--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audioamplify.c Wed Mar 24 18:04:17 2010 -0500
@@ -0,0 +1,495 @@
+/*
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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.
+ * <title>Example launch line</title>
+ * <refsect2>
+ * |[
+ * gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
+ * gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 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 <gst/controller/gstcontroller.h>
+
+#include "audioamplify.h"
+
+#define GST_CAT_DEFAULT gst_audio_amplify_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("Audio amplifier",
+ "Filter/Effect/Audio",
+ "Amplifies an audio stream by a given factor",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* 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}
+ };
+
+ /* FIXME 0.11: rename to GstAudioAmplifyClippingMethod */
+ gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values);
+ }
+ return gtype;
+}
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-int," \
+ " depth=(int)8," \
+ " width=(int)8," \
+ " endianness=(int)BYTE_ORDER," \
+ " signed=(bool)TRUE," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX]; " \
+ "audio/x-raw-int," \
+ " depth=(int)16," \
+ " width=(int)16," \
+ " endianness=(int)BYTE_ORDER," \
+ " signed=(bool)TRUE," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX]; " \
+ "audio/x-raw-int," \
+ " depth=(int)32," \
+ " width=(int)32," \
+ " endianness=(int)BYTE_ORDER," \
+ " signed=(bool)TRUE," \
+ " 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]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element");
+
+GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstAudioFilter,
+ GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static gboolean gst_audio_amplify_set_process_function (GstAudioAmplify *
+ filter, gint clipping, gint format, gint width);
+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,
+ GstRingBufferSpec * format);
+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_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = (GObjectClass *) 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", 0.0, G_MAXFLOAT,
+ 1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ /**
+ * 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));
+
+ GST_AUDIO_FILTER_CLASS (klass)->setup =
+ GST_DEBUG_FUNCPTR (gst_audio_amplify_setup);
+ GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip);
+}
+
+static void
+gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass)
+{
+ filter->amplification = 1.0;
+ gst_audio_amplify_set_process_function (filter, METHOD_CLIP,
+ GST_BUFTYPE_LINEAR, 16);
+ 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, gint format, gint width)
+{
+ static const struct process
+ {
+ gint format;
+ gint width;
+ gint clipping;
+ GstAudioAmplifyProcessFunc func;
+ } process[] = {
+ {
+ GST_BUFTYPE_FLOAT, 32, METHOD_CLIP,
+ gst_audio_amplify_transform_gfloat_clip}, {
+ GST_BUFTYPE_FLOAT, 32, METHOD_WRAP_NEGATIVE,
+ gst_audio_amplify_transform_gfloat_wrap_negative}, {
+ GST_BUFTYPE_FLOAT, 32, METHOD_WRAP_POSITIVE,
+ gst_audio_amplify_transform_gfloat_wrap_positive}, {
+ GST_BUFTYPE_FLOAT, 32, METHOD_NOCLIP,
+ gst_audio_amplify_transform_gfloat_noclip}, {
+ GST_BUFTYPE_FLOAT, 64, METHOD_CLIP,
+ gst_audio_amplify_transform_gdouble_clip}, {
+ GST_BUFTYPE_FLOAT, 64, METHOD_WRAP_NEGATIVE,
+ gst_audio_amplify_transform_gdouble_wrap_negative}, {
+ GST_BUFTYPE_FLOAT, 64, METHOD_WRAP_POSITIVE,
+ gst_audio_amplify_transform_gdouble_wrap_positive}, {
+ GST_BUFTYPE_FLOAT, 64, METHOD_NOCLIP,
+ gst_audio_amplify_transform_gdouble_noclip}, {
+ GST_BUFTYPE_LINEAR, 8, METHOD_CLIP, gst_audio_amplify_transform_gint8_clip}, {
+ GST_BUFTYPE_LINEAR, 8, METHOD_WRAP_NEGATIVE,
+ gst_audio_amplify_transform_gint8_wrap_negative}, {
+ GST_BUFTYPE_LINEAR, 8, METHOD_WRAP_POSITIVE,
+ gst_audio_amplify_transform_gint8_wrap_positive}, {
+ GST_BUFTYPE_LINEAR, 8, METHOD_NOCLIP,
+ gst_audio_amplify_transform_gint8_noclip}, {
+ GST_BUFTYPE_LINEAR, 16, METHOD_CLIP,
+ gst_audio_amplify_transform_gint16_clip}, {
+ GST_BUFTYPE_LINEAR, 16, METHOD_WRAP_NEGATIVE,
+ gst_audio_amplify_transform_gint16_wrap_negative}, {
+ GST_BUFTYPE_LINEAR, 16, METHOD_WRAP_POSITIVE,
+ gst_audio_amplify_transform_gint16_wrap_positive}, {
+ GST_BUFTYPE_LINEAR, 16, METHOD_NOCLIP,
+ gst_audio_amplify_transform_gint16_noclip}, {
+ GST_BUFTYPE_LINEAR, 32, METHOD_CLIP,
+ gst_audio_amplify_transform_gint32_clip}, {
+ GST_BUFTYPE_LINEAR, 32, METHOD_WRAP_NEGATIVE,
+ gst_audio_amplify_transform_gint32_wrap_negative}, {
+ GST_BUFTYPE_LINEAR, 32, METHOD_WRAP_POSITIVE,
+ gst_audio_amplify_transform_gint32_wrap_positive}, {
+ GST_BUFTYPE_LINEAR, 32, METHOD_NOCLIP,
+ gst_audio_amplify_transform_gint32_noclip}, {
+ 0, 0, 0, NULL}
+ };
+ const struct process *p;
+
+ for (p = process; p->func; p++)
+ if (p->format == format && p->width == width && p->clipping == clipping)
+ return p->func;
+ return NULL;
+}
+
+static gboolean
+gst_audio_amplify_set_process_function (GstAudioAmplify * filter, gint
+ clipping_method, gint format, gint width)
+{
+ GstAudioAmplifyProcessFunc process;
+
+ /* set processing function */
+
+ process = gst_audio_amplify_process_function (clipping_method, format, width);
+ if (!process) {
+ GST_DEBUG ("wrong format");
+ return FALSE;
+ }
+
+ filter->process = process;
+ filter->clipping_method = clipping_method;
+ filter->format = format;
+ filter->width = width;
+
+ 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, filter->width);
+ 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, GstRingBufferSpec * format)
+{
+ GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
+
+ return gst_audio_amplify_set_process_function (filter,
+ filter->clipping_method, format->type, format->width);
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+ GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (gst_base_transform_is_passthrough (base) ||
+ G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
+ return GST_FLOW_OK;
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}