| /* |
| * 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 clipping-method=wrap-negative ! alsasink |
| * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 clipping-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; |
| } |
