gstreamer/gst/cutter/gstcutter.c
Vineeth TM 15b08e0bd5 cutter: Fix buffer leak
Buffer is added to the internal cache, and pushed only when accumulated
buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
is not freed. Freeing the cache when the state changes from PAUSED to READY.

https://bugzilla.gnome.org/show_bug.cgi?id=754212
2015-10-05 12:03:33 +01:00

513 lines
17 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 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-cutter
*
* Analyses the audio signal for periods of silence. The start and end of
* silence is signalled by bus messages named
* <classname>&quot;cutter&quot;</classname>.
* The message's structure contains two fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;timestamp&quot;</classname>:
* the timestamp of the buffer that triggered the message.
* </para>
* </listitem>
* <listitem>
* <para>
* gboolean
* <classname>&quot;above&quot;</classname>:
* %TRUE for begin of silence and %FALSE for end of silence.
* </para>
* </listitem>
* </itemizedlist>
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 -m filesrc location=foo.ogg ! decodebin ! audioconvert ! cutter ! autoaudiosink
* ]| Show cut messages.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstcutter.h"
#include "math.h"
GST_DEBUG_CATEGORY_STATIC (cutter_debug);
#define GST_CAT_DEFAULT cutter_debug
#define CUTTER_DEFAULT_THRESHOLD_LEVEL 0.1
#define CUTTER_DEFAULT_THRESHOLD_LENGTH (500 * GST_MSECOND)
#define CUTTER_DEFAULT_PRE_LENGTH (200 * GST_MSECOND)
static GstStaticPadTemplate cutter_src_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) " }, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], "
"layout = (string) interleaved")
);
static GstStaticPadTemplate cutter_sink_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) " }, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], "
"layout = (string) interleaved")
);
enum
{
PROP_0,
PROP_THRESHOLD,
PROP_THRESHOLD_DB,
PROP_RUN_LENGTH,
PROP_PRE_LENGTH,
PROP_LEAKY
};
#define gst_cutter_parent_class parent_class
G_DEFINE_TYPE (GstCutter, gst_cutter, GST_TYPE_ELEMENT);
static GstStateChangeReturn
gst_cutter_change_state (GstElement * element, GstStateChange transition);
static void gst_cutter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_cutter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_cutter_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstFlowReturn gst_cutter_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static void
gst_cutter_class_init (GstCutterClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
gobject_class = (GObjectClass *) klass;
element_class = (GstElementClass *) klass;
gobject_class->set_property = gst_cutter_set_property;
gobject_class->get_property = gst_cutter_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_THRESHOLD,
g_param_spec_double ("threshold", "Threshold",
"Volume threshold before trigger",
-G_MAXDOUBLE, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_THRESHOLD_DB,
g_param_spec_double ("threshold-dB", "Threshold (dB)",
"Volume threshold before trigger (in dB)",
-G_MAXDOUBLE, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_RUN_LENGTH,
g_param_spec_uint64 ("run-length", "Run length",
"Length of drop below threshold before cut_stop (in nanoseconds)",
0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PRE_LENGTH,
g_param_spec_uint64 ("pre-length", "Pre-recording buffer length",
"Length of pre-recording buffer (in nanoseconds)",
0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LEAKY,
g_param_spec_boolean ("leaky", "Leaky",
"do we leak buffers when below threshold ?",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (cutter_debug, "cutter", 0, "Audio cutting");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&cutter_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&cutter_sink_factory));
gst_element_class_set_static_metadata (element_class, "Audio cutter",
"Filter/Editor/Audio",
"Audio Cutter to split audio into non-silent bits",
"Thomas Vander Stichele <thomas at apestaart dot org>");
element_class->change_state = gst_cutter_change_state;
}
static void
gst_cutter_init (GstCutter * filter)
{
filter->sinkpad =
gst_pad_new_from_static_template (&cutter_sink_factory, "sink");
gst_pad_set_chain_function (filter->sinkpad, gst_cutter_chain);
gst_pad_set_event_function (filter->sinkpad, gst_cutter_event);
gst_pad_use_fixed_caps (filter->sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
filter->srcpad =
gst_pad_new_from_static_template (&cutter_src_factory, "src");
gst_pad_use_fixed_caps (filter->srcpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
filter->threshold_level = CUTTER_DEFAULT_THRESHOLD_LEVEL;
filter->threshold_length = CUTTER_DEFAULT_THRESHOLD_LENGTH;
filter->silent_run_length = 0 * GST_SECOND;
filter->silent = TRUE;
filter->silent_prev = FALSE; /* previous value of silent */
filter->pre_length = CUTTER_DEFAULT_PRE_LENGTH;
filter->pre_run_length = 0 * GST_SECOND;
filter->pre_buffer = NULL;
filter->leaky = FALSE;
}
static GstMessage *
gst_cutter_message_new (GstCutter * c, gboolean above, GstClockTime timestamp)
{
GstStructure *s;
s = gst_structure_new ("cutter",
"above", G_TYPE_BOOLEAN, above,
"timestamp", GST_TYPE_CLOCK_TIME, timestamp, NULL);
return gst_message_new_element (GST_OBJECT (c), s);
}
/* Calculate the Normalized Cumulative Square over a buffer of the given type
* and over all channels combined */
#define DEFINE_CUTTER_CALCULATOR(TYPE, RESOLUTION) \
static void inline \
gst_cutter_calculate_##TYPE (TYPE * in, guint num, \
double *NCS) \
{ \
register int j; \
double squaresum = 0.0; /* square sum of the integer samples */ \
register double square = 0.0; /* Square */ \
gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
\
*NCS = 0.0; /* Normalized Cumulative Square */ \
\
normalizer = (double) (1 << (RESOLUTION * 2)); \
\
for (j = 0; j < num; j++) \
{ \
square = ((double) in[j]) * in[j]; \
squaresum += square; \
} \
\
\
*NCS = squaresum / normalizer; \
}
DEFINE_CUTTER_CALCULATOR (gint16, 15);
DEFINE_CUTTER_CALCULATOR (gint8, 7);
static gboolean
gst_cutter_setcaps (GstCutter * filter, GstCaps * caps)
{
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
filter->info = info;
return gst_pad_set_caps (filter->srcpad, caps);
}
static GstStateChangeReturn
gst_cutter_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstCutter *filter = GST_CUTTER (element);
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
g_list_free_full (filter->pre_buffer, (GDestroyNotify) gst_buffer_unref);
filter->pre_buffer = NULL;
break;
default:
break;
}
return ret;
}
static gboolean
gst_cutter_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean ret;
GstCutter *filter;
filter = GST_CUTTER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_cutter_setcaps (filter, caps);
gst_event_unref (event);
break;
}
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static GstFlowReturn
gst_cutter_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstCutter *filter;
GstMapInfo map;
gint16 *in_data;
gint bpf, rate;
gsize in_size;
guint num_samples;
gdouble NCS = 0.0; /* Normalized Cumulative Square of buffer */
gdouble RMS = 0.0; /* RMS of signal in buffer */
gdouble NMS = 0.0; /* Normalized Mean Square of buffer */
GstBuffer *prebuf; /* pointer to a prebuffer element */
GstClockTime duration;
filter = GST_CUTTER (parent);
if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN)
goto not_negotiated;
bpf = GST_AUDIO_INFO_BPF (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
gst_buffer_map (buf, &map, GST_MAP_READ);
in_data = (gint16 *) map.data;
in_size = map.size;
GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
/* calculate mean square value on buffer */
switch (GST_AUDIO_INFO_FORMAT (&filter->info)) {
case GST_AUDIO_FORMAT_S16:
num_samples = in_size / 2;
gst_cutter_calculate_gint16 (in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
case GST_AUDIO_FORMAT_S8:
num_samples = in_size;
gst_cutter_calculate_gint8 ((gint8 *) in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
default:
/* this shouldn't happen */
g_warning ("no mean square function for format");
break;
}
gst_buffer_unmap (buf, &map);
filter->silent_prev = filter->silent;
duration = gst_util_uint64_scale (in_size / bpf, GST_SECOND, rate);
RMS = sqrt (NMS);
/* if RMS below threshold, add buffer length to silent run length count
* if not, reset
*/
GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", NMS,
RMS, gst_guint64_to_gdouble (duration));
if (RMS < filter->threshold_level)
filter->silent_run_length += gst_guint64_to_gdouble (duration);
else {
filter->silent_run_length = 0 * GST_SECOND;
filter->silent = FALSE;
}
if (filter->silent_run_length > filter->threshold_length)
/* it has been silent long enough, flag it */
filter->silent = TRUE;
/* has the silent status changed ? if so, send right signal
* and, if from silent -> not silent, flush pre_record buffer
*/
if (filter->silent != filter->silent_prev) {
if (filter->silent) {
GstMessage *m =
gst_cutter_message_new (filter, FALSE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_STOP");
gst_element_post_message (GST_ELEMENT (filter), m);
} else {
gint count = 0;
GstMessage *m =
gst_cutter_message_new (filter, TRUE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_START");
gst_element_post_message (GST_ELEMENT (filter), m);
/* first of all, flush current buffer */
GST_DEBUG_OBJECT (filter, "flushing buffer of length %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
while (filter->pre_buffer) {
prebuf = (g_list_first (filter->pre_buffer))->data;
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
gst_pad_push (filter->srcpad, prebuf);
++count;
}
GST_DEBUG_OBJECT (filter, "flushed %d buffers", count);
filter->pre_run_length = 0 * GST_SECOND;
}
}
/* now check if we have to send the new buffer to the internal buffer cache
* or to the srcpad */
if (filter->silent) {
filter->pre_buffer = g_list_append (filter->pre_buffer, buf);
filter->pre_run_length += gst_guint64_to_gdouble (duration);
while (filter->pre_run_length > filter->pre_length) {
GstClockTime pduration;
gsize psize;
prebuf = (g_list_first (filter->pre_buffer))->data;
g_assert (GST_IS_BUFFER (prebuf));
psize = gst_buffer_get_size (prebuf);
pduration = gst_util_uint64_scale (psize / bpf, GST_SECOND, rate);
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
filter->pre_run_length -= gst_guint64_to_gdouble (pduration);
/* only pass buffers if we don't leak */
if (!filter->leaky)
ret = gst_pad_push (filter->srcpad, prebuf);
else
gst_buffer_unref (prebuf);
}
} else
ret = gst_pad_push (filter->srcpad, buf);
return ret;
/* ERRORS */
not_negotiated:
{
return GST_FLOW_NOT_NEGOTIATED;
}
}
static void
gst_cutter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstCutter *filter;
g_return_if_fail (GST_IS_CUTTER (object));
filter = GST_CUTTER (object);
switch (prop_id) {
case PROP_THRESHOLD:
filter->threshold_level = g_value_get_double (value);
GST_DEBUG ("DEBUG: set threshold level to %f", filter->threshold_level);
break;
case PROP_THRESHOLD_DB:
/* set the level given in dB
* value in dB = 20 * log (value)
* values in dB < 0 result in values between 0 and 1
*/
filter->threshold_level = pow (10, g_value_get_double (value) / 20);
GST_DEBUG_OBJECT (filter, "set threshold level to %f",
filter->threshold_level);
break;
case PROP_RUN_LENGTH:
/* set the minimum length of the silent run required */
filter->threshold_length =
gst_guint64_to_gdouble (g_value_get_uint64 (value));
break;
case PROP_PRE_LENGTH:
/* set the length of the pre-record block */
filter->pre_length = gst_guint64_to_gdouble (g_value_get_uint64 (value));
break;
case PROP_LEAKY:
/* set if the pre-record buffer is leaky or not */
filter->leaky = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_cutter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstCutter *filter;
g_return_if_fail (GST_IS_CUTTER (object));
filter = GST_CUTTER (object);
switch (prop_id) {
case PROP_RUN_LENGTH:
g_value_set_uint64 (value, filter->threshold_length);
break;
case PROP_THRESHOLD:
g_value_set_double (value, filter->threshold_level);
break;
case PROP_THRESHOLD_DB:
g_value_set_double (value, 20 * log (filter->threshold_level));
break;
case PROP_PRE_LENGTH:
g_value_set_uint64 (value, filter->pre_length);
break;
case PROP_LEAKY:
g_value_set_boolean (value, filter->leaky);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "cutter", GST_RANK_NONE, GST_TYPE_CUTTER))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
cutter,
"Audio Cutter to split audio into non-silent bits",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);