/* GStreamer * Copyright (C) <1999> Erik Walthinsen * Copyright (C) 2000,2001,2002,2003,2005 * Thomas Vander Stichele * * 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 * message property is #TRUE, * generates an element message named * "level": * after each interval of time given by the * interval property. * The message's structure contains four fields: * * * * #GstClockTime * "endtime": * the end time of the buffer that triggered the message * * * * * #GstValueList of #gdouble * "peak": * the peak power level in dB for each channel * * * * * #GstValueList 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 the time to live. * When the decaying peak level drops, it does so at the decay rate * as specified by the * the peak falloff rate. * * * * * #GstValueList of #gdouble * "rms": * the Root Mean Square (or average power) level in dB for each channel * * * * * Example application * * * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include "gstlevel.h" GST_DEBUG_CATEGORY_STATIC (level_debug); #define GST_CAT_DEFAULT level_debug static const GstElementDetails level_details = GST_ELEMENT_DETAILS ("Level", "Filter/Analyzer/Audio", "RMS/Peak/Decaying Peak Level messager for audio/raw", "Thomas Vander Stichele "); 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 }, " "depth = (int) { 8, 16 }, " "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 }, " "depth = (int) { 8, 16 }, " "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_pad_template (element_class, gst_static_pad_template_get (&sink_template_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template_factory)); gst_element_class_set_details (element_class, &level_details); } 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", "mesage", "Post a level message for each passed interval", TRUE, G_PARAM_READWRITE)); 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_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_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)); 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); 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) (1 << (RESOLUTION * 2)); \ \ /* 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 / normalizer; \ *NPS = peaksquare / normalizer; \ } 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); 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; int 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; } } 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 (double, filter->channels); filter->peak = g_new (double, filter->channels); filter->last_peak = g_new (double, filter->channels); filter->decay_peak = g_new (double, filter->channels); filter->decay_peak_base = g_new (double, 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_GINT64_CONSTANT (0); } 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 * l, GstClockTime endtime) { GstStructure *s; GValue v = { 0, }; g_value_init (&v, GST_TYPE_LIST); s = gst_structure_new ("level", "endtime", GST_TYPE_CLOCK_TIME, endtime, 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 (l), 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; double CS; guint i; guint num_frames = 0; guint num_int_samples = 0; /* number of interleaved samples * ie. total count for all channels combined */ 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 */ if (gst_guint64_to_gdouble (filter->decay_peak_age[i]) > filter->decay_peak_ttl) { double falloff_dB; double falloff; GstClockTimeDiff falloff_time; double length; /* length of falloff time in seconds */ falloff_time = GST_CLOCK_DIFF (filter->decay_peak_ttl, gst_guint64_to_gdouble (filter->decay_peak_age[i])); length = (gdouble) falloff_time / 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); } } filter->num_frames += num_frames; /* do we need to message ? */ if (filter->num_frames >= GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate)) { if (filter->message) { GstMessage *m; GstClockTime endtime = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (in)) + GST_FRAMES_TO_CLOCK_TIME (num_frames, filter->rate); m = gst_level_message_new (filter, endtime); GST_LOG_OBJECT (filter, "message: end time %" GST_TIME_FORMAT ", num_frames %d", GST_TIME_ARGS (endtime), filter->num_frames); for (i = 0; i < filter->channels; ++i) { double RMS; double 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); /* peak values are square sums, ie. power, so 10 * log 10 */ lastdB = 10 * log10 (filter->last_peak[i]); decaydB = 10 * log10 (filter->decay_peak[i]); 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) { 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);