| /* GStreamer |
| * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
| * 2000 Wim Taymans <wtay@chello.be> |
| * |
| * 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 <gst/gst.h> |
| #include <gst/audio/audio.h> |
| #include <string.h> |
| #include <errno.h> |
| |
| #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_pad_template (element_class, |
| gst_static_pad_template_get (&afsrc_src_factory)); |
| gst_element_class_set_details_simple (element_class, "Audiofile source", |
| "Source/Audio", |
| "Read audio files from disk using libaudiofile", |
| "Thomas <thomas@apestaart.org>"); |
| } |
| |
| 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; |
| } |