",
212 1, G_MAXUINT64, GST_SECOND / 10,
213 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
214 g_object_class_install_property (gobject_class, PROP_PEAK_TTL,
215 g_param_spec_uint64 ("peak-ttl", "Peak TTL",
216 "Time To Live of decay peak before it falls back (in nanoseconds)",
217 0, G_MAXUINT64, GST_SECOND / 10 * 3,
218 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
219 g_object_class_install_property (gobject_class, PROP_PEAK_FALLOFF,
220 g_param_spec_double ("peak-falloff", "Peak Falloff",
221 "Decay rate of decay peak after TTL (in dB/sec)",
222 0.0, G_MAXDOUBLE, 10.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
224 GST_DEBUG_CATEGORY_INIT (level_debug, "level", 0, "Level calculation");
226 gst_element_class_add_pad_template (element_class,
227 gst_static_pad_template_get (&sink_template_factory));
228 gst_element_class_add_pad_template (element_class,
229 gst_static_pad_template_get (&src_template_factory));
230 gst_element_class_set_static_metadata (element_class, "Level",
231 "Filter/Analyzer/Audio",
232 "RMS/Peak/Decaying Peak Level messager for audio/raw",
233 "Thomas Vander Stichele <thomas at apestaart dot org>");
235 trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_level_set_caps);
236 trans_class->start = GST_DEBUG_FUNCPTR (gst_level_start);
237 trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_level_transform_ip);
238 trans_class->sink_event = GST_DEBUG_FUNCPTR (gst_level_sink_event);
239 trans_class->passthrough_on_same_caps = TRUE;
243 gst_level_init (GstLevel * filter)
247 filter->last_peak = NULL;
248 filter->decay_peak = NULL;
249 filter->decay_peak_base = NULL;
250 filter->decay_peak_age = NULL;
252 gst_audio_info_init (&filter->info);
254 filter->interval = GST_SECOND / 10;
255 filter->decay_peak_ttl = GST_SECOND / 10 * 3;
256 filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
258 filter->post_messages = TRUE;
260 filter->process = NULL;
262 gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
266 gst_level_finalize (GObject * obj)
268 GstLevel *filter = GST_LEVEL (obj);
271 g_free (filter->peak);
272 g_free (filter->last_peak);
273 g_free (filter->decay_peak);
274 g_free (filter->decay_peak_base);
275 g_free (filter->decay_peak_age);
279 filter->last_peak = NULL;
280 filter->decay_peak = NULL;
281 filter->decay_peak_base = NULL;
282 filter->decay_peak_age = NULL;
284 G_OBJECT_CLASS (parent_class)->finalize (obj);
288 gst_level_set_property (GObject * object, guint prop_id,
289 const GValue * value, GParamSpec * pspec)
291 GstLevel *filter = GST_LEVEL (object);
294 case PROP_POST_MESSAGES:
297 filter->post_messages = g_value_get_boolean (value);
300 filter->interval = g_value_get_uint64 (value);
301 if (GST_AUDIO_INFO_RATE (&filter->info)) {
302 filter->interval_frames =
303 GST_CLOCK_TIME_TO_FRAMES (filter->interval,
304 GST_AUDIO_INFO_RATE (&filter->info));
308 filter->decay_peak_ttl =
309 gst_guint64_to_gdouble (g_value_get_uint64 (value));
311 case PROP_PEAK_FALLOFF:
312 filter->decay_peak_falloff = g_value_get_double (value);
315 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
321 gst_level_get_property (GObject * object, guint prop_id,
322 GValue * value, GParamSpec * pspec)
324 GstLevel *filter = GST_LEVEL (object);
327 case PROP_POST_MESSAGES:
330 g_value_set_boolean (value, filter->post_messages);
333 g_value_set_uint64 (value, filter->interval);
336 g_value_set_uint64 (value, filter->decay_peak_ttl);
338 case PROP_PEAK_FALLOFF:
339 g_value_set_double (value, filter->decay_peak_falloff);
342 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
348 /* process one (interleaved) channel of incoming samples
349 * calculate square sum of samples
350 * normalize and average over number of samples
351 * returns a normalized cumulative square value, which can be averaged
352 * to return the average power as a double between 0 and 1
353 * also returns the normalized peak power (square of the highest amplitude)
355 * caller must assure num is a multiple of channels
356 * samples for multiple channels are interleaved
357 * input sample data enters in *in_data and is not modified
358 * this filter only accepts signed audio data, so mid level is always 0
360 * for integers, this code considers the non-existant positive max value to be
361 * full-scale; so max-1 will not map to 1.0
364 #define DEFINE_INT_LEVEL_CALCULATOR(TYPE, RESOLUTION) \
366 gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
367 gdouble *NCS, gdouble *NPS) \
369 TYPE * in = (TYPE *)data; \
371 gdouble squaresum = 0.0; /* square sum of the input samples */ \
372 register gdouble square = 0.0; /* Square */ \
373 register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
374 gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
376 /* *NCS = 0.0; Normalized Cumulative Square */ \
377 /* *NPS = 0.0; Normalized Peak Square */ \
379 for (j = 0; j < num; j += channels) { \
380 square = ((gdouble) in[j]) * in[j]; \
381 if (square > peaksquare) peaksquare = square; \
382 squaresum += square; \
385 normalizer = (gdouble) (G_GINT64_CONSTANT(1) << (RESOLUTION * 2)); \
386 *NCS = squaresum / normalizer; \
387 *NPS = peaksquare / normalizer; \
390 DEFINE_INT_LEVEL_CALCULATOR (gint32, 31);
391 DEFINE_INT_LEVEL_CALCULATOR (gint16, 15);
392 DEFINE_INT_LEVEL_CALCULATOR (gint8, 7);
394 /* FIXME: use orc to calculate squaresums? */
395 #define DEFINE_FLOAT_LEVEL_CALCULATOR(TYPE) \
397 gst_level_calculate_##TYPE (gpointer data, guint num, guint channels, \
398 gdouble *NCS, gdouble *NPS) \
400 TYPE * in = (TYPE *)data; \
402 gdouble squaresum = 0.0; /* square sum of the input samples */ \
403 register gdouble square = 0.0; /* Square */ \
404 register gdouble peaksquare = 0.0; /* Peak Square Sample */ \
406 /* *NCS = 0.0; Normalized Cumulative Square */ \
407 /* *NPS = 0.0; Normalized Peak Square */ \
409 /* orc_level_squaresum_f64(&squaresum,in,num); */ \
410 for (j = 0; j < num; j += channels) { \
411 square = ((gdouble) in[j]) * in[j]; \
412 if (square > peaksquare) peaksquare = square; \
413 squaresum += square; \
420 DEFINE_FLOAT_LEVEL_CALCULATOR (gfloat);
421 DEFINE_FLOAT_LEVEL_CALCULATOR (gdouble);
423 /* we would need stride to deinterleave also
425 gst_level_calculate_gdouble (gpointer data, guint num, guint channels,
426 gdouble *NCS, gdouble *NPS)
428 orc_level_squaresum_f64(NCS,(gdouble *)data,num);
435 gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
437 GstLevel *filter = GST_LEVEL (trans);
439 gint i, channels, rate;
441 if (!gst_audio_info_from_caps (&info, in))
444 switch (GST_AUDIO_INFO_FORMAT (&info)) {
445 case GST_AUDIO_FORMAT_S8:
446 filter->process = gst_level_calculate_gint8;
448 case GST_AUDIO_FORMAT_S16:
449 filter->process = gst_level_calculate_gint16;
451 case GST_AUDIO_FORMAT_S32:
452 filter->process = gst_level_calculate_gint32;
454 case GST_AUDIO_FORMAT_F32:
455 filter->process = gst_level_calculate_gfloat;
457 case GST_AUDIO_FORMAT_F64:
458 filter->process = gst_level_calculate_gdouble;
461 filter->process = NULL;
467 channels = GST_AUDIO_INFO_CHANNELS (&info);
468 rate = GST_AUDIO_INFO_RATE (&info);
470 /* allocate channel variable arrays */
472 g_free (filter->peak);
473 g_free (filter->last_peak);
474 g_free (filter->decay_peak);
475 g_free (filter->decay_peak_base);
476 g_free (filter->decay_peak_age);
477 filter->CS = g_new (gdouble, channels);
478 filter->peak = g_new (gdouble, channels);
479 filter->last_peak = g_new (gdouble, channels);
480 filter->decay_peak = g_new (gdouble, channels);
481 filter->decay_peak_base = g_new (gdouble, channels);
483 filter->decay_peak_age = g_new (GstClockTime, channels);
485 for (i = 0; i < channels; ++i) {
486 filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
487 filter->decay_peak[i] = filter->decay_peak_base[i] = 0.0;
488 filter->decay_peak_age[i] = G_GUINT64_CONSTANT (0);
491 filter->interval_frames = GST_CLOCK_TIME_TO_FRAMES (filter->interval, rate);
497 gst_level_start (GstBaseTransform * trans)
499 GstLevel *filter = GST_LEVEL (trans);
501 filter->num_frames = 0;
502 filter->message_ts = GST_CLOCK_TIME_NONE;
508 gst_level_message_new (GstLevel * level, GstClockTime timestamp,
509 GstClockTime duration)
511 GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (level);
514 GstClockTime endtime, running_time, stream_time;
516 running_time = gst_segment_to_running_time (&trans->segment, GST_FORMAT_TIME,
518 stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
520 /* endtime is for backwards compatibility */
521 endtime = stream_time + duration;
523 s = gst_structure_new ("level",
524 "endtime", GST_TYPE_CLOCK_TIME, endtime,
525 "timestamp", G_TYPE_UINT64, timestamp,
526 "stream-time", G_TYPE_UINT64, stream_time,
527 "running-time", G_TYPE_UINT64, running_time,
528 "duration", G_TYPE_UINT64, duration, NULL);
530 g_value_init (&v, G_TYPE_VALUE_ARRAY);
531 g_value_take_boxed (&v, g_value_array_new (0));
532 gst_structure_take_value (s, "rms", &v);
534 g_value_init (&v, G_TYPE_VALUE_ARRAY);
535 g_value_take_boxed (&v, g_value_array_new (0));
536 gst_structure_take_value (s, "peak", &v);
538 g_value_init (&v, G_TYPE_VALUE_ARRAY);
539 g_value_take_boxed (&v, g_value_array_new (0));
540 gst_structure_take_value (s, "decay", &v);
542 return gst_message_new_element (GST_OBJECT (level), s);
546 gst_level_message_append_channel (GstMessage * m, gdouble rms, gdouble peak,
549 const GValue *array_val;
554 g_value_init (&v, G_TYPE_DOUBLE);
556 s = (GstStructure *) gst_message_get_structure (m);
558 array_val = gst_structure_get_value (s, "rms");
559 arr = (GValueArray *) g_value_get_boxed (array_val);
560 g_value_set_double (&v, rms);
561 g_value_array_append (arr, &v); /* copies by value */
563 array_val = gst_structure_get_value (s, "peak");
564 arr = (GValueArray *) g_value_get_boxed (array_val);
565 g_value_set_double (&v, peak);
566 g_value_array_append (arr, &v); /* copies by value */
568 array_val = gst_structure_get_value (s, "decay");
569 arr = (GValueArray *) g_value_get_boxed (array_val);
570 g_value_set_double (&v, decay);
571 g_value_array_append (arr, &v); /* copies by value */
577 gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
586 guint num_int_samples = 0; /* number of interleaved samples
587 * ie. total count for all channels combined */
588 guint block_size, block_int_size; /* we subdivide buffers to not skip message
590 GstClockTimeDiff falloff_time;
591 gint channels, rate, bps;
593 filter = GST_LEVEL (trans);
595 channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
596 bps = GST_AUDIO_INFO_BPS (&filter->info);
597 rate = GST_AUDIO_INFO_RATE (&filter->info);
599 gst_buffer_map (in, &map, GST_MAP_READ);
603 num_int_samples = in_size / bps;
605 GST_LOG_OBJECT (filter, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
606 num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (in)));
608 g_return_val_if_fail (num_int_samples % channels == 0, GST_FLOW_ERROR);
610 if (GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_DISCONT)) {
611 filter->message_ts = GST_BUFFER_TIMESTAMP (in);
612 filter->num_frames = 0;
614 if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (filter->message_ts))) {
615 filter->message_ts = GST_BUFFER_TIMESTAMP (in);
618 num_frames = num_int_samples / channels;
619 while (num_frames > 0) {
620 block_size = filter->interval_frames - filter->num_frames;
621 block_size = MIN (block_size, num_frames);
622 block_int_size = block_size * channels;
624 for (i = 0; i < channels; ++i) {
625 if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) {
626 filter->process (in_data, block_int_size, channels, &CS,
628 GST_LOG_OBJECT (filter,
629 "[%d]: cumulative squares %lf, over %d samples/%d channels",
630 i, CS, block_int_size, channels);
633 filter->peak[i] = 0.0;
637 filter->decay_peak_age[i] += GST_FRAMES_TO_CLOCK_TIME (num_frames, rate);
638 GST_LOG_OBJECT (filter,
639 "[%d]: peak %f, last peak %f, decay peak %f, age %" GST_TIME_FORMAT,
640 i, filter->peak[i], filter->last_peak[i], filter->decay_peak[i],
641 GST_TIME_ARGS (filter->decay_peak_age[i]));
643 /* update running peak */
644 if (filter->peak[i] > filter->last_peak[i])
645 filter->last_peak[i] = filter->peak[i];
647 /* make decay peak fall off if too old */
649 GST_CLOCK_DIFF (gst_gdouble_to_guint64 (filter->decay_peak_ttl),
650 filter->decay_peak_age[i]);
651 if (falloff_time > 0) {
654 gdouble length; /* length of falloff time in seconds */
656 length = (gdouble) falloff_time / (gdouble) GST_SECOND;
657 falloff_dB = filter->decay_peak_falloff * length;
658 falloff = pow (10, falloff_dB / -20.0);
660 GST_LOG_OBJECT (filter,
661 "falloff: current %f, base %f, interval %" GST_TIME_FORMAT
662 ", dB falloff %f, factor %e",
663 filter->decay_peak[i], filter->decay_peak_base[i],
664 GST_TIME_ARGS (falloff_time), falloff_dB, falloff);
665 filter->decay_peak[i] = filter->decay_peak_base[i] * falloff;
666 GST_LOG_OBJECT (filter,
667 "peak is %" GST_TIME_FORMAT " old, decayed with factor %e to %f",
668 GST_TIME_ARGS (filter->decay_peak_age[i]), falloff,
669 filter->decay_peak[i]);
671 GST_LOG_OBJECT (filter, "peak not old enough, not decaying");
674 /* if the peak of this run is higher, the decay peak gets reset */
675 if (filter->peak[i] >= filter->decay_peak[i]) {
676 GST_LOG_OBJECT (filter, "new peak, %f", filter->peak[i]);
677 filter->decay_peak[i] = filter->peak[i];
678 filter->decay_peak_base[i] = filter->peak[i];
679 filter->decay_peak_age[i] = G_GINT64_CONSTANT (0);
682 in_data += ((block_int_size * bps) - bps);
684 filter->num_frames += block_size;
685 num_frames -= block_size;
687 /* do we need to message ? */
688 if (filter->num_frames >= filter->interval_frames) {
689 gst_level_post_message (filter);
693 gst_buffer_unmap (in, &map);
699 gst_level_post_message (GstLevel * filter)
702 gint channels, rate, frames = filter->num_frames;
703 GstClockTime duration;
705 channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
706 rate = GST_AUDIO_INFO_RATE (&filter->info);
707 duration = GST_FRAMES_TO_CLOCK_TIME (frames, rate);
709 if (filter->post_messages) {
711 gst_level_message_new (filter, filter->message_ts, duration);
713 GST_LOG_OBJECT (filter,
714 "message: ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
715 ", num_frames %d", GST_TIME_ARGS (filter->message_ts),
716 GST_TIME_ARGS (duration), frames);
718 for (i = 0; i < channels; ++i) {
720 gdouble RMSdB, peakdB, decaydB;
722 RMS = sqrt (filter->CS[i] / frames);
723 GST_LOG_OBJECT (filter,
724 "message: channel %d, CS %f, RMS %f", i, filter->CS[i], RMS);
725 GST_LOG_OBJECT (filter,
726 "message: last_peak: %f, decay_peak: %f",
727 filter->last_peak[i], filter->decay_peak[i]);
728 /* RMS values are calculated in amplitude, so 20 * log 10 */
729 RMSdB = 20 * log10 (RMS + EPSILON);
730 /* peak values are square sums, ie. power, so 10 * log 10 */
731 peakdB = 10 * log10 (filter->last_peak[i] + EPSILON);
732 decaydB = 10 * log10 (filter->decay_peak[i] + EPSILON);
734 if (filter->decay_peak[i] < filter->last_peak[i]) {
735 /* this can happen in certain cases, for example when
736 * the last peak is between decay_peak and decay_peak_base */
737 GST_DEBUG_OBJECT (filter,
738 "message: decay peak dB %f smaller than last peak dB %f, copying",
740 filter->decay_peak[i] = filter->last_peak[i];
742 GST_LOG_OBJECT (filter,
743 "message: RMS %f dB, peak %f dB, decay %f dB",
744 RMSdB, peakdB, decaydB);
746 gst_level_message_append_channel (m, RMSdB, peakdB, decaydB);
748 /* reset cumulative and normal peak */
750 filter->last_peak[i] = 0.0;
753 gst_element_post_message (GST_ELEMENT (filter), m);
756 filter->num_frames -= frames;
757 filter->message_ts += duration;
762 gst_level_sink_event (GstBaseTransform * trans, GstEvent * event)
764 if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
765 GstLevel *filter = GST_LEVEL (trans);
767 gst_level_post_message (filter);
770 return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
774 plugin_init (GstPlugin * plugin)
776 return gst_element_register (plugin, "level", GST_RANK_NONE, GST_TYPE_LEVEL);
779 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
782 "Audio level plugin",
783 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);