gstreamer/gst/spectrum/gstspectrum.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* <2006> Stefan Kost <ensonic@users.sf.net>
* <2007-2009> Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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-spectrum
*
* The Spectrum element analyzes the frequency spectrum of an audio signal.
* If the #GstSpectrum:message property is #TRUE, it sends analysis results as
* application messages named
* <classname>&quot;spectrum&quot;</classname> after each interval of time given
* by the #GstSpectrum:interval property.
*
* The message's structure contains some combination of these fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;timestamp&quot;</classname>:
* the timestamp of the buffer that triggered the message.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;stream-time&quot;</classname>:
* the stream time of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;running-time&quot;</classname>:
* the running_time of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;duration&quot;</classname>:
* the duration of the buffer.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;endtime&quot;</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 #gfloat
* <classname>&quot;magnitude&quot;</classname>:
* the level for each frequency band in dB. All values below the value of the
* #GstSpectrum:threshold property will be set to the threshold. Only present
* if the message-magnitude property is true.
* </para>
* </listitem>
* <listitem>
* <para>
* #GstValueList of #gfloat
* <classname>&quot;phase&quot;</classname>:
* The phase for each frequency band. The value is between -pi and pi. Only
* present if the message-phase property is true.
* </para>
* </listitem>
docs/plugins/Makefile.am: Update include list. Original commit message from CVS: * docs/plugins/Makefile.am: Update include list. * docs/plugins/gst-plugins-bad-plugins-docs.sgml: Update xml includes. * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvbsrc.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstinterlace.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-switch.xml: * docs/plugins/inspect/plugin-timidity.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoparse.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: Regenerate files. * gst/spectrum/gstspectrum.c: Fix broken XML fragment in doc snippet. * tests/check/elements/.cvsignore: Add test binary to ignores.
2008-01-21 07:54:02 +00:00
* </itemizedlist>
*
* <refsect2>
* <title>Example application</title>
* |[
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
* ]|
* </refsect2>
*
* Last reviewed on 2009-01-14 (0.10.12)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include "gstspectrum.h"
GST_DEBUG_CATEGORY_STATIC (gst_spectrum_debug);
#define GST_CAT_DEFAULT gst_spectrum_debug
/* elementfactory information */
#define ALLOWED_CAPS \
"audio/x-raw-int, " \
" width = (int) 16, " \
" depth = (int) 16, " \
" signed = (boolean) true, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ]; " \
"audio/x-raw-int, " \
" width = (int) 32, " \
" depth = (int) 32, " \
" signed = (boolean) true, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ]; " \
"audio/x-raw-float, " \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ]"
/* Spectrum properties */
#define DEFAULT_MESSAGE TRUE
#define DEFAULT_MESSAGE_MAGNITUDE TRUE
#define DEFAULT_MESSAGE_PHASE FALSE
#define DEFAULT_INTERVAL (GST_SECOND / 10)
#define DEFAULT_BANDS 128
#define DEFAULT_THRESHOLD -60
enum
{
PROP_0,
PROP_MESSAGE,
PROP_MESSAGE_MAGNITUDE,
PROP_MESSAGE_PHASE,
PROP_INTERVAL,
PROP_BANDS,
PROP_THRESHOLD
};
GST_BOILERPLATE (GstSpectrum, gst_spectrum, GstAudioFilter,
GST_TYPE_AUDIO_FILTER);
static void gst_spectrum_finalize (GObject * object);
static void gst_spectrum_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_spectrum_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_spectrum_start (GstBaseTransform * trans);
static gboolean gst_spectrum_stop (GstBaseTransform * trans);
static GstFlowReturn gst_spectrum_transform_ip (GstBaseTransform * trans,
GstBuffer * in);
static gboolean gst_spectrum_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
static void
gst_spectrum_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details_simple (element_class, "Spectrum analyzer",
"Filter/Analyzer/Audio",
"Run an FFT on the audio signal, output spectrum data",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Stefan Kost <ensonic@users.sf.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_spectrum_class_init (GstSpectrumClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
gobject_class->set_property = gst_spectrum_set_property;
gobject_class->get_property = gst_spectrum_get_property;
gobject_class->finalize = gst_spectrum_finalize;
trans_class->start = GST_DEBUG_FUNCPTR (gst_spectrum_start);
trans_class->stop = GST_DEBUG_FUNCPTR (gst_spectrum_stop);
trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_spectrum_transform_ip);
trans_class->passthrough_on_same_caps = TRUE;
filter_class->setup = GST_DEBUG_FUNCPTR (gst_spectrum_setup);
g_object_class_install_property (gobject_class, PROP_MESSAGE,
g_param_spec_boolean ("message", "Message",
"Whether to post a 'spectrum' element message on the bus for each "
"passed interval", DEFAULT_MESSAGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MESSAGE_MAGNITUDE,
g_param_spec_boolean ("message-magnitude", "Magnitude",
"Whether to add a 'magnitude' field to the structure of any "
"'spectrum' element messages posted on the bus",
DEFAULT_MESSAGE_MAGNITUDE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MESSAGE_PHASE,
g_param_spec_boolean ("message-phase", "Phase",
"Whether to add a 'phase' field to the structure of any "
"'spectrum' element messages posted on the bus",
DEFAULT_MESSAGE_PHASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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, DEFAULT_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BANDS,
g_param_spec_uint ("bands", "Bands", "Number of frequency bands",
0, G_MAXUINT, DEFAULT_BANDS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_THRESHOLD,
g_param_spec_int ("threshold", "Threshold",
"dB threshold for result. All lower values will be set to this",
G_MININT, 0, DEFAULT_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (gst_spectrum_debug, "spectrum", 0,
"audio spectrum analyser element");
}
static void
gst_spectrum_init (GstSpectrum * spectrum, GstSpectrumClass * g_class)
{
spectrum->message = DEFAULT_MESSAGE;
spectrum->message_magnitude = DEFAULT_MESSAGE_MAGNITUDE;
spectrum->message_phase = DEFAULT_MESSAGE_PHASE;
spectrum->interval = DEFAULT_INTERVAL;
spectrum->bands = DEFAULT_BANDS;
spectrum->threshold = DEFAULT_THRESHOLD;
}
static void
gst_spectrum_reset_state (GstSpectrum * spectrum)
{
if (spectrum->fft_ctx)
gst_fft_f32_free (spectrum->fft_ctx);
g_free (spectrum->input);
g_free (spectrum->input_tmp);
g_free (spectrum->freqdata);
g_free (spectrum->spect_magnitude);
g_free (spectrum->spect_phase);
spectrum->fft_ctx = NULL;
spectrum->input = NULL;
spectrum->input_tmp = NULL;
spectrum->spect_magnitude = NULL;
spectrum->spect_phase = NULL;
spectrum->freqdata = NULL;
spectrum->num_frames = 0;
spectrum->num_fft = 0;
spectrum->accumulated_error = 0;
}
static void
gst_spectrum_finalize (GObject * object)
{
GstSpectrum *spectrum = GST_SPECTRUM (object);
gst_spectrum_reset_state (spectrum);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_spectrum_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSpectrum *filter = GST_SPECTRUM (object);
switch (prop_id) {
case PROP_MESSAGE:
filter->message = g_value_get_boolean (value);
break;
case PROP_MESSAGE_MAGNITUDE:
filter->message_magnitude = g_value_get_boolean (value);
break;
case PROP_MESSAGE_PHASE:
filter->message_phase = g_value_get_boolean (value);
break;
case PROP_INTERVAL:
GST_BASE_TRANSFORM_LOCK (filter);
filter->interval = g_value_get_uint64 (value);
gst_spectrum_reset_state (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_BANDS:
GST_BASE_TRANSFORM_LOCK (filter);
if (filter->bands == g_value_get_uint (value)) {
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
}
filter->bands = g_value_get_uint (value);
gst_spectrum_reset_state (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_THRESHOLD:
GST_BASE_TRANSFORM_LOCK (filter);
filter->threshold = g_value_get_int (value);
gst_spectrum_reset_state (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_spectrum_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSpectrum *filter = GST_SPECTRUM (object);
switch (prop_id) {
case PROP_MESSAGE:
g_value_set_boolean (value, filter->message);
break;
case PROP_MESSAGE_MAGNITUDE:
g_value_set_boolean (value, filter->message_magnitude);
break;
case PROP_MESSAGE_PHASE:
g_value_set_boolean (value, filter->message_phase);
break;
case PROP_INTERVAL:
g_value_set_uint64 (value, filter->interval);
break;
case PROP_BANDS:
g_value_set_uint (value, filter->bands);
break;
case PROP_THRESHOLD:
g_value_set_int (value, filter->threshold);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_spectrum_start (GstBaseTransform * trans)
{
GstSpectrum *filter = GST_SPECTRUM (trans);
filter->num_frames = 0;
filter->num_fft = 0;
return TRUE;
}
static gboolean
gst_spectrum_stop (GstBaseTransform * trans)
{
GstSpectrum *filter = GST_SPECTRUM (trans);
gst_spectrum_reset_state (filter);
return TRUE;
}
static gboolean
gst_spectrum_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstSpectrum *filter = GST_SPECTRUM (base);
gst_spectrum_reset_state (filter);
return TRUE;
}
static GstMessage *
gst_spectrum_message_new (GstSpectrum * spectrum, GstClockTime timestamp,
GstClockTime duration)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (spectrum);
GstStructure *s;
GValue v = { 0, };
GValue *l;
guint i;
gfloat *spect_magnitude = spectrum->spect_magnitude;
gfloat *spect_phase = spectrum->spect_phase;
GstClockTime endtime, running_time, stream_time;
GST_DEBUG_OBJECT (spectrum, "preparing message, spect = %p, bands =%d ",
spect_magnitude, spectrum->bands);
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 ("spectrum",
"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);
if (spectrum->message_magnitude) {
/* FIXME 0.11: this should be an array, not a list */
g_value_init (&v, GST_TYPE_LIST);
/* will copy-by-value */
gst_structure_set_value (s, "magnitude", &v);
g_value_unset (&v);
g_value_init (&v, G_TYPE_FLOAT);
l = (GValue *) gst_structure_get_value (s, "magnitude");
for (i = 0; i < spectrum->bands; i++) {
g_value_set_float (&v, spect_magnitude[i]);
gst_value_list_append_value (l, &v); /* copies by value */
}
g_value_unset (&v);
}
if (spectrum->message_phase) {
/* FIXME 0.11: this should be an array, not a list */
g_value_init (&v, GST_TYPE_LIST);
/* will copy-by-value */
gst_structure_set_value (s, "phase", &v);
g_value_unset (&v);
g_value_init (&v, G_TYPE_FLOAT);
l = (GValue *) gst_structure_get_value (s, "phase");
for (i = 0; i < spectrum->bands; i++) {
g_value_set_float (&v, spect_phase[i]);
gst_value_list_append_value (l, &v); /* copies by value */
}
g_value_unset (&v);
}
return gst_message_new_element (GST_OBJECT (spectrum), s);
}
static GstFlowReturn
gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
{
GstSpectrum *spectrum = GST_SPECTRUM (trans);
guint i;
guint rate = GST_AUDIO_FILTER (spectrum)->format.rate;
guint channels = GST_AUDIO_FILTER (spectrum)->format.channels;
guint width = GST_AUDIO_FILTER (spectrum)->format.width / 8;
gboolean fp = (GST_AUDIO_FILTER (spectrum)->format.type == GST_BUFTYPE_FLOAT);
guint bands = spectrum->bands;
guint nfft = 2 * bands - 2;
gint threshold = spectrum->threshold;
gfloat *input;
gfloat *input_tmp;
GstFFTF32Complex *freqdata;
gfloat *spect_magnitude;
gfloat *spect_phase;
GstFFTF32 *fft_ctx;
const guint8 *data = GST_BUFFER_DATA (buffer);
guint size = GST_BUFFER_SIZE (buffer);
GST_LOG_OBJECT (spectrum, "input size: %d bytes", GST_BUFFER_SIZE (buffer));
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (spectrum, "Discontinuity detected -- resetting state");
gst_spectrum_reset_state (spectrum);
}
/* If we don't have a FFT context yet get one and
* allocate memory for everything
*/
if (spectrum->fft_ctx == NULL) {
spectrum->input = g_new0 (gfloat, nfft);
spectrum->input_tmp = g_new0 (gfloat, nfft);
spectrum->freqdata = g_new0 (GstFFTF32Complex, bands);
spectrum->spect_magnitude = g_new0 (gfloat, bands);
spectrum->spect_phase = g_new0 (gfloat, bands);
spectrum->fft_ctx = gst_fft_f32_new (nfft, FALSE);
spectrum->frames_per_interval =
gst_util_uint64_scale (spectrum->interval, rate, GST_SECOND);
spectrum->error_per_interval = (spectrum->interval * rate) % GST_SECOND;
if (spectrum->frames_per_interval == 0)
spectrum->frames_per_interval = 1;
spectrum->num_frames = 0;
spectrum->num_fft = 0;
spectrum->accumulated_error = 0;
}
if (spectrum->num_frames == 0)
spectrum->message_ts = GST_BUFFER_TIMESTAMP (buffer);
input = spectrum->input;
input_tmp = spectrum->input_tmp;
freqdata = spectrum->freqdata;
spect_magnitude = spectrum->spect_magnitude;
spect_phase = spectrum->spect_phase;
fft_ctx = spectrum->fft_ctx;
while (size >= width * channels) {
/* Move the current frame into our ringbuffer and
* take the average of all channels
*/
spectrum->input[spectrum->input_pos] = 0.0;
if (fp && width == 4) {
gfloat *in = (gfloat *) data;
for (i = 0; i < channels; i++)
spectrum->input[spectrum->input_pos] += in[i];
} else if (fp && width == 8) {
gdouble *in = (gdouble *) data;
for (i = 0; i < channels; i++)
spectrum->input[spectrum->input_pos] += in[i];
} else if (!fp && width == 4) {
gint32 *in = (gint32 *) data;
for (i = 0; i < channels; i++)
spectrum->input[spectrum->input_pos] += ((gfloat) in[i]) / G_MAXINT32;
} else if (!fp && width == 2) {
gint16 *in = (gint16 *) data;
for (i = 0; i < channels; i++)
spectrum->input[spectrum->input_pos] += ((gfloat) in[i]) / G_MAXINT16;
} else {
g_assert_not_reached ();
}
spectrum->input[spectrum->input_pos] /= channels;
data += width * channels;
size -= width * channels;
spectrum->input_pos = (spectrum->input_pos + 1) % nfft;
spectrum->num_frames++;
/* If we have enough frames for an FFT or we
* have all frames required for the interval run
* an FFT. In the last case we probably take the
* FFT of frames that we already handled.
*/
if (spectrum->num_frames % nfft == 0 ||
((spectrum->accumulated_error < GST_SECOND
&& spectrum->num_frames == spectrum->frames_per_interval)
|| (spectrum->accumulated_error >= GST_SECOND
&& spectrum->num_frames - 1 ==
spectrum->frames_per_interval))) {
for (i = 0; i < nfft; i++)
input_tmp[i] = input[(spectrum->input_pos + i) % nfft];
gst_fft_f32_window (fft_ctx, input_tmp, GST_FFT_WINDOW_HAMMING);
gst_fft_f32_fft (fft_ctx, input_tmp, freqdata);
spectrum->num_fft++;
/* Calculate magnitude in db */
for (i = 0; i < bands; i++) {
gdouble val = 0.0;
val = freqdata[i].r * freqdata[i].r;
val += freqdata[i].i * freqdata[i].i;
val /= nfft * nfft;
val = 10.0 * log10 (val);
if (val < threshold)
val = threshold;
spect_magnitude[i] += val;
}
/* Calculate phase */
for (i = 0; i < bands; i++)
spect_phase[i] += atan2 (freqdata[i].i, freqdata[i].r);
}
/* Do we have the FFTs for one interval? */
if ((spectrum->accumulated_error < GST_SECOND
&& spectrum->num_frames == spectrum->frames_per_interval)
|| (spectrum->accumulated_error >= GST_SECOND
&& spectrum->num_frames - 1 == spectrum->frames_per_interval)) {
if (spectrum->accumulated_error >= GST_SECOND)
spectrum->accumulated_error -= GST_SECOND;
else
spectrum->accumulated_error += spectrum->error_per_interval;
if (spectrum->message) {
GstMessage *m;
/* Calculate average */
for (i = 0; i < bands; i++) {
spect_magnitude[i] /= spectrum->num_fft;
spect_phase[i] /= spectrum->num_fft;
}
m = gst_spectrum_message_new (spectrum, spectrum->message_ts,
spectrum->interval);
gst_element_post_message (GST_ELEMENT (spectrum), m);
}
memset (spect_magnitude, 0, bands * sizeof (gfloat));
memset (spect_phase, 0, bands * sizeof (gfloat));
if (GST_CLOCK_TIME_IS_VALID (spectrum->message_ts))
spectrum->message_ts +=
gst_util_uint64_scale (spectrum->num_frames, GST_SECOND, rate);
spectrum->num_frames = 0;
spectrum->num_fft = 0;
}
}
g_assert (size == 0);
return GST_FLOW_OK;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "spectrum", GST_RANK_NONE,
GST_TYPE_SPECTRUM);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"spectrum",
"Run an FFT on the audio signal, output spectrum data",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)