| /* 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_static_pad_template (element_class, &src_factory); |
| gst_element_class_add_static_pad_template (element_class, &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; |
| } |