/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include /*#define DEBUG_ENABLED */ #include #include #include /* elementfactory information */ static GstElementDetails audioscale_details = { "Audio scaler", "Filter/Audio", "LGPL", "Audio resampler", VERSION, "Wim Taymans ", "(C) 2000", }; /* Audioscale signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_FREQUENCY, ARG_FILTERLEN, ARG_METHOD, /* FILL ME */ }; GST_PAD_TEMPLATE_FACTORY (sink_factory, "sink", GST_PAD_SINK, GST_PAD_ALWAYS, gst_caps_new ("audioscale_sink", "audio/x-raw-int", GST_AUDIO_INT_PAD_TEMPLATE_PROPS) ); GST_PAD_TEMPLATE_FACTORY (src_factory, "src", GST_PAD_SRC, GST_PAD_ALWAYS, gst_caps_new ("audioscale_src", "audio/x-raw-int", GST_AUDIO_INT_PAD_TEMPLATE_PROPS) ); #define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type()) static GType gst_audioscale_method_get_type (void) { static GType audioscale_method_type = 0; static GEnumValue audioscale_methods[] = { { RESAMPLE_NEAREST, "0", "Nearest" }, { RESAMPLE_BILINEAR, "1", "Bilinear" }, { RESAMPLE_SINC, "2", "Sinc" }, { 0, NULL, NULL }, }; if(!audioscale_method_type){ audioscale_method_type = g_enum_register_static("GstAudioscaleMethod", audioscale_methods); } return audioscale_method_type; } static void gst_audioscale_class_init (AudioscaleClass *klass); static void gst_audioscale_init (Audioscale *audioscale); static void gst_audioscale_chain (GstPad *pad, GstBuffer *buf); static void gst_audioscale_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstElementClass *parent_class = NULL; /*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */ GType audioscale_get_type (void) { static GType audioscale_type = 0; if (!audioscale_type) { static const GTypeInfo audioscale_info = { sizeof(AudioscaleClass), NULL, NULL, (GClassInitFunc)gst_audioscale_class_init, NULL, NULL, sizeof(Audioscale), 0, (GInstanceInitFunc)gst_audioscale_init, }; audioscale_type = g_type_register_static(GST_TYPE_ELEMENT, "Audioscale", &audioscale_info, 0); } return audioscale_type; } static void gst_audioscale_class_init (AudioscaleClass *klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass*)klass; gstelement_class = (GstElementClass*)klass; g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY, g_param_spec_int ("frequency","frequency","frequency", 0,G_MAXINT,44100,G_PARAM_READWRITE|G_PARAM_CONSTRUCT)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN, g_param_spec_int ("filter_length", "filter_length", "filter_length", 0, G_MAXINT, 16, G_PARAM_READWRITE|G_PARAM_CONSTRUCT)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_METHOD, g_param_spec_enum ("method", "method", "method", GST_TYPE_AUDIOSCALE_METHOD, RESAMPLE_SINC, G_PARAM_READWRITE|G_PARAM_CONSTRUCT)); parent_class = g_type_class_ref(GST_TYPE_ELEMENT); gobject_class->set_property = gst_audioscale_set_property; gobject_class->get_property = gst_audioscale_get_property; } static GstPadLinkReturn gst_audioscale_sinkconnect (GstPad * pad, GstCaps * caps) { Audioscale *audioscale; resample_t *r; GstCaps *newcaps; gint rate; audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); r = audioscale->resample; gst_caps_get_int (caps, "rate", &rate); gst_caps_get_int (caps, "channels", &r->channels); r->i_rate = rate; resample_reinit(r); newcaps = gst_caps_copy (caps); gst_caps_set (newcaps, "rate", GST_PROPS_INT_TYPE, audioscale->targetfrequency, NULL); if (GST_CAPS_IS_FIXED (caps)) return gst_pad_try_set_caps (audioscale->srcpad, newcaps); else return GST_PAD_LINK_DELAYED; } static void * gst_audioscale_get_buffer (void *priv, unsigned int size) { Audioscale * audioscale = priv; audioscale->outbuf = gst_buffer_new(); GST_BUFFER_SIZE(audioscale->outbuf) = size; GST_BUFFER_DATA(audioscale->outbuf) = g_malloc(size); GST_BUFFER_TIMESTAMP(audioscale->outbuf) = audioscale->offset * GST_SECOND / audioscale->targetfrequency; audioscale->offset += size / sizeof(gint16) / audioscale->resample->channels; return GST_BUFFER_DATA(audioscale->outbuf); } static void gst_audioscale_init (Audioscale *audioscale) { resample_t *r; audioscale->sinkpad = gst_pad_new_from_template ( GST_PAD_TEMPLATE_GET (sink_factory), "sink"); gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->sinkpad); gst_pad_set_chain_function(audioscale->sinkpad,gst_audioscale_chain); gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_sinkconnect); audioscale->srcpad = gst_pad_new_from_template ( GST_PAD_TEMPLATE_GET (src_factory), "src"); gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->srcpad); r = g_new0(resample_t,1); audioscale->resample = r; r->priv = audioscale; r->get_buffer = gst_audioscale_get_buffer; r->method = RESAMPLE_SINC; r->channels = 0; r->filter_length = 16; r->i_rate = -1; r->o_rate = -1; r->format = RESAMPLE_S16; /*r->verbose = 1; */ resample_init(r); /* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */ } static void gst_audioscale_chain (GstPad *pad, GstBuffer *buf) { Audioscale *audioscale; guchar *data; gulong size; g_return_if_fail(pad != NULL); g_return_if_fail(GST_IS_PAD(pad)); g_return_if_fail(buf != NULL); audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); data = GST_BUFFER_DATA(buf); size = GST_BUFFER_SIZE(buf); GST_DEBUG ( "gst_audioscale_chain: got buffer of %ld bytes in '%s'\n", size, gst_element_get_name (GST_ELEMENT (audioscale))); resample_scale (audioscale->resample, data, size); gst_pad_push (audioscale->srcpad, audioscale->outbuf); gst_buffer_unref (buf); } static void gst_audioscale_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { Audioscale *src; resample_t *r; /* it's not null if we got it, but it might not be ours */ g_return_if_fail(GST_IS_AUDIOSCALE(object)); src = GST_AUDIOSCALE(object); r = src->resample; switch (prop_id) { case ARG_FREQUENCY: src->targetfrequency = g_value_get_int (value); r->o_rate = src->targetfrequency; break; case ARG_FILTERLEN: r->filter_length = g_value_get_int (value); GST_DEBUG_OBJECT (GST_ELEMENT(src), "new filter length %d\n", r->filter_length); break; case ARG_METHOD: r->method = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } resample_reinit (r); } static void gst_audioscale_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) { Audioscale *src; resample_t *r; src = GST_AUDIOSCALE (object); r = src->resample; switch (prop_id) { case ARG_FREQUENCY: g_value_set_int (value, src->targetfrequency); break; case ARG_FILTERLEN: g_value_set_int (value, r->filter_length); break; case ARG_METHOD: g_value_set_enum (value, r->method); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean plugin_init (GModule *module, GstPlugin *plugin) { GstElementFactory *factory; /* create an elementfactory for the audioscale element */ factory = gst_element_factory_new ("audioscale", GST_TYPE_AUDIOSCALE, &audioscale_details); g_return_val_if_fail(factory != NULL, FALSE); gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (src_factory)); gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (sink_factory)); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); /* load support library */ if (!gst_library_load ("gstresample")) return FALSE; return TRUE; } GstPluginDesc plugin_desc = { GST_VERSION_MAJOR, GST_VERSION_MINOR, "audioscale", plugin_init };