mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 01:00:37 +00:00
4f3158a64c
Original commit message from CVS: * gst/audioscale/gstaudioscale.c: (gst_audioscale_expand_value), (gst_audioscale_getcaps): Fix getcaps to expand and union lists. (bug #138225) * gst/debug/Makefile.am: * gst/debug/breakmydata.c: (gst_break_my_data_plugin_init): * gst/debug/gstdebug.c: (plugin_init): Merge elements into one plugin. * gst/debug/negotiation.c: (gst_gst_negotiation_get_type), (gst_negotiation_base_init), (gst_negotiation_class_init), (gst_negotiation_init), (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain), (gst_negotiation_set_property), (gst_negotiation_get_property), (gst_negotiation_plugin_init): New element to talk about random negotiation things happening in a pipeline.
488 lines
13 KiB
C
488 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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/* Element-Checklist-Version: 5 */
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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 <math.h>
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/*#define DEBUG_ENABLED */
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#include <gstaudioscale.h>
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#include <gst/audio/audio.h>
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#include <gst/resample/resample.h>
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/* elementfactory information */
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static GstElementDetails gst_audioscale_details =
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GST_ELEMENT_DETAILS ("Audio scaler",
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"Filter/Converter/Audio",
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"Resample audio",
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"David Schleef <ds@schleef.org>");
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/* Audioscale 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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ARG_0,
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ARG_FILTERLEN,
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ARG_METHOD,
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/* FILL ME */
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};
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static GstStaticPadTemplate gst_audioscale_sink_template =
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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 (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate gst_audioscale_src_template =
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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 (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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#define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type())
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static GType
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gst_audioscale_method_get_type (void)
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{
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static GType audioscale_method_type = 0;
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static GEnumValue audioscale_methods[] = {
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{GST_RESAMPLE_NEAREST, "0", "Nearest"},
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{GST_RESAMPLE_BILINEAR, "1", "Bilinear"},
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{GST_RESAMPLE_SINC, "2", "Sinc"},
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{0, NULL, NULL},
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};
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if (!audioscale_method_type) {
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audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod",
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audioscale_methods);
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}
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return audioscale_method_type;
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}
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static void gst_audioscale_base_init (gpointer g_class);
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static void gst_audioscale_class_init (AudioscaleClass * klass);
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static void gst_audioscale_init (Audioscale * audioscale);
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static void gst_audioscale_chain (GstPad * pad, GstData * _data);
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static void gst_audioscale_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audioscale_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
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GType
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audioscale_get_type (void)
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{
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static GType audioscale_type = 0;
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if (!audioscale_type) {
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static const GTypeInfo audioscale_info = {
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sizeof (AudioscaleClass),
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gst_audioscale_base_init,
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NULL,
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(GClassInitFunc) gst_audioscale_class_init,
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NULL,
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NULL,
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sizeof (Audioscale),
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0,
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(GInstanceInitFunc) gst_audioscale_init,
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};
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audioscale_type =
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g_type_register_static (GST_TYPE_ELEMENT, "Audioscale",
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&audioscale_info, 0);
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}
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return audioscale_type;
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}
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static void
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gst_audioscale_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioscale_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioscale_sink_template));
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gst_element_class_set_details (gstelement_class, &gst_audioscale_details);
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}
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static void
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gst_audioscale_class_init (AudioscaleClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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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_audioscale_set_property;
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gobject_class->get_property = gst_audioscale_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
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g_param_spec_int ("filter_length", "filter_length", "filter_length",
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0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_METHOD,
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g_param_spec_enum ("method", "method", "method",
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GST_TYPE_AUDIOSCALE_METHOD, GST_RESAMPLE_SINC,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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}
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static void
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gst_audioscale_expand_value (GValue * dest, const GValue * src)
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{
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int rate_min, rate_max;
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if (G_VALUE_TYPE (src) == G_TYPE_INT ||
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G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE) {
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if (G_VALUE_TYPE (src) == G_TYPE_INT) {
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rate_min = g_value_get_int (src);
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rate_max = rate_min;
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} else {
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rate_min = gst_value_get_int_range_min (src);
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rate_max = gst_value_get_int_range_max (src);
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}
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rate_min /= 2;
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if (rate_min < 1)
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rate_min = 1;
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if (rate_max < G_MAXINT / 2) {
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rate_max *= 2;
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} else {
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rate_max = G_MAXINT;
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}
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g_value_init (dest, GST_TYPE_INT_RANGE);
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gst_value_set_int_range (dest, rate_min, rate_max);
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return;
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}
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if (G_VALUE_TYPE (src) == GST_TYPE_LIST) {
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int i;
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g_value_init (dest, GST_TYPE_LIST);
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for (i = 0; i < gst_value_list_get_size (src); i++) {
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const GValue *s = gst_value_list_get_value (src, i);
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GValue d = { 0 };
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int j;
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gst_audioscale_expand_value (&d, s);
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for (j = 0; j < gst_value_list_get_size (dest); j++) {
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const GValue *s2 = gst_value_list_get_value (dest, j);
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GValue d2 = { 0 };
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gst_value_union (&d2, &d, s2);
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if (G_VALUE_TYPE (&d2) == GST_TYPE_INT_RANGE) {
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g_value_unset ((GValue *) s2);
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gst_value_init_and_copy ((GValue *) s2, &d2);
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break;
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}
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g_value_unset (&d2);
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}
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if (j == gst_value_list_get_size (dest)) {
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gst_value_list_append_value (dest, &d);
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}
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g_value_unset (&d);
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}
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if (gst_value_list_get_size (dest) == 1) {
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const GValue *s = gst_value_list_get_value (dest, 0);
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GValue d = { 0 };
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gst_value_init_and_copy (&d, s);
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g_value_unset (dest);
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gst_value_init_and_copy (dest, &d);
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g_value_unset (&d);
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}
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return;
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}
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GST_ERROR ("unexpected value type");
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}
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static GstCaps *
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gst_audioscale_getcaps (GstPad * pad)
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{
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Audioscale *audioscale;
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GstCaps *caps;
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GstPad *otherpad;
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int i;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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otherpad = (pad == audioscale->srcpad) ? audioscale->sinkpad :
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audioscale->srcpad;
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caps = gst_pad_get_allowed_caps (otherpad);
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/* we do this hack, because the audioscale lib doesn't handle
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* rate conversions larger than a factor of 2 */
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for (i = 0; i < gst_caps_get_size (caps); i++) {
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GstStructure *structure = gst_caps_get_structure (caps, i);
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const GValue *value;
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GValue dest = { 0 };
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value = gst_structure_get_value (structure, "rate");
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if (value == NULL) {
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GST_ERROR ("caps structure doesn't have required rate field");
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return NULL;
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}
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gst_audioscale_expand_value (&dest, value);
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gst_structure_set_value (structure, "rate", &dest);
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}
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return caps;
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}
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static GstPadLinkReturn
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gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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{
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Audioscale *audioscale;
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gst_resample_t *r;
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GstStructure *structure;
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int rate;
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int channels;
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int ret;
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GstPadLinkReturn link_ret;
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GstPad *otherpad;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = audioscale->gst_resample;
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otherpad = (pad == audioscale->srcpad) ? audioscale->sinkpad
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: audioscale->srcpad;
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "rate", &rate);
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ret &= gst_structure_get_int (structure, "channels", &channels);
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link_ret = gst_pad_try_set_caps (otherpad, gst_caps_copy (caps));
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if (GST_PAD_LINK_SUCCESSFUL (link_ret)) {
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audioscale->passthru = TRUE;
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r->channels = channels;
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r->i_rate = rate;
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r->o_rate = rate;
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return link_ret;
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}
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audioscale->passthru = FALSE;
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if (gst_pad_is_negotiated (otherpad)) {
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GstCaps *trycaps = gst_caps_copy (caps);
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gst_caps_set_simple (trycaps,
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"rate", G_TYPE_INT,
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(int) ((pad == audioscale->srcpad) ? r->i_rate : r->o_rate), NULL);
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link_ret = gst_pad_try_set_caps (otherpad, trycaps);
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if (GST_PAD_LINK_FAILED (link_ret)) {
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return link_ret;
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}
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}
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r->channels = channels;
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if (pad == audioscale->srcpad) {
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r->o_rate = rate;
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} else {
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r->i_rate = rate;
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}
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gst_resample_reinit (r);
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return GST_PAD_LINK_OK;
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}
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static void *
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gst_audioscale_get_buffer (void *priv, unsigned int size)
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{
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Audioscale *audioscale = priv;
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audioscale->outbuf = gst_buffer_new ();
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GST_BUFFER_SIZE (audioscale->outbuf) = size;
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GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
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GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
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audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate;
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audioscale->offset +=
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size / sizeof (gint16) / audioscale->gst_resample->channels;
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return GST_BUFFER_DATA (audioscale->outbuf);
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}
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static void
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gst_audioscale_init (Audioscale * audioscale)
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{
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gst_resample_t *r;
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audioscale->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audioscale_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (audioscale), audioscale->sinkpad);
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gst_pad_set_chain_function (audioscale->sinkpad, gst_audioscale_chain);
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gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_link);
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gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps);
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audioscale->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audioscale_src_template), "src");
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gst_element_add_pad (GST_ELEMENT (audioscale), audioscale->srcpad);
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gst_pad_set_link_function (audioscale->srcpad, gst_audioscale_link);
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gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
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r = g_new0 (gst_resample_t, 1);
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audioscale->gst_resample = r;
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r->priv = audioscale;
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r->get_buffer = gst_audioscale_get_buffer;
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r->method = GST_RESAMPLE_SINC;
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r->channels = 0;
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r->filter_length = 16;
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r->i_rate = -1;
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r->o_rate = -1;
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r->format = GST_RESAMPLE_S16;
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/*r->verbose = 1; */
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gst_resample_init (r);
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/* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */
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}
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static void
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gst_audioscale_chain (GstPad * pad, GstData * _data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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Audioscale *audioscale;
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guchar *data;
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gulong size;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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if (audioscale->passthru) {
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gst_pad_push (audioscale->srcpad, GST_DATA (buf));
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return;
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}
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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GST_DEBUG ("gst_audioscale_chain: got buffer of %ld bytes in '%s'\n",
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size, gst_element_get_name (GST_ELEMENT (audioscale)));
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gst_resample_scale (audioscale->gst_resample, data, size);
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gst_pad_push (audioscale->srcpad, GST_DATA (audioscale->outbuf));
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gst_buffer_unref (buf);
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}
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static void
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gst_audioscale_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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Audioscale *src;
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gst_resample_t *r;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail (GST_IS_AUDIOSCALE (object));
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src = GST_AUDIOSCALE (object);
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r = src->gst_resample;
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switch (prop_id) {
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case ARG_FILTERLEN:
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r->filter_length = g_value_get_int (value);
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GST_DEBUG_OBJECT (GST_ELEMENT (src), "new filter length %d\n",
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r->filter_length);
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break;
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case ARG_METHOD:
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r->method = g_value_get_enum (value);
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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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gst_resample_reinit (r);
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}
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static void
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gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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Audioscale *src;
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gst_resample_t *r;
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src = GST_AUDIOSCALE (object);
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r = src->gst_resample;
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switch (prop_id) {
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case ARG_FILTERLEN:
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g_value_set_int (value, r->filter_length);
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break;
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case ARG_METHOD:
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g_value_set_enum (value, r->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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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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/* load support library */
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if (!gst_library_load ("gstresample"))
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return FALSE;
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if (!gst_element_register (plugin, "audioscale", GST_RANK_NONE,
|
|
GST_TYPE_AUDIOSCALE)) {
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"audioscale",
|
|
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
|