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gstreamer/gst/audiofx/audioamplify.c
Ravi Kiran K N fd6a5a5d90 audiofx: fix typo in example pipelines 
Fix typo in example pipelines

https://bugzilla.gnome.org/show_bug.cgi?id=748022
2015-04-17 09:53:46 +01:00

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/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
* Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
*
* 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-audioamplify
*
* Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
* The difference between the clipping modes is best evaluated by testing.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-negative ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-positive ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include "audioamplify.h"
#define GST_CAT_DEFAULT gst_audio_amplify_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_AMPLIFICATION,
PROP_CLIPPING_METHOD
};
enum
{
METHOD_CLIP = 0,
METHOD_WRAP_NEGATIVE,
METHOD_WRAP_POSITIVE,
METHOD_NOCLIP,
NUM_METHODS
};
#define GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD (gst_audio_amplify_clipping_method_get_type ())
static GType
gst_audio_amplify_clipping_method_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{METHOD_CLIP, "Normal clipping (default)", "clip"},
{METHOD_WRAP_NEGATIVE,
"Push overdriven values back from the opposite side",
"wrap-negative"},
{METHOD_WRAP_POSITIVE, "Push overdriven values back from the same side",
"wrap-positive"},
{METHOD_NOCLIP, "No clipping", "none"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioAmplifyClippingMethod", values);
}
return gtype;
}
#define ALLOWED_CAPS \
"audio/x-raw," \
" format=(string) {S8,"GST_AUDIO_NE(S16)","GST_AUDIO_NE(S32)"," \
GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}," \
" rate=(int)[1,MAX]," \
" channels=(int)[1,MAX], " \
" layout=(string) {interleaved, non-interleaved}"
G_DEFINE_TYPE (GstAudioAmplify, gst_audio_amplify, GST_TYPE_AUDIO_FILTER);
static gboolean gst_audio_amplify_set_process_function (GstAudioAmplify *
filter, gint clipping, GstAudioFormat format);
static void gst_audio_amplify_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_amplify_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_amplify_setup (GstAudioFilter * filter,
const GstAudioInfo * info);
static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
#define MIN_gint8 G_MININT8
#define MAX_gint8 G_MAXINT8
#define MIN_gint16 G_MININT16
#define MAX_gint16 G_MAXINT16
#define MIN_gint32 G_MININT32
#define MAX_gint32 G_MAXINT32
#define MAKE_INT_FUNCS(type,largetype) \
static void \
gst_audio_amplify_transform_##type##_clip (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
largetype val = *d * filter->amplification; \
*d++ = CLAMP (val, MIN_##type, MAX_##type); \
} \
} \
static void \
gst_audio_amplify_transform_##type##_wrap_negative (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
largetype val = *d * filter->amplification; \
if (val > MAX_##type) \
val = MIN_##type + (val - MIN_##type) % ((largetype) MAX_##type + 1 - \
MIN_##type); \
else if (val < MIN_##type) \
val = MAX_##type - (MAX_##type - val) % ((largetype) MAX_##type + 1 - \
MIN_##type); \
*d++ = val; \
} \
} \
static void \
gst_audio_amplify_transform_##type##_wrap_positive (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
largetype val = *d * filter->amplification; \
do { \
if (val > MAX_##type) \
val = MAX_##type - (val - MAX_##type); \
else if (val < MIN_##type) \
val = MIN_##type + (MIN_##type - val); \
else \
break; \
} while (1); \
*d++ = val; \
} \
} \
static void \
gst_audio_amplify_transform_##type##_noclip (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) \
*d++ *= filter->amplification; \
}
#define MAKE_FLOAT_FUNCS(type) \
static void \
gst_audio_amplify_transform_##type##_clip (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
type val = *d* filter->amplification; \
*d++ = CLAMP (val, -1.0, +1.0); \
} \
} \
static void \
gst_audio_amplify_transform_##type##_wrap_negative (GstAudioAmplify * \
filter, void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
type val = *d * filter->amplification; \
do { \
if (val > 1.0) \
val = -1.0 + (val - 1.0); \
else if (val < -1.0) \
val = 1.0 - (1.0 - val); \
else \
break; \
} while (1); \
*d++ = val; \
} \
} \
static void \
gst_audio_amplify_transform_##type##_wrap_positive (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) { \
type val = *d* filter->amplification; \
do { \
if (val > 1.0) \
val = 1.0 - (val - 1.0); \
else if (val < -1.0) \
val = -1.0 + (-1.0 - val); \
else \
break; \
} while (1); \
*d++ = val; \
} \
} \
static void \
gst_audio_amplify_transform_##type##_noclip (GstAudioAmplify * filter, \
void * data, guint num_samples) \
{ \
type *d = data; \
\
while (num_samples--) \
*d++ *= filter->amplification; \
}
/* *INDENT-OFF* */
MAKE_INT_FUNCS (gint8,gint)
MAKE_INT_FUNCS (gint16,gint)
MAKE_INT_FUNCS (gint32,gint64)
MAKE_FLOAT_FUNCS (gfloat)
MAKE_FLOAT_FUNCS (gdouble)
/* *INDENT-ON* */
/* GObject vmethod implementations */
static void
gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0,
"audioamplify element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audio_amplify_set_property;
gobject_class->get_property = gst_audio_amplify_get_property;
g_object_class_install_property (gobject_class, PROP_AMPLIFICATION,
g_param_spec_float ("amplification", "Amplification",
"Factor of amplification", -G_MAXFLOAT, G_MAXFLOAT,
1.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioAmplify:clipping-method
*
* Clipping method: clip mode set values higher than the maximum to the
* maximum. The wrap-negative mode pushes those values back from the
* opposite side, wrap-positive pushes them back from the same side.
*
**/
g_object_class_install_property (gobject_class, PROP_CLIPPING_METHOD,
g_param_spec_enum ("clipping-method", "Clipping method",
"Selects how to handle values higher than the maximum",
GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (gstelement_class, "Audio amplifier",
"Filter/Effect/Audio",
"Amplifies an audio stream by a given factor",
"Sebastian Dröge <slomo@circular-chaos.org>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_amplify_setup);
}
static void
gst_audio_amplify_init (GstAudioAmplify * filter)
{
filter->amplification = 1.0;
gst_audio_amplify_set_process_function (filter, METHOD_CLIP,
GST_AUDIO_FORMAT_S16);
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
}
static GstAudioAmplifyProcessFunc
gst_audio_amplify_process_function (gint clipping, GstAudioFormat format)
{
static const struct process
{
GstAudioFormat format;
gint clipping;
GstAudioAmplifyProcessFunc func;
} process[] = {
{
GST_AUDIO_FORMAT_F32, METHOD_CLIP, gst_audio_amplify_transform_gfloat_clip}, {
GST_AUDIO_FORMAT_F32, METHOD_WRAP_NEGATIVE,
gst_audio_amplify_transform_gfloat_wrap_negative}, {
GST_AUDIO_FORMAT_F32, METHOD_WRAP_POSITIVE,
gst_audio_amplify_transform_gfloat_wrap_positive}, {
GST_AUDIO_FORMAT_F32, METHOD_NOCLIP,
gst_audio_amplify_transform_gfloat_noclip}, {
GST_AUDIO_FORMAT_F64, METHOD_CLIP,
gst_audio_amplify_transform_gdouble_clip}, {
GST_AUDIO_FORMAT_F64, METHOD_WRAP_NEGATIVE,
gst_audio_amplify_transform_gdouble_wrap_negative}, {
GST_AUDIO_FORMAT_F64, METHOD_WRAP_POSITIVE,
gst_audio_amplify_transform_gdouble_wrap_positive}, {
GST_AUDIO_FORMAT_F64, METHOD_NOCLIP,
gst_audio_amplify_transform_gdouble_noclip}, {
GST_AUDIO_FORMAT_S8, METHOD_CLIP, gst_audio_amplify_transform_gint8_clip}, {
GST_AUDIO_FORMAT_S8, METHOD_WRAP_NEGATIVE,
gst_audio_amplify_transform_gint8_wrap_negative}, {
GST_AUDIO_FORMAT_S8, METHOD_WRAP_POSITIVE,
gst_audio_amplify_transform_gint8_wrap_positive}, {
GST_AUDIO_FORMAT_S8, METHOD_NOCLIP,
gst_audio_amplify_transform_gint8_noclip}, {
GST_AUDIO_FORMAT_S16, METHOD_CLIP, gst_audio_amplify_transform_gint16_clip}, {
GST_AUDIO_FORMAT_S16, METHOD_WRAP_NEGATIVE,
gst_audio_amplify_transform_gint16_wrap_negative}, {
GST_AUDIO_FORMAT_S16, METHOD_WRAP_POSITIVE,
gst_audio_amplify_transform_gint16_wrap_positive}, {
GST_AUDIO_FORMAT_S16, METHOD_NOCLIP,
gst_audio_amplify_transform_gint16_noclip}, {
GST_AUDIO_FORMAT_S32, METHOD_CLIP, gst_audio_amplify_transform_gint32_clip}, {
GST_AUDIO_FORMAT_S32, METHOD_WRAP_NEGATIVE,
gst_audio_amplify_transform_gint32_wrap_negative}, {
GST_AUDIO_FORMAT_S32, METHOD_WRAP_POSITIVE,
gst_audio_amplify_transform_gint32_wrap_positive}, {
GST_AUDIO_FORMAT_S32, METHOD_NOCLIP,
gst_audio_amplify_transform_gint32_noclip}, {
0, 0, NULL}
};
const struct process *p;
for (p = process; p->func; p++)
if (p->format == format && p->clipping == clipping)
return p->func;
return NULL;
}
static gboolean
gst_audio_amplify_set_process_function (GstAudioAmplify * filter, gint
clipping_method, GstAudioFormat format)
{
GstAudioAmplifyProcessFunc process;
/* set processing function */
process = gst_audio_amplify_process_function (clipping_method, format);
if (!process) {
GST_DEBUG ("wrong format");
return FALSE;
}
filter->process = process;
filter->clipping_method = clipping_method;
filter->format = format;
return TRUE;
}
static void
gst_audio_amplify_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
switch (prop_id) {
case PROP_AMPLIFICATION:
filter->amplification = g_value_get_float (value);
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
filter->amplification == 1.0);
break;
case PROP_CLIPPING_METHOD:
gst_audio_amplify_set_process_function (filter, g_value_get_enum (value),
filter->format);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_amplify_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
switch (prop_id) {
case PROP_AMPLIFICATION:
g_value_set_float (value, filter->amplification);
break;
case PROP_CLIPPING_METHOD:
g_value_set_enum (value, filter->clipping_method);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_amplify_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
return gst_audio_amplify_set_process_function (filter,
filter->clipping_method, GST_AUDIO_INFO_FORMAT (info));
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
guint num_samples;
GstClockTime timestamp, stream_time;
GstMapInfo map;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (filter), stream_time);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
filter->process (filter, map.data, num_samples);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
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