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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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431 lines
12 KiB
C
431 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.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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/* 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 "gstaudioresample.h"
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#include <gst/audio/audio.h>
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GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
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#define GST_CAT_DEFAULT audioresample_debug
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/* Audioresample 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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};
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#define SUPPORTED_CAPS \
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GST_STATIC_CAPS (\
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (boolean) true"
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#if 0
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/* disabled because it segfaults */
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"audio/x-raw-float, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, " "width = (int) 32"
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#endif
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)
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static GstStaticPadTemplate gst_audioresample_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static GstStaticPadTemplate gst_audioresample_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static void gst_audioresample_base_init (gpointer g_class);
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static void gst_audioresample_class_init (AudioresampleClass * klass);
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static void gst_audioresample_init (Audioresample * audioresample);
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static void gst_audioresample_dispose (GObject * object);
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static void gst_audioresample_chain (GstPad * pad, GstData * _data);
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static void gst_audioresample_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audioresample_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
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GType audioresample_get_type (void)
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{
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static GType audioresample_type = 0;
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if (!audioresample_type)
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{
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static const GTypeInfo audioresample_info = {
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sizeof (AudioresampleClass),
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gst_audioresample_base_init,
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NULL,
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(GClassInitFunc) gst_audioresample_class_init,
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NULL,
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NULL,
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sizeof (Audioresample), 0,
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(GInstanceInitFunc) gst_audioresample_init,};
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audioresample_type =
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g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
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&audioresample_info, 0);
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}
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return audioresample_type;
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}
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static void gst_audioresample_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_audioresample_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_audioresample_sink_template));
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gst_element_class_set_details_simple (gstelement_class, "Audio scaler",
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"Filter/Converter/Audio",
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"Resample audio", "David Schleef <ds@schleef.org>");
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}
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static void gst_audioresample_class_init (AudioresampleClass * 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_audioresample_set_property;
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gobject_class->get_property = gst_audioresample_get_property;
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gobject_class->dispose = gst_audioresample_dispose;
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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,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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parent_class = g_type_class_peek_parent (klass);
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GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
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"audioresample element");
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}
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static void gst_audioresample_expand_caps (GstCaps * caps)
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{
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gint i;
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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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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;
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}
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
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}
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}
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static GstCaps *gst_audioresample_getcaps (GstPad * pad)
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{
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Audioresample *audioresample;
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GstCaps *caps;
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GstPad *otherpad;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
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audioresample->srcpad;
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caps = gst_pad_get_allowed_caps (otherpad);
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gst_audioresample_expand_caps (caps);
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return caps;
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}
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static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
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{
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Audioresample *audioresample;
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GstPad *otherpad;
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int rate;
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GstCaps *copy;
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GstStructure *structure;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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if (pad == audioresample->srcpad) {
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otherpad = audioresample->sinkpad;
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rate = audioresample->i_rate;
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} else
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{
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otherpad = audioresample->srcpad;
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rate = audioresample->o_rate;
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}
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if (!GST_PAD_IS_NEGOTIATING (otherpad))
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return NULL;
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if (gst_caps_get_size (caps) > 1)
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return NULL;
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copy = gst_caps_copy (caps);
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structure = gst_caps_get_structure (copy, 0);
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if (rate) {
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if (gst_structure_fixate_field_nearest_int (structure, "rate", rate)) {
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return copy;
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}
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}
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gst_caps_free (copy);
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return NULL;
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}
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static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
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const GstCaps * caps)
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{
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Audioresample *audioresample;
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GstStructure *structure;
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int rate;
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int channels;
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gboolean ret;
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GstPad *otherpad;
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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
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audioresample->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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if (!ret)
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{
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return GST_PAD_LINK_REFUSED;
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}
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if (gst_pad_is_negotiated (otherpad))
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{
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GstCaps *othercaps = gst_caps_copy (caps);
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int otherrate;
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GstPadLinkReturn linkret;
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if (pad == audioresample->srcpad) {
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otherrate = audioresample->i_rate;
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} else {
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otherrate = audioresample->o_rate;
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}
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gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
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linkret = gst_pad_try_set_caps (otherpad, othercaps);
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if (GST_PAD_LINK_FAILED (linkret)) {
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return GST_PAD_LINK_REFUSED;
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}
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}
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audioresample->channels = channels;
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resample_set_n_channels (audioresample->resample, audioresample->channels);
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if (pad == audioresample->srcpad) {
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audioresample->o_rate = rate;
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resample_set_output_rate (audioresample->resample, audioresample->o_rate);
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GST_DEBUG ("set o_rate to %d", rate);
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} else {
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audioresample->i_rate = rate;
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resample_set_input_rate (audioresample->resample, audioresample->i_rate);
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GST_DEBUG ("set i_rate to %d", rate);
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}
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return GST_PAD_LINK_OK;
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}
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static void gst_audioresample_init (Audioresample * audioresample)
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{
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ResampleState *r;
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audioresample->sinkpad =
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gst_pad_new_from_static_template (&gst_audioresample_sink_template,
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"sink");
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gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
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gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
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gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
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gst_pad_set_getcaps_function (audioresample->sinkpad,
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gst_audioresample_getcaps);
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gst_pad_set_fixate_function (audioresample->sinkpad,
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gst_audioresample_fixate);
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audioresample->srcpad =
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gst_pad_new_from_static_template (&gst_audioresample_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
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gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
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gst_pad_set_getcaps_function (audioresample->srcpad,
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gst_audioresample_getcaps);
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gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
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r = resample_new ();
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audioresample->resample = r;
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resample_set_filter_length (r, 64);
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resample_set_format (r, RESAMPLE_FORMAT_S16);
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}
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static void gst_audioresample_dispose (GObject * object)
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{
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Audioresample *audioresample = GST_AUDIORESAMPLE (object);
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if (audioresample->resample) {
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resample_free (audioresample->resample);
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void gst_audioresample_chain (GstPad * pad, GstData * _data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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Audioresample *audioresample;
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ResampleState *r;
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guchar *data;
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gulong size;
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int outsize;
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GstBuffer *outbuf;
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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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audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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if (!GST_IS_BUFFER (_data)) {
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gst_pad_push (audioresample->srcpad, _data);
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return;
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}
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if (audioresample->passthru) {
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gst_pad_push (audioresample->srcpad, GST_DATA (buf));
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return;
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}
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r = audioresample->resample;
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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GST_DEBUG ("got buffer of %ld bytes", size);
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resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
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buf);
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outsize = resample_get_output_size (r);
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/* FIXME this is audioresample being dumb. dunno why */
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if (outsize == 0) {
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GST_ERROR ("overriding outbuf size");
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outsize = size;
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}
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outbuf = gst_buffer_new_and_alloc (outsize);
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outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
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GST_BUFFER_SIZE (outbuf) = outsize;
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GST_BUFFER_TIMESTAMP (outbuf) =
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audioresample->offset * GST_SECOND / audioresample->o_rate;
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audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
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gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
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}
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static void
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gst_audioresample_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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Audioresample *audioresample;
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g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
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audioresample = GST_AUDIORESAMPLE (object);
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switch (prop_id) {
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case ARG_FILTERLEN:
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audioresample->filter_length = g_value_get_int (value);
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GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n",
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audioresample->filter_length);
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resample_set_filter_length (audioresample->resample,
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audioresample->filter_length);
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break;
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default: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_audioresample_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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Audioresample *audioresample;
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g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
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audioresample = GST_AUDIORESAMPLE (object);
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switch (prop_id) {
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case ARG_FILTERLEN:
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g_value_set_int (value, audioresample->filter_length);
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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 plugin_init (GstPlugin * plugin)
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{
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resample_init ();
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if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
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GST_TYPE_AUDIORESAMPLE)) {
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return FALSE;
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}
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"audioresample",
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"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
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GST_PACKAGE_ORIGIN)
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