/* 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 * @title: level * * Level analyses incoming audio buffers and, if the #GstLevel:message property * is %TRUE, generates an element message named * `level`: after each interval of time given by the #GstLevel:interval property. * The message's structure contains these fields: * * * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message. * * #GstClockTime `stream-time`: the stream time of the buffer. * * #GstClockTime `running-time`: the running_time of the buffer. * * #GstClockTime `duration`: the duration of the buffer. * * #GstClockTime `endtime`: 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) * * #GValueArray of #gdouble `peak`: the peak power level in dB for each channel * * #GValueArray of #gdouble `decay`: 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. * * #GValueArray of #gdouble `rms`: the Root Mean Square (or average power) level in dB * for each channel * * ## Example application * * {{ tests/examples/level/level-example.c }} * */ #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, PROP_AUDIO_LEVEL_META, }; #define gst_level_parent_class parent_class G_DEFINE_TYPE (GstLevel, gst_level, GST_TYPE_BASE_TRANSFORM); GST_ELEMENT_REGISTER_DEFINE (level, "level", GST_RANK_NONE, GST_TYPE_LEVEL); 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)); /** * GstLevel:audio-level-meta: * * If %TRUE, generate or update GstAudioLevelMeta on output buffers. * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_AUDIO_LEVEL_META, g_param_spec_boolean ("audio-level-meta", "Audio Level Meta", "Set GstAudioLevelMeta on buffers", FALSE, 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; case PROP_AUDIO_LEVEL_META: filter->audio_level_meta = g_value_get_boolean (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; case PROP_AUDIO_LEVEL_META: g_value_set_boolean (value, filter->audio_level_meta); 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-existent 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 void gst_level_rtp_audio_level_meta (GstLevel * self, GstBuffer * buffer, guint8 level) { GstAudioLevelMeta *meta; /* Update the existing meta, if any, so we can have an upstream element * filling the voice activity part of the meta. */ meta = gst_buffer_get_audio_level_meta (buffer); if (meta) { meta->level = level; } else { /* Assume audio does not contain voice, it can be detected by another * downstream element. */ gst_buffer_add_audio_level_meta (buffer, level, FALSE); } } 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; gdouble CS_tot = 0; /* Total Cumulative Square on all samples */ 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]); CS_tot += CS; 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); if (filter->audio_level_meta) { gdouble RMS = sqrt (CS_tot / num_int_samples); gdouble RMSdB = 20 * log10 (RMS + EPSILON); gst_level_rtp_audio_level_meta (filter, in, -RMSdB); } 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 (level, plugin); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, level, "Audio level plugin", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);