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0fa3992e37
Must remove a possibly-fixed channel-mask field if we're going to set unfixed channels on the structure, or a different channel count.
661 lines
19 KiB
C
661 lines
19 KiB
C
/*
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* GStreamer
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* Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
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* Copyright (C) 2006 Sebastian Dröge <slomo@circular-chaos.org>
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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-audiopanorama
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*
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* Stereo panorama effect with controllable pan position. One can choose between the default psychoacoustic panning method,
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* which keeps the same perceived loudness, and a simple panning method that just controls the volume on one channel.
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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 ! audiopanorama panorama=-1.00 ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiopanorama panorama=-1.00 ! alsasink
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* gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama panorama=-1.00 ! audioconvert ! alsasink
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* gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama method=simple panorama=-0.50 ! 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 <string.h>
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include "audiopanorama.h"
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#define GST_CAT_DEFAULT gst_audio_panorama_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_PANORAMA,
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PROP_METHOD
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};
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enum
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{
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METHOD_PSYCHOACOUSTIC = 0,
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METHOD_SIMPLE,
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NUM_METHODS
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};
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#define GST_TYPE_AUDIO_PANORAMA_METHOD (gst_audio_panorama_method_get_type ())
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static GType
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gst_audio_panorama_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_PSYCHOACOUSTIC, "Psychoacoustic Panning (default)",
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"psychoacoustic"},
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{METHOD_SIMPLE, "Simple Panning", "simple"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioPanoramaMethod", values);
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}
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return gtype;
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}
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
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"rate = (int) [ 1, MAX ], " "channels = (int) 1, "
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"layout = (string) interleaved;"
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"audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
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"rate = (int) [ 1, MAX ], " "channels = (int) 2, "
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"layout = (string) interleaved, " "channel-mask = (bitmask) 0x3")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (S32) ", " GST_AUDIO_NE (S16) "}, "
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"rate = (int) [ 1, MAX ], " "channels = (int) 2, "
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"layout = (string) interleaved, " "channel-mask = (bitmask)0x3")
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);
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G_DEFINE_TYPE (GstAudioPanorama, gst_audio_panorama, GST_TYPE_BASE_TRANSFORM);
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static void gst_audio_panorama_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_panorama_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_panorama_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, gsize * size);
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static GstCaps *gst_audio_panorama_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter);
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static gboolean gst_audio_panorama_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static void gst_audio_panorama_transform_m2s_int (GstAudioPanorama * filter,
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gint16 * idata, gint16 * odata, guint num_samples);
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static void gst_audio_panorama_transform_s2s_int (GstAudioPanorama * filter,
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gint16 * idata, gint16 * odata, guint num_samples);
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static void gst_audio_panorama_transform_m2s_float (GstAudioPanorama * filter,
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gfloat * idata, gfloat * odata, guint num_samples);
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static void gst_audio_panorama_transform_s2s_float (GstAudioPanorama * filter,
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gfloat * idata, gfloat * odata, guint num_samples);
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static void gst_audio_panorama_transform_m2s_int_simple (GstAudioPanorama *
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filter, gint16 * idata, gint16 * odata, guint num_samples);
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static void gst_audio_panorama_transform_s2s_int_simple (GstAudioPanorama *
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filter, gint16 * idata, gint16 * odata, guint num_samples);
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static void gst_audio_panorama_transform_m2s_float_simple (GstAudioPanorama *
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filter, gfloat * idata, gfloat * odata, guint num_samples);
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static void gst_audio_panorama_transform_s2s_float_simple (GstAudioPanorama *
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filter, gfloat * idata, gfloat * odata, guint num_samples);
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static GstFlowReturn gst_audio_panorama_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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/* Table with processing functions: [channels][format][method] */
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static GstAudioPanoramaProcessFunc panorama_process_functions[2][2][2] = {
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{
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{(GstAudioPanoramaProcessFunc) gst_audio_panorama_transform_m2s_int,
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(GstAudioPanoramaProcessFunc)
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gst_audio_panorama_transform_m2s_int_simple},
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{(GstAudioPanoramaProcessFunc) gst_audio_panorama_transform_m2s_float,
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(GstAudioPanoramaProcessFunc)
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gst_audio_panorama_transform_m2s_float_simple}
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},
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{
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{(GstAudioPanoramaProcessFunc) gst_audio_panorama_transform_s2s_int,
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(GstAudioPanoramaProcessFunc)
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gst_audio_panorama_transform_s2s_int_simple},
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{(GstAudioPanoramaProcessFunc) gst_audio_panorama_transform_s2s_float,
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(GstAudioPanoramaProcessFunc)
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gst_audio_panorama_transform_s2s_float_simple}
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}
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};
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/* GObject vmethod implementations */
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static void
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gst_audio_panorama_class_init (GstAudioPanoramaClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GST_DEBUG_CATEGORY_INIT (gst_audio_panorama_debug, "audiopanorama", 0,
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"audiopanorama element");
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_audio_panorama_set_property;
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gobject_class->get_property = gst_audio_panorama_get_property;
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g_object_class_install_property (gobject_class, PROP_PANORAMA,
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g_param_spec_float ("panorama", "Panorama",
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"Position in stereo panorama (-1.0 left -> 1.0 right)", -1.0, 1.0,
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0.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAudioPanorama:method
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*
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* Panning method: psychoacoustic mode keeps the same perceived loudness,
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* while simple mode just controls the volume of one channel. It's merely
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* a matter of taste which method should be chosen.
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*
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* Since: 0.10.6
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**/
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g_object_class_install_property (gobject_class, PROP_METHOD,
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g_param_spec_enum ("method", "Panning method",
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"Psychoacoustic mode keeps same perceived loudness, "
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"simple mode just controls volume of one channel.",
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GST_TYPE_AUDIO_PANORAMA_METHOD, METHOD_PSYCHOACOUSTIC,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class, "Stereo positioning",
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"Filter/Effect/Audio",
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"Positions audio streams in the stereo panorama",
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"Stefan Kost <ensonic@users.sf.net>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_panorama_get_unit_size);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
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GST_DEBUG_FUNCPTR (gst_audio_panorama_transform_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
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GST_DEBUG_FUNCPTR (gst_audio_panorama_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (gst_audio_panorama_transform);
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}
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static void
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gst_audio_panorama_init (GstAudioPanorama * filter)
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{
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filter->panorama = 0;
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filter->method = METHOD_PSYCHOACOUSTIC;
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gst_audio_info_init (&filter->info);
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filter->process = NULL;
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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}
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static gboolean
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gst_audio_panorama_set_process_function (GstAudioPanorama * filter,
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GstAudioInfo * info)
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{
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gint channel_index, format_index, method_index;
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const GstAudioFormatInfo *finfo = info->finfo;
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/* set processing function */
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channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1;
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if (channel_index > 1 || channel_index < 0) {
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filter->process = NULL;
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return FALSE;
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}
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format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0;
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method_index = filter->method;
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if (method_index >= NUM_METHODS || method_index < 0)
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method_index = METHOD_PSYCHOACOUSTIC;
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filter->process =
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panorama_process_functions[channel_index][format_index][method_index];
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return TRUE;
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}
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static void
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gst_audio_panorama_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioPanorama *filter = GST_AUDIO_PANORAMA (object);
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switch (prop_id) {
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case PROP_PANORAMA:
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filter->panorama = g_value_get_float (value);
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break;
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case PROP_METHOD:
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filter->method = g_value_get_enum (value);
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gst_audio_panorama_set_process_function (filter, &filter->info);
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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_panorama_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioPanorama *filter = GST_AUDIO_PANORAMA (object);
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switch (prop_id) {
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case PROP_PANORAMA:
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g_value_set_float (value, filter->panorama);
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break;
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case PROP_METHOD:
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g_value_set_enum (value, filter->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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}
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/* GstBaseTransform vmethod implementations */
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static gboolean
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gst_audio_panorama_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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gsize * size)
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{
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GstAudioInfo info;
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g_assert (size);
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if (!gst_audio_info_from_caps (&info, caps))
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return FALSE;
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*size = GST_AUDIO_INFO_BPF (&info);
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return TRUE;
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}
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static GstCaps *
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gst_audio_panorama_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter)
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{
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GstCaps *res;
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GstStructure *structure;
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/* transform caps gives one single caps so we can just replace
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* the channel property with our range. */
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res = gst_caps_copy (caps);
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structure = gst_caps_get_structure (res, 0);
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if (direction == GST_PAD_SRC) {
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GST_INFO ("allow 1-2 channels");
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gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
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gst_structure_remove_field (structure, "channel-mask");
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} else {
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GST_INFO ("allow 2 channels");
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gst_structure_set (structure, "channels", G_TYPE_INT, 2, NULL);
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gst_structure_remove_field (structure, "channel-mask");
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}
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return res;
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}
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static gboolean
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gst_audio_panorama_set_caps (GstBaseTransform * base, GstCaps * incaps,
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GstCaps * outcaps)
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{
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GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
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GstAudioInfo info;
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/*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
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if (!gst_audio_info_from_caps (&info, incaps))
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goto no_format;
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GST_DEBUG ("try to process %d input with %d channels",
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GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info));
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if (!gst_audio_panorama_set_process_function (filter, &info))
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goto no_format;
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filter->info = info;
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return TRUE;
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no_format:
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{
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GST_DEBUG ("invalid caps");
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return FALSE;
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}
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}
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/* psychoacoustic processing functions */
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static void
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gst_audio_panorama_transform_m2s_int (GstAudioPanorama * filter, gint16 * idata,
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gint16 * odata, guint num_samples)
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{
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guint i;
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gdouble val;
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glong lval, rval;
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gdouble rpan, lpan;
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/* pan: -1.0 0.0 1.0
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* lpan: 1.0 0.5 0.0
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* rpan: 0.0 0.5 1.0
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*
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* FIXME: we should use -3db (1/sqtr(2)) for 50:50
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*/
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rpan = (gdouble) (filter->panorama + 1.0) / 2.0;
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lpan = 1.0 - rpan;
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for (i = 0; i < num_samples; i++) {
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val = (gdouble) * idata++;
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lval = (glong) (val * lpan);
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rval = (glong) (val * rpan);
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*odata++ = (gint16) CLAMP (lval, G_MININT16, G_MAXINT16);
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*odata++ = (gint16) CLAMP (rval, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_panorama_transform_s2s_int (GstAudioPanorama * filter, gint16 * idata,
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gint16 * odata, guint num_samples)
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{
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guint i;
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glong lval, rval;
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gdouble lival, rival;
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gdouble lrpan, llpan, rrpan, rlpan;
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/* pan: -1.0 0.0 1.0
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* llpan: 1.0 1.0 0.0
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* lrpan: 1.0 0.0 0.0
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* rrpan: 0.0 1.0 1.0
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* rlpan: 0.0 0.0 1.0
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*/
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if (filter->panorama > 0) {
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rlpan = (gdouble) filter->panorama;
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llpan = 1.0 - rlpan;
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lrpan = 0.0;
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rrpan = 1.0;
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} else {
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rrpan = (gdouble) (1.0 + filter->panorama);
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lrpan = 1.0 - rrpan;
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rlpan = 0.0;
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llpan = 1.0;
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}
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for (i = 0; i < num_samples; i++) {
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lival = (gdouble) * idata++;
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rival = (gdouble) * idata++;
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lval = lival * llpan + rival * lrpan;
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rval = lival * rlpan + rival * rrpan;
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*odata++ = (gint16) CLAMP (lval, G_MININT16, G_MAXINT16);
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*odata++ = (gint16) CLAMP (rval, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_panorama_transform_m2s_float (GstAudioPanorama * filter,
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gfloat * idata, gfloat * odata, guint num_samples)
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{
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guint i;
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gfloat val;
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gdouble rpan, lpan;
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/* pan: -1.0 0.0 1.0
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* lpan: 1.0 0.5 0.0
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* rpan: 0.0 0.5 1.0
|
|
*
|
|
* FIXME: we should use -3db (1/sqtr(2)) for 50:50
|
|
*/
|
|
rpan = (gdouble) (filter->panorama + 1.0) / 2.0;
|
|
lpan = 1.0 - rpan;
|
|
|
|
for (i = 0; i < num_samples; i++) {
|
|
val = *idata++;
|
|
|
|
*odata++ = val * lpan;
|
|
*odata++ = val * rpan;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_panorama_transform_s2s_float (GstAudioPanorama * filter,
|
|
gfloat * idata, gfloat * odata, guint num_samples)
|
|
{
|
|
guint i;
|
|
gfloat lival, rival;
|
|
gdouble lrpan, llpan, rrpan, rlpan;
|
|
|
|
/* pan: -1.0 0.0 1.0
|
|
* llpan: 1.0 1.0 0.0
|
|
* lrpan: 1.0 0.0 0.0
|
|
* rrpan: 0.0 1.0 1.0
|
|
* rlpan: 0.0 0.0 1.0
|
|
*/
|
|
if (filter->panorama > 0) {
|
|
rlpan = (gdouble) filter->panorama;
|
|
llpan = 1.0 - rlpan;
|
|
lrpan = 0.0;
|
|
rrpan = 1.0;
|
|
} else {
|
|
rrpan = (gdouble) (1.0 + filter->panorama);
|
|
lrpan = 1.0 - rrpan;
|
|
rlpan = 0.0;
|
|
llpan = 1.0;
|
|
}
|
|
|
|
for (i = 0; i < num_samples; i++) {
|
|
lival = *idata++;
|
|
rival = *idata++;
|
|
|
|
*odata++ = lival * llpan + rival * lrpan;
|
|
*odata++ = lival * rlpan + rival * rrpan;
|
|
}
|
|
}
|
|
|
|
/* simple processing functions */
|
|
static void
|
|
gst_audio_panorama_transform_m2s_int_simple (GstAudioPanorama * filter,
|
|
gint16 * idata, gint16 * odata, guint num_samples)
|
|
{
|
|
guint i;
|
|
gdouble pan;
|
|
glong lval, rval;
|
|
|
|
if (filter->panorama > 0.0) {
|
|
pan = 1.0 - filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
rval = *idata++;
|
|
lval = (glong) ((gdouble) rval * pan);
|
|
|
|
*odata++ = (gint16) CLAMP (lval, G_MININT16, G_MAXINT16);
|
|
*odata++ = (gint16) rval;
|
|
}
|
|
} else {
|
|
pan = 1.0 + filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
lval = *idata++;
|
|
rval = (glong) ((gdouble) lval * pan);
|
|
|
|
*odata++ = (gint16) lval;
|
|
*odata++ = (gint16) CLAMP (rval, G_MININT16, G_MAXINT16);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_panorama_transform_s2s_int_simple (GstAudioPanorama * filter,
|
|
gint16 * idata, gint16 * odata, guint num_samples)
|
|
{
|
|
guint i;
|
|
glong lval, rval;
|
|
gdouble lival, rival, pan;
|
|
|
|
if (filter->panorama > 0.0) {
|
|
pan = 1.0 - filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
lival = (gdouble) * idata++;
|
|
rival = (gdouble) * idata++;
|
|
|
|
lval = (glong) (lival * pan);
|
|
rval = (glong) rival;
|
|
|
|
*odata++ = (gint16) CLAMP (lval, G_MININT16, G_MAXINT16);
|
|
*odata++ = (gint16) rval;
|
|
}
|
|
} else {
|
|
pan = 1.0 + filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
lival = (gdouble) * idata++;
|
|
rival = (gdouble) * idata++;
|
|
|
|
lval = (glong) lival;
|
|
rval = (glong) (rival * pan);
|
|
|
|
*odata++ = (gint16) lval;
|
|
*odata++ = (gint16) CLAMP (rval, G_MININT16, G_MAXINT16);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_panorama_transform_m2s_float_simple (GstAudioPanorama * filter,
|
|
gfloat * idata, gfloat * odata, guint num_samples)
|
|
{
|
|
guint i;
|
|
gfloat val, pan;
|
|
|
|
if (filter->panorama > 0.0) {
|
|
pan = 1.0 - filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
val = *idata++;
|
|
|
|
*odata++ = val * pan;
|
|
*odata++ = val;
|
|
}
|
|
} else {
|
|
pan = 1.0 + filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
val = *idata++;
|
|
|
|
*odata++ = val;
|
|
*odata++ = val * pan;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_panorama_transform_s2s_float_simple (GstAudioPanorama * filter,
|
|
gfloat * idata, gfloat * odata, guint num_samples)
|
|
{
|
|
guint i;
|
|
gfloat lival, rival, pan;
|
|
|
|
if (filter->panorama > 0.0) {
|
|
pan = 1.0 - filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
lival = *idata++;
|
|
rival = *idata++;
|
|
|
|
*odata++ = lival * pan;
|
|
*odata++ = rival;
|
|
}
|
|
} else {
|
|
pan = 1.0 + filter->panorama;
|
|
for (i = 0; i < num_samples; i++) {
|
|
lival = *idata++;
|
|
rival = *idata++;
|
|
|
|
*odata++ = lival;
|
|
*odata++ = rival * pan;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* this function does the actual processing
|
|
*/
|
|
static GstFlowReturn
|
|
gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
|
GstBuffer * outbuf)
|
|
{
|
|
GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
|
|
GstClockTime timestamp, stream_time;
|
|
GstMapInfo inmap, outmap;
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
|
|
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);
|
|
|
|
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
|
|
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
|
|
|
|
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
|
|
memset (outmap.data, 0, outmap.size);
|
|
} else {
|
|
/* output always stereo, input mono or stereo,
|
|
* and info describes input format */
|
|
guint num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info));
|
|
|
|
filter->process (filter, inmap.data, outmap.data, num_samples);
|
|
}
|
|
|
|
gst_buffer_unmap (inbuf, &inmap);
|
|
gst_buffer_unmap (outbuf, &outmap);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|