mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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202ba2ae4a
Original commit message from CVS: State change fixes in the threading element. Added bufferpools.
301 lines
8.7 KiB
C
301 lines
8.7 KiB
C
/* Gnome-Streamer
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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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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <sys/soundcard.h>
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#include <sys/ioctl.h>
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#include <unistd.h>
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#include <gstaudiosrc.h>
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GstElementDetails gst_audiosrc_details = {
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"Audio (OSS) Source",
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"Source/Audio",
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"Read from the sound card",
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VERSION,
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"Erik Walthinsen <omega@cse.ogi.edu>",
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"(C) 1999",
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};
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/* AudioSrc signals and args */
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enum {
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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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ARG_0,
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ARG_BYTESPERREAD,
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ARG_CUROFFSET,
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ARG_FORMAT,
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ARG_CHANNELS,
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ARG_FREQUENCY,
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};
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static void gst_audiosrc_class_init(GstAudioSrcClass *klass);
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static void gst_audiosrc_init(GstAudioSrc *audiosrc);
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static void gst_audiosrc_set_arg(GtkObject *object,GtkArg *arg,guint id);
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static void gst_audiosrc_get_arg(GtkObject *object,GtkArg *arg,guint id);
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static GstElementStateReturn gst_audiosrc_change_state(GstElement *element);
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static void gst_audiosrc_close_audio(GstAudioSrc *src);
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static gboolean gst_audiosrc_open_audio(GstAudioSrc *src);
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void gst_audiosrc_sync_parms(GstAudioSrc *audiosrc);
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void gst_audiosrc_push(GstSrc *src);
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static GstSrcClass *parent_class = NULL;
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//static guint gst_audiosrc_signals[LAST_SIGNAL] = { 0 };
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GtkType
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gst_audiosrc_get_type(void) {
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static GtkType audiosrc_type = 0;
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if (!audiosrc_type) {
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static const GtkTypeInfo audiosrc_info = {
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"GstAudioSrc",
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sizeof(GstAudioSrc),
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sizeof(GstAudioSrcClass),
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(GtkClassInitFunc)gst_audiosrc_class_init,
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(GtkObjectInitFunc)gst_audiosrc_init,
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(GtkArgSetFunc)gst_audiosrc_set_arg,
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(GtkArgGetFunc)gst_audiosrc_get_arg,
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(GtkClassInitFunc)NULL,
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};
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audiosrc_type = gtk_type_unique(GST_TYPE_SRC,&audiosrc_info);
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}
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return audiosrc_type;
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}
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static void
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gst_audiosrc_class_init(GstAudioSrcClass *klass) {
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GtkObjectClass *gtkobject_class;
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GstElementClass *gstelement_class;
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GstSrcClass *gstsrc_class;
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gtkobject_class = (GtkObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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gstsrc_class = (GstSrcClass*)klass;
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parent_class = gtk_type_class(GST_TYPE_SRC);
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gtk_object_add_arg_type("GstAudioSrc::bytes_per_read", GTK_TYPE_ULONG,
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GTK_ARG_READWRITE, ARG_BYTESPERREAD);
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gtk_object_add_arg_type("GstAudioSrc::curoffset", GTK_TYPE_ULONG,
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GTK_ARG_READABLE, ARG_CUROFFSET);
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gtk_object_add_arg_type("GstAudioSrc::format", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_FORMAT);
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gtk_object_add_arg_type("GstAudioSrc::channels", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_CHANNELS);
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gtk_object_add_arg_type("GstAudioSrc::frequency", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_FREQUENCY);
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gtkobject_class->set_arg = gst_audiosrc_set_arg;
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gtkobject_class->get_arg = gst_audiosrc_get_arg;
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gstelement_class->change_state = gst_audiosrc_change_state;
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gstsrc_class->push = gst_audiosrc_push;
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}
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static void gst_audiosrc_init(GstAudioSrc *audiosrc) {
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audiosrc->srcpad = gst_pad_new("src",GST_PAD_SRC);
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gst_element_add_pad(GST_ELEMENT(audiosrc),audiosrc->srcpad);
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audiosrc->fd = -1;
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// audiosrc->meta = (MetaAudioRaw *)gst_meta_new();
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// audiosrc->meta->format = AFMT_S16_LE;
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// audiosrc->meta->channels = 2;
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// audiosrc->meta->frequency = 44100;
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// audiosrc->meta->bps = 4;
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audiosrc->bytes_per_read = 4096;
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audiosrc->curoffset = 0;
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audiosrc->seq = 0;
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}
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GstElement *gst_audiosrc_new(gchar *name) {
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GstElement *audiosrc = GST_ELEMENT(gtk_type_new(GST_TYPE_AUDIOSRC));
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gst_element_set_name(GST_ELEMENT(audiosrc),name);
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return audiosrc;
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}
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void gst_audiosrc_push(GstSrc *src) {
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GstAudioSrc *audiosrc;
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GstBuffer *buf;
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glong readbytes;
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g_return_if_fail(src != NULL);
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g_return_if_fail(GST_IS_AUDIOSRC(src));
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audiosrc = GST_AUDIOSRC(src);
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// g_print("attempting to read something from soundcard\n");
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buf = gst_buffer_new();
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g_return_if_fail(buf);
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GST_BUFFER_DATA(buf) = (gpointer)g_malloc(audiosrc->bytes_per_read);
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readbytes = read(audiosrc->fd,GST_BUFFER_DATA(buf),
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audiosrc->bytes_per_read);
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if (readbytes == 0) {
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gst_src_signal_eos(GST_SRC(audiosrc));
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return;
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}
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GST_BUFFER_SIZE(buf) = readbytes;
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GST_BUFFER_OFFSET(buf) = audiosrc->curoffset;
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audiosrc->curoffset += readbytes;
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// gst_buffer_add_meta(buf,GST_META(newmeta));
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gst_pad_push(audiosrc->srcpad,buf);
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// g_print("pushed buffer from soundcard of %d bytes\n",readbytes);
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}
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static void gst_audiosrc_set_arg(GtkObject *object,GtkArg *arg,guint id) {
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GstAudioSrc *src;
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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_AUDIOSRC(object));
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src = GST_AUDIOSRC(object);
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switch (id) {
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case ARG_BYTESPERREAD:
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src->bytes_per_read = GTK_VALUE_INT(*arg);
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break;
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case ARG_FORMAT:
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src->format = GTK_VALUE_INT(*arg);
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break;
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case ARG_CHANNELS:
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src->channels = GTK_VALUE_INT(*arg);
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break;
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case ARG_FREQUENCY:
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src->frequency = GTK_VALUE_INT(*arg);
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break;
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default:
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break;
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}
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}
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static void gst_audiosrc_get_arg(GtkObject *object,GtkArg *arg,guint id) {
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GstAudioSrc *src;
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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_AUDIOSRC(object));
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src = GST_AUDIOSRC(object);
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switch (id) {
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case ARG_BYTESPERREAD:
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GTK_VALUE_INT(*arg) = src->bytes_per_read;
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break;
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case ARG_FORMAT:
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GTK_VALUE_INT(*arg) = src->format;
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break;
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case ARG_CHANNELS:
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GTK_VALUE_INT(*arg) = src->channels;
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break;
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case ARG_FREQUENCY:
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GTK_VALUE_INT(*arg) = src->frequency;
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break;
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default:
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arg->type = GTK_TYPE_INVALID;
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break;
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}
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}
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static GstElementStateReturn gst_audiosrc_change_state(GstElement *element) {
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g_return_val_if_fail(GST_IS_AUDIOSRC(element), FALSE);
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/* if going down into NULL state, close the file if it's open */
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if (GST_STATE_PENDING(element) == GST_STATE_NULL) {
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if (GST_FLAG_IS_SET(element,GST_AUDIOSRC_OPEN))
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gst_audiosrc_close_audio(GST_AUDIOSRC(element));
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/* otherwise (READY or higher) we need to open the sound card */
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} else {
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if (!GST_FLAG_IS_SET(element,GST_AUDIOSRC_OPEN)) {
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if (!gst_audiosrc_open_audio(GST_AUDIOSRC(element)))
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return GST_STATE_FAILURE;
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}
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}
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if (GST_ELEMENT_CLASS(parent_class)->change_state)
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return GST_ELEMENT_CLASS(parent_class)->change_state(element);
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return GST_STATE_SUCCESS;
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}
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static gboolean gst_audiosrc_open_audio(GstAudioSrc *src) {
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g_return_val_if_fail(!GST_FLAG_IS_SET(src,GST_AUDIOSRC_OPEN), FALSE);
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/* first try to open the sound card */
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src->fd = open("/dev/dsp",O_RDONLY);
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/* if we have it, set the default parameters and go have fun */
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if (src->fd > 0) {
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int arg = 0x7fff0006;
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if (ioctl(src->fd, SNDCTL_DSP_SETFRAGMENT, &arg)) perror("uh");
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/* set card state */
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gst_audiosrc_sync_parms(src);
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DEBUG("opened audio\n");
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GST_FLAG_SET(src,GST_AUDIOSRC_OPEN);
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return TRUE;
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}
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return FALSE;
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}
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static void gst_audiosrc_close_audio(GstAudioSrc *src) {
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g_return_if_fail(GST_FLAG_IS_SET(src,GST_AUDIOSRC_OPEN));
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close(src->fd);
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src->fd = -1;
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GST_FLAG_UNSET(src,GST_AUDIOSRC_OPEN);
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}
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void gst_audiosrc_sync_parms(GstAudioSrc *audiosrc) {
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audio_buf_info ospace;
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g_return_if_fail(audiosrc != NULL);
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g_return_if_fail(GST_IS_AUDIOSRC(audiosrc));
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g_return_if_fail(audiosrc->fd > 0);
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ioctl(audiosrc->fd,SNDCTL_DSP_RESET,0);
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ioctl(audiosrc->fd,SNDCTL_DSP_SETFMT,&audiosrc->format);
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ioctl(audiosrc->fd,SNDCTL_DSP_CHANNELS,&audiosrc->channels);
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ioctl(audiosrc->fd,SNDCTL_DSP_SPEED,&audiosrc->frequency);
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ioctl(audiosrc->fd,SNDCTL_DSP_GETOSPACE,&ospace);
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g_print("setting sound card to %dKHz %d bit %s (%d bytes buffer)\n",
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audiosrc->frequency,audiosrc->format,
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(audiosrc->channels == 2) ? "stereo" : "mono",ospace.bytes);
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// audiosrc->meta.format = audiosrc->format;
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// audiosrc->meta.channels = audiosrc->channels;
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// audiosrc->meta.frequency = audiosrc->frequency;
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// audiosrc->sentmeta = FALSE;
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}
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