/* * 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 method=wrap-negative ! alsasink * gst-launch-1.0 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 "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; }