/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000 Wim Taymans * * gstafsrc.c: * * 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 "gst/gst-i18n-plugin.h" #include #include #include #include #include "gstafsrc.h" /* AFSrc signals and args */ enum { /* FILL ME */ SIGNAL_HANDOFF, LAST_SIGNAL }; enum { ARG_0, ARG_LOCATION }; /* added a src factory function to force audio/raw MIME type */ /* I think the caps can be broader, we need to change that somehow */ static GstStaticPadTemplate afsrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "width = (int) { 8, 16 }, " "depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }") ); /* we use an enum for the output type arg */ #define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type()) /* FIXME: fix the string ints to be string-converted from the audiofile.h types */ /* defined but not used static GType gst_afsrc_types_get_type (void) { static GType afsrc_types_type = 0; static GEnumValue afsrc_types[] = { {AF_FILE_RAWDATA, "0", "raw PCM"}, {AF_FILE_AIFFC, "1", "AIFFC"}, {AF_FILE_AIFF, "2", "AIFF"}, {AF_FILE_NEXTSND, "3", "Next/SND"}, {AF_FILE_WAVE, "4", "Wave"}, {0, NULL, NULL}, }; if (!afsrc_types_type) { afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types); } return afsrc_types_type; } */ static void gst_afsrc_base_init (gpointer g_class); static void gst_afsrc_class_init (GstAFSrcClass * klass); static void gst_afsrc_init (GstAFSrc * afsrc); static gboolean gst_afsrc_open_file (GstAFSrc * src); static void gst_afsrc_close_file (GstAFSrc * src); static GstData *gst_afsrc_get (GstPad * pad); static void gst_afsrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_afsrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_afsrc_change_state (GstElement * element, GstStateChange transition); static GstElementClass *parent_class = NULL; static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 }; GType gst_afsrc_get_type (void) { static GType afsrc_type = 0; if (!afsrc_type) { static const GTypeInfo afsrc_info = { sizeof (GstAFSrcClass), gst_afsrc_base_init, NULL, (GClassInitFunc) gst_afsrc_class_init, NULL, NULL, sizeof (GstAFSrc), 0, (GInstanceInitFunc) gst_afsrc_init, }; afsrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0); } return afsrc_type; } static void gst_afsrc_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_static_pad_template (element_class, &afsrc_src_factory); gst_element_class_set_details_simple (element_class, "Audiofile source", "Source/Audio", "Read audio files from disk using libaudiofile", "Thomas "); } static void gst_afsrc_class_init (GstAFSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass), "location", ARG_LOCATION, G_PARAM_READWRITE, NULL); gst_afsrc_signals[SIGNAL_HANDOFF] = g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAFSrcClass, handoff), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0); gobject_class->set_property = gst_afsrc_set_property; gobject_class->get_property = gst_afsrc_get_property; gstelement_class->change_state = gst_afsrc_change_state; } static void gst_afsrc_init (GstAFSrc * afsrc) { /* no need for a template, caps are set based on file, right ? */ afsrc->srcpad = gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT (afsrc), "src"), "src"); gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad); gst_pad_use_explicit_caps (afsrc->srcpad); gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get); afsrc->bytes_per_read = 4096; afsrc->curoffset = 0; afsrc->seq = 0; afsrc->filename = NULL; afsrc->file = NULL; /* default values, should never be needed */ afsrc->channels = 2; afsrc->width = 16; afsrc->rate = 44100; afsrc->type = AF_FILE_WAVE; afsrc->endianness_data = 1234; afsrc->endianness_wanted = 1234; afsrc->framestamp = 0; } static GstData * gst_afsrc_get (GstPad * pad) { GstAFSrc *src; GstBuffer *buf; glong readbytes, readframes; glong frameCount; g_return_val_if_fail (pad != NULL, NULL); src = GST_AFSRC (gst_pad_get_parent (pad)); buf = gst_buffer_new (); g_return_val_if_fail (buf, NULL); GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read); /* calculate frameCount to read based on file info */ frameCount = src->bytes_per_read / (src->channels * src->width / 8); /* g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount); */ readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf), frameCount); readbytes = readframes * (src->channels * src->width / 8); if (readbytes == 0) { gst_element_set_eos (GST_ELEMENT (src)); return GST_DATA (gst_event_new (GST_EVENT_EOS)); } GST_BUFFER_SIZE (buf) = readbytes; GST_BUFFER_OFFSET (buf) = src->curoffset; src->curoffset += readbytes; src->framestamp += gst_audio_frame_length (src->srcpad, buf); GST_BUFFER_TIMESTAMP (buf) = src->framestamp * 1E9 / gst_audio_frame_rate (src->srcpad); /* printf ("DEBUG: afsrc: timestamp set on output buffer: %f sec\n", GST_BUFFER_TIMESTAMP (buf) / 1E9); */ /* g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes); */ return GST_DATA (buf); } static void gst_afsrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAFSrc *src; src = GST_AFSRC (object); switch (prop_id) { case ARG_LOCATION: if (src->filename) g_free (src->filename); src->filename = g_strdup (g_value_get_string (value)); break; default: break; } } static void gst_afsrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAFSrc *src; g_return_if_fail (GST_IS_AFSRC (object)); src = GST_AFSRC (object); switch (prop_id) { case ARG_LOCATION: g_value_set_string (value, src->filename); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } gboolean gst_afsrc_plugin_init (GstPlugin * plugin) { /* load audio support library */ if (!gst_library_load ("gstaudio")) return FALSE; if (!gst_element_register (plugin, "afsrc", GST_RANK_NONE, GST_TYPE_AFSRC)) return FALSE; #ifdef ENABLE_NLS setlocale (LC_ALL, ""); bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); #endif /* ENABLE_NLS */ return TRUE; } /* this is where we open the audiofile */ static gboolean gst_afsrc_open_file (GstAFSrc * src) { g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE); /* open the file */ src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP); if (src->file == AF_NULL_FILEHANDLE) { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (_("Could not open file \"%s\" for reading."), src->filename), ("system error: %s", strerror (errno))); return FALSE; } /* get the audiofile audio parameters */ { int sampleFormat, sampleWidth; src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK); afGetSampleFormat (src->file, AF_DEFAULT_TRACK, &sampleFormat, &sampleWidth); switch (sampleFormat) { case AF_SAMPFMT_TWOSCOMP: src->is_signed = TRUE; break; case AF_SAMPFMT_UNSIGNED: src->is_signed = FALSE; break; case AF_SAMPFMT_FLOAT: case AF_SAMPFMT_DOUBLE: GST_DEBUG ("ERROR: float data not supported yet !\n"); } src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK); src->width = sampleWidth; GST_DEBUG ("input file: %d channels, %d width, %d rate, signed %s\n", src->channels, src->width, src->rate, src->is_signed ? "yes" : "no"); } /* set caps on src */ gst_pad_set_explicit_caps (src->srcpad, gst_caps_new_simple ("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, src->is_signed, "width", G_TYPE_INT, src->width, "depth", G_TYPE_INT, src->width, "rate", G_TYPE_INT, src->rate, "channels", G_TYPE_INT, src->channels, NULL)); GST_OBJECT_FLAG_SET (src, GST_AFSRC_OPEN); return TRUE; } static void gst_afsrc_close_file (GstAFSrc * src) { /* g_print ("DEBUG: closing srcfile...\n"); */ g_return_if_fail (GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN)); /* g_print ("DEBUG: past flag test\n"); */ /* if (fclose (src->file) != 0) */ if (afCloseFile (src->file) != 0) { GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (_("Error closing file \"%s\"."), src->filename), GST_ERROR_SYSTEM); } else { GST_OBJECT_FLAG_UNSET (src, GST_AFSRC_OPEN); } } static GstStateChangeReturn gst_afsrc_change_state (GstElement * element, GstStateChange transition) { g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_CHANGE_FAILURE); /* if going to NULL then close the file */ if (GST_STATE_PENDING (element) == GST_STATE_NULL) { /* printf ("DEBUG: afsrc state change: null pending\n"); */ if (GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) { /* g_print ("DEBUG: trying to close the src file\n"); */ gst_afsrc_close_file (GST_AFSRC (element)); } } else if (GST_STATE_PENDING (element) == GST_STATE_READY) { /* g_print ("DEBUG: afsrc: ready state pending. This shouldn't happen at the *end* of a stream\n"); */ if (!GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) { /* g_print ("DEBUG: GST_AFSRC_OPEN not set\n"); */ if (!gst_afsrc_open_file (GST_AFSRC (element))) { /* g_print ("DEBUG: element tries to open file\n"); */ return GST_STATE_CHANGE_FAILURE; } } } if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); return GST_STATE_CHANGE_SUCCESS; }