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authorjk7744.park <jk7744.park@samsung.com>2015-10-24 16:49:08 +0900
committerjk7744.park <jk7744.park@samsung.com>2015-10-24 16:49:08 +0900
commit32e864f0f32b7628d995e84d07646b01dc5bf2d5 (patch)
tree27f97eaa1ca7e171815e4f88b348448b05f02748 /gst/replaygain/gstrglimiter.c
parent9093d777f57b3f1dc62010bb8e0c7ed3d72a76d2 (diff)
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+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrglimiter.c: Element to apply signal compression to raw audio data
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rglimiter
+ * @see_also: #GstRgVolume
+ *
+ * This element applies signal compression/limiting to raw audio data. It
+ * performs strict hard limiting with soft-knee characteristics, using a
+ * threshold of -6 dB. This type of filter is mentioned in the proposed <ulink
+ * url="http://replaygain.org">ReplayGain standard</ulink>.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 filesrc location=filename.ext ! decodebin ! audioconvert \
+ * ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \
+ * ! audioconvert ! audioresample ! alsasink
+ * ]|Playback of a file
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <math.h>
+#include <gst/audio/audio.h>
+
+#include "gstrglimiter.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug);
+#define GST_CAT_DEFAULT gst_rg_limiter_debug
+
+enum
+{
+ PROP_0,
+ PROP_ENABLED,
+};
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (F32) ", "
+ "layout = (string) { interleaved, non-interleaved }, "
+ "channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (F32) ", "
+ "layout = (string) { interleaved, non-interleaved}, "
+ "channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
+
+#define gst_rg_limiter_parent_class parent_class
+G_DEFINE_TYPE (GstRgLimiter, gst_rg_limiter, GST_TYPE_BASE_TRANSFORM);
+
+static void gst_rg_limiter_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rg_limiter_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+
+static void
+gst_rg_limiter_class_init (GstRgLimiterClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *element_class;
+ GstBaseTransformClass *trans_class;
+
+ gobject_class = (GObjectClass *) klass;
+ element_class = (GstElementClass *) klass;
+
+ gobject_class->set_property = gst_rg_limiter_set_property;
+ gobject_class->get_property = gst_rg_limiter_get_property;
+
+ g_object_class_install_property (gobject_class, PROP_ENABLED,
+ g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ trans_class = GST_BASE_TRANSFORM_CLASS (klass);
+ trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip);
+ trans_class->passthrough_on_same_caps = FALSE;
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_factory));
+ gst_element_class_set_static_metadata (element_class, "ReplayGain limiter",
+ "Filter/Effect/Audio",
+ "Apply signal compression to raw audio data",
+ "Ren\xc3\xa9 Stadler <mail@renestadler.de>");
+
+ GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0,
+ "ReplayGain limiter element");
+}
+
+static void
+gst_rg_limiter_init (GstRgLimiter * filter)
+{
+ GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
+
+ gst_base_transform_set_passthrough (base, FALSE);
+ gst_base_transform_set_gap_aware (base, TRUE);
+
+ filter->enabled = TRUE;
+}
+
+static void
+gst_rg_limiter_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+ switch (prop_id) {
+ case PROP_ENABLED:
+ filter->enabled = g_value_get_boolean (value);
+ gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
+ !filter->enabled);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rg_limiter_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+ switch (prop_id) {
+ case PROP_ENABLED:
+ g_value_set_boolean (value, filter->enabled);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+#define LIMIT 1.0
+#define THRES 0.5 /* ca. -6 dB */
+#define COMPL 0.5 /* LIMIT - THRESH */
+
+static GstFlowReturn
+gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+ GstRgLimiter *filter = GST_RG_LIMITER (base);
+ gfloat *input;
+ GstMapInfo map;
+ guint count;
+ guint i;
+
+ if (!filter->enabled)
+ return GST_FLOW_OK;
+
+ if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
+ return GST_FLOW_OK;
+
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ input = (gfloat *) map.data;
+ count = gst_buffer_get_size (buf) / sizeof (gfloat);
+
+ for (i = count; i--;) {
+ if (*input > THRES)
+ *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES;
+ else if (*input < -THRES)
+ *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES;
+ input++;
+ }
+
+ gst_buffer_unmap (buf, &map);
+
+ return GST_FLOW_OK;
+}