| /* |
| * 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-audioinvert |
| * |
| * Swaps upper and lower half of audio samples. Mixing an inverted sample on top of |
| * the original with a slight delay can produce effects that sound like resonance. |
| * Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds. |
| * |
| * <refsect2> |
| * <title>Example launch line</title> |
| * |[ |
| * gst-launch-1.0 audiotestsrc wave=saw ! audioinvert degree=0.4 ! alsasink |
| * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert degree=0.4 ! alsasink |
| * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioinvert degree=0.4 ! 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 "audioinvert.h" |
| |
| #define GST_CAT_DEFAULT gst_audio_invert_debug |
| GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); |
| |
| /* Filter signals and args */ |
| enum |
| { |
| /* FILL ME */ |
| LAST_SIGNAL |
| }; |
| |
| enum |
| { |
| PROP_0, |
| PROP_DEGREE |
| }; |
| |
| #define ALLOWED_CAPS \ |
| "audio/x-raw," \ |
| " format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \ |
| " rate=(int)[1,MAX]," \ |
| " channels=(int)[1,MAX]," \ |
| " layout=(string) {interleaved, non-interleaved}" |
| |
| G_DEFINE_TYPE (GstAudioInvert, gst_audio_invert, GST_TYPE_AUDIO_FILTER); |
| |
| static void gst_audio_invert_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec); |
| static void gst_audio_invert_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec); |
| |
| static gboolean gst_audio_invert_setup (GstAudioFilter * filter, |
| const GstAudioInfo * info); |
| static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base, |
| GstBuffer * buf); |
| |
| static void gst_audio_invert_transform_int (GstAudioInvert * filter, |
| gint16 * data, guint num_samples); |
| static void gst_audio_invert_transform_float (GstAudioInvert * filter, |
| gfloat * data, guint num_samples); |
| |
| /* GObject vmethod implementations */ |
| |
| static void |
| gst_audio_invert_class_init (GstAudioInvertClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstCaps *caps; |
| |
| GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0, |
| "audioinvert element"); |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| |
| gobject_class->set_property = gst_audio_invert_set_property; |
| gobject_class->get_property = gst_audio_invert_get_property; |
| |
| g_object_class_install_property (gobject_class, PROP_DEGREE, |
| g_param_spec_float ("degree", "Degree", |
| "Degree of inversion", 0.0, 1.0, |
| 0.0, |
| G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); |
| |
| gst_element_class_set_static_metadata (gstelement_class, "Audio inversion", |
| "Filter/Effect/Audio", |
| "Swaps upper and lower half of audio samples", |
| "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_invert_transform_ip); |
| GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE; |
| |
| GST_AUDIO_FILTER_CLASS (klass)->setup = |
| GST_DEBUG_FUNCPTR (gst_audio_invert_setup); |
| } |
| |
| static void |
| gst_audio_invert_init (GstAudioInvert * filter) |
| { |
| filter->degree = 0.0; |
| gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); |
| gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE); |
| } |
| |
| static void |
| gst_audio_invert_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| GstAudioInvert *filter = GST_AUDIO_INVERT (object); |
| |
| switch (prop_id) { |
| case PROP_DEGREE: |
| filter->degree = g_value_get_float (value); |
| gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), |
| filter->degree == 0.0); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_audio_invert_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec) |
| { |
| GstAudioInvert *filter = GST_AUDIO_INVERT (object); |
| |
| switch (prop_id) { |
| case PROP_DEGREE: |
| g_value_set_float (value, filter->degree); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| /* GstAudioFilter vmethod implementations */ |
| |
| static gboolean |
| gst_audio_invert_setup (GstAudioFilter * base, const GstAudioInfo * info) |
| { |
| GstAudioInvert *filter = GST_AUDIO_INVERT (base); |
| gboolean ret = TRUE; |
| |
| switch (GST_AUDIO_INFO_FORMAT (info)) { |
| case GST_AUDIO_FORMAT_S16: |
| filter->process = (GstAudioInvertProcessFunc) |
| gst_audio_invert_transform_int; |
| break; |
| case GST_AUDIO_FORMAT_F32: |
| filter->process = (GstAudioInvertProcessFunc) |
| gst_audio_invert_transform_float; |
| break; |
| default: |
| ret = FALSE; |
| break; |
| } |
| return ret; |
| } |
| |
| static void |
| gst_audio_invert_transform_int (GstAudioInvert * filter, |
| gint16 * data, guint num_samples) |
| { |
| gint i; |
| gfloat dry = 1.0 - filter->degree; |
| glong val; |
| |
| for (i = 0; i < num_samples; i++) { |
| val = (*data) * dry + (-1 - (*data)) * filter->degree; |
| *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16); |
| } |
| } |
| |
| static void |
| gst_audio_invert_transform_float (GstAudioInvert * filter, |
| gfloat * data, guint num_samples) |
| { |
| gint i; |
| gfloat dry = 1.0 - filter->degree; |
| glong val; |
| |
| for (i = 0; i < num_samples; i++) { |
| val = (*data) * dry - (*data) * filter->degree; |
| *data++ = val; |
| } |
| } |
| |
| /* GstBaseTransform vmethod implementations */ |
| static GstFlowReturn |
| gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf) |
| { |
| GstAudioInvert *filter = GST_AUDIO_INVERT (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; |
| } |