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43ebe8235f
The title tag belongs into the refsect2.
503 lines
22 KiB
C
503 lines
22 KiB
C
/*
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* GStreamer
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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* Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioamplify
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*
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* Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
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* The difference between the clipping modes is best evaluated by testing.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
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* gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink
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* ]|
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <gst/controller/gstcontroller.h>
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#include "audioamplify.h"
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#define GST_CAT_DEFAULT gst_audio_amplify_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_AMPLIFICATION,
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PROP_CLIPPING_METHOD
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};
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enum
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{
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METHOD_CLIP = 0,
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METHOD_WRAP_NEGATIVE,
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METHOD_WRAP_POSITIVE,
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METHOD_NOCLIP,
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NUM_METHODS
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};
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#define GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD (gst_audio_amplify_clipping_method_get_type ())
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static GType
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gst_audio_amplify_clipping_method_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{METHOD_CLIP, "Normal clipping (default)", "clip"},
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{METHOD_WRAP_NEGATIVE,
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"Push overdriven values back from the opposite side",
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"wrap-negative"},
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{METHOD_WRAP_POSITIVE, "Push overdriven values back from the same side",
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"wrap-positive"},
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{METHOD_NOCLIP, "No clipping", "none"},
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{0, NULL, NULL}
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};
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/* FIXME 0.11: rename to GstAudioAmplifyClippingMethod */
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gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values);
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}
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return gtype;
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}
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#define ALLOWED_CAPS \
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"audio/x-raw-int," \
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" depth=(int)8," \
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" width=(int)8," \
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" endianness=(int)BYTE_ORDER," \
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" signed=(bool)TRUE," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]; " \
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"audio/x-raw-int," \
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" depth=(int)16," \
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" width=(int)16," \
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" endianness=(int)BYTE_ORDER," \
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" signed=(bool)TRUE," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]; " \
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"audio/x-raw-int," \
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" depth=(int)32," \
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" width=(int)32," \
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" endianness=(int)BYTE_ORDER," \
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" signed=(bool)TRUE," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]; " \
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"audio/x-raw-float," \
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" width=(int){32,64}," \
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" endianness=(int)BYTE_ORDER," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]"
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element");
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GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static gboolean gst_audio_amplify_set_process_function (GstAudioAmplify *
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filter, gint clipping, gint format, gint width);
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static void gst_audio_amplify_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_amplify_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_amplify_setup (GstAudioFilter * filter,
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GstRingBufferSpec * format);
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static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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#define MIN_gint8 G_MININT8
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#define MAX_gint8 G_MAXINT8
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#define MIN_gint16 G_MININT16
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#define MAX_gint16 G_MAXINT16
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#define MIN_gint32 G_MININT32
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#define MAX_gint32 G_MAXINT32
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#define MAKE_INT_FUNCS(type,largetype) \
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static void \
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gst_audio_amplify_transform_##type##_clip (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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largetype val = *d * filter->amplification; \
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*d++ = CLAMP (val, MIN_##type, MAX_##type); \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_wrap_negative (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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largetype val = *d * filter->amplification; \
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if (val > MAX_##type) \
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val = MIN_##type + (val - MIN_##type) % ((largetype) MAX_##type + 1 - \
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MIN_##type); \
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else if (val < MIN_##type) \
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val = MAX_##type - (MAX_##type - val) % ((largetype) MAX_##type + 1 - \
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MIN_##type); \
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*d++ = val; \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_wrap_positive (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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largetype val = *d * filter->amplification; \
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do { \
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if (val > MAX_##type) \
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val = MAX_##type - (val - MAX_##type); \
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else if (val < MIN_##type) \
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val = MIN_##type + (MIN_##type - val); \
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else \
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break; \
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} while (1); \
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*d++ = val; \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_noclip (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) \
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*d++ *= filter->amplification; \
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}
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#define MAKE_FLOAT_FUNCS(type) \
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static void \
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gst_audio_amplify_transform_##type##_clip (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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type val = *d* filter->amplification; \
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*d++ = CLAMP (val, -1.0, +1.0); \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_wrap_negative (GstAudioAmplify * \
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filter, void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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type val = *d * filter->amplification; \
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do { \
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if (val > 1.0) \
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val = -1.0 + (val - 1.0); \
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else if (val < -1.0) \
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val = 1.0 - (1.0 - val); \
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else \
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break; \
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} while (1); \
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*d++ = val; \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_wrap_positive (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) { \
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type val = *d* filter->amplification; \
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do { \
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if (val > 1.0) \
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val = 1.0 - (val - 1.0); \
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else if (val < -1.0) \
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val = -1.0 + (-1.0 - val); \
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else \
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break; \
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} while (1); \
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*d++ = val; \
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} \
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} \
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static void \
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gst_audio_amplify_transform_##type##_noclip (GstAudioAmplify * filter, \
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void * data, guint num_samples) \
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{ \
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type *d = data; \
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\
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while (num_samples--) \
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*d++ *= filter->amplification; \
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}
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/* *INDENT-OFF* */
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MAKE_INT_FUNCS (gint8,gint)
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MAKE_INT_FUNCS (gint16,gint)
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MAKE_INT_FUNCS (gint32,gint64)
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MAKE_FLOAT_FUNCS (gfloat)
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MAKE_FLOAT_FUNCS (gdouble)
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/* *INDENT-ON* */
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/* GObject vmethod implementations */
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static void
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gst_audio_amplify_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstCaps *caps;
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gst_element_class_set_details_simple (element_class, "Audio amplifier",
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"Filter/Effect/Audio",
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"Amplifies an audio stream by a given factor",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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}
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static void
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gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_audio_amplify_set_property;
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gobject_class->get_property = gst_audio_amplify_get_property;
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g_object_class_install_property (gobject_class, PROP_AMPLIFICATION,
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g_param_spec_float ("amplification", "Amplification",
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"Factor of amplification", -G_MAXFLOAT, G_MAXFLOAT,
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1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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/**
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* GstAudioAmplify:clipping-method
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*
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* Clipping method: clip mode set values higher than the maximum to the
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* maximum. The wrap-negative mode pushes those values back from the
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* opposite side, wrap-positive pushes them back from the same side.
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*
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**/
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g_object_class_install_property (gobject_class, PROP_CLIPPING_METHOD,
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g_param_spec_enum ("clipping-method", "Clipping method",
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"Selects how to handle values higher than the maximum",
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GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP,
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G_PARAM_READWRITE));
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GST_AUDIO_FILTER_CLASS (klass)->setup =
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GST_DEBUG_FUNCPTR (gst_audio_amplify_setup);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip);
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}
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static void
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gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass)
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{
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filter->amplification = 1.0;
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gst_audio_amplify_set_process_function (filter, METHOD_CLIP,
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GST_BUFTYPE_LINEAR, 16);
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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}
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static GstAudioAmplifyProcessFunc
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gst_audio_amplify_process_function (gint clipping, gint format, gint width)
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{
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static const struct process
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{
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gint format;
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gint width;
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gint clipping;
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GstAudioAmplifyProcessFunc func;
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} process[] = {
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{
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GST_BUFTYPE_FLOAT, 32, METHOD_CLIP,
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gst_audio_amplify_transform_gfloat_clip}, {
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GST_BUFTYPE_FLOAT, 32, METHOD_WRAP_NEGATIVE,
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gst_audio_amplify_transform_gfloat_wrap_negative}, {
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GST_BUFTYPE_FLOAT, 32, METHOD_WRAP_POSITIVE,
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gst_audio_amplify_transform_gfloat_wrap_positive}, {
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GST_BUFTYPE_FLOAT, 32, METHOD_NOCLIP,
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gst_audio_amplify_transform_gfloat_noclip}, {
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GST_BUFTYPE_FLOAT, 64, METHOD_CLIP,
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gst_audio_amplify_transform_gdouble_clip}, {
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GST_BUFTYPE_FLOAT, 64, METHOD_WRAP_NEGATIVE,
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gst_audio_amplify_transform_gdouble_wrap_negative}, {
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GST_BUFTYPE_FLOAT, 64, METHOD_WRAP_POSITIVE,
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gst_audio_amplify_transform_gdouble_wrap_positive}, {
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GST_BUFTYPE_FLOAT, 64, METHOD_NOCLIP,
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gst_audio_amplify_transform_gdouble_noclip}, {
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GST_BUFTYPE_LINEAR, 8, METHOD_CLIP, gst_audio_amplify_transform_gint8_clip}, {
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GST_BUFTYPE_LINEAR, 8, METHOD_WRAP_NEGATIVE,
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gst_audio_amplify_transform_gint8_wrap_negative}, {
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GST_BUFTYPE_LINEAR, 8, METHOD_WRAP_POSITIVE,
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gst_audio_amplify_transform_gint8_wrap_positive}, {
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GST_BUFTYPE_LINEAR, 8, METHOD_NOCLIP,
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gst_audio_amplify_transform_gint8_noclip}, {
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GST_BUFTYPE_LINEAR, 16, METHOD_CLIP,
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gst_audio_amplify_transform_gint16_clip}, {
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GST_BUFTYPE_LINEAR, 16, METHOD_WRAP_NEGATIVE,
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gst_audio_amplify_transform_gint16_wrap_negative}, {
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GST_BUFTYPE_LINEAR, 16, METHOD_WRAP_POSITIVE,
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gst_audio_amplify_transform_gint16_wrap_positive}, {
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GST_BUFTYPE_LINEAR, 16, METHOD_NOCLIP,
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gst_audio_amplify_transform_gint16_noclip}, {
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GST_BUFTYPE_LINEAR, 32, METHOD_CLIP,
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gst_audio_amplify_transform_gint32_clip}, {
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GST_BUFTYPE_LINEAR, 32, METHOD_WRAP_NEGATIVE,
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gst_audio_amplify_transform_gint32_wrap_negative}, {
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GST_BUFTYPE_LINEAR, 32, METHOD_WRAP_POSITIVE,
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gst_audio_amplify_transform_gint32_wrap_positive}, {
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GST_BUFTYPE_LINEAR, 32, METHOD_NOCLIP,
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gst_audio_amplify_transform_gint32_noclip}, {
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0, 0, 0, NULL}
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};
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const struct process *p;
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for (p = process; p->func; p++)
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if (p->format == format && p->width == width && p->clipping == clipping)
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return p->func;
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return NULL;
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}
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static gboolean
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gst_audio_amplify_set_process_function (GstAudioAmplify * filter, gint
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clipping_method, gint format, gint width)
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{
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GstAudioAmplifyProcessFunc process;
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/* set processing function */
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process = gst_audio_amplify_process_function (clipping_method, format, width);
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if (!process) {
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GST_DEBUG ("wrong format");
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return FALSE;
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}
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filter->process = process;
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filter->clipping_method = clipping_method;
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filter->format = format;
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filter->width = width;
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return TRUE;
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}
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static void
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gst_audio_amplify_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
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switch (prop_id) {
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case PROP_AMPLIFICATION:
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filter->amplification = g_value_get_float (value);
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gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
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filter->amplification == 1.0);
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break;
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case PROP_CLIPPING_METHOD:
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gst_audio_amplify_set_process_function (filter, g_value_get_enum (value),
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filter->format, filter->width);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_amplify_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
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switch (prop_id) {
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case PROP_AMPLIFICATION:
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g_value_set_float (value, filter->amplification);
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break;
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case PROP_CLIPPING_METHOD:
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g_value_set_enum (value, filter->clipping_method);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
|
|
|
|
/* 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;
|
|
GstClockTime timestamp, stream_time;
|
|
|
|
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 (G_OBJECT (filter), stream_time);
|
|
|
|
num_samples =
|
|
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
|
|
|
|
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;
|
|
}
|