mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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fa4a4319f0
Original commit message from CVS: A first (rude) attempt at autoplug. Autoplugging selects appropriate codecs to connect src to sink, adds them to the pipeline and connect pads. Autoplugging will run the typedetect plugin if the src pad has no MIME type. No autoplugging is done on the src and sink pads, it's hardcoded: connect 'src to sink'. No attempt at creating threads. No attempt at dynamically autoplugging not yet existing pads. Changes to (some) plugins to properly set their MIME types.
355 lines
11 KiB
C
355 lines
11 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 <sys/ioctl.h>
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#include <fcntl.h>
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#include <sys/soundcard.h>
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#include <unistd.h>
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#include <gstaudiosink.h>
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#include <gst/meta/audioraw.h>
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GstElementDetails gst_audiosink_details = {
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"Audio Sink (OSS)",
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"Sink/Audio",
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"Output to a sound card via OSS",
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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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static gboolean gst_audiosink_open_audio(GstAudioSink *sink);
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static void gst_audiosink_close_audio(GstAudioSink *sink);
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static GstElementStateReturn gst_audiosink_change_state(GstElement *element);
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static void gst_audiosink_set_arg(GtkObject *object,GtkArg *arg,guint id);
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static void gst_audiosink_get_arg(GtkObject *object,GtkArg *arg,guint id);
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void gst_audiosink_chain(GstPad *pad,GstBuffer *buf);
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/* AudioSink signals and args */
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enum {
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SIGNAL_HANDOFF,
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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_MUTE,
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ARG_FORMAT,
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ARG_CHANNELS,
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ARG_FREQUENCY,
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/* FILL ME */
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};
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static void gst_audiosink_class_init(GstAudioSinkClass *klass);
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static void gst_audiosink_init(GstAudioSink *audiosink);
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static GstSinkClass *parent_class = NULL;
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static guint gst_audiosink_signals[LAST_SIGNAL] = { 0 };
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static guint16 gst_audiosink_type_audio = 0;
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GtkType
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gst_audiosink_get_type(void) {
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static GtkType audiosink_type = 0;
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if (!audiosink_type) {
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static const GtkTypeInfo audiosink_info = {
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"GstAudioSink",
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sizeof(GstAudioSink),
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sizeof(GstAudioSinkClass),
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(GtkClassInitFunc)gst_audiosink_class_init,
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(GtkObjectInitFunc)gst_audiosink_init,
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(GtkArgSetFunc)NULL,
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(GtkArgGetFunc)NULL,
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(GtkClassInitFunc)NULL,
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};
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audiosink_type = gtk_type_unique(GST_TYPE_SINK,&audiosink_info);
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}
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if (!gst_audiosink_type_audio)
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gst_audiosink_type_audio = gst_type_find_by_mime("audio/raw");
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return audiosink_type;
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}
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static void
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gst_audiosink_class_init(GstAudioSinkClass *klass) {
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GtkObjectClass *gtkobject_class;
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GstElementClass *gstelement_class;
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gtkobject_class = (GtkObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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parent_class = gtk_type_class(GST_TYPE_FILTER);
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gtk_object_add_arg_type("GstAudioSink::mute", GTK_TYPE_BOOL,
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GTK_ARG_READWRITE, ARG_MUTE);
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gtk_object_add_arg_type("GstAudioSink::format", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_FORMAT);
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gtk_object_add_arg_type("GstAudioSink::channels", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_CHANNELS);
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gtk_object_add_arg_type("GstAudioSink::frequency", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_FREQUENCY);
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gtkobject_class->set_arg = gst_audiosink_set_arg;
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gtkobject_class->get_arg = gst_audiosink_get_arg;
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gst_audiosink_signals[SIGNAL_HANDOFF] =
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gtk_signal_new("handoff",GTK_RUN_LAST,gtkobject_class->type,
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GTK_SIGNAL_OFFSET(GstAudioSinkClass,handoff),
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gtk_marshal_NONE__POINTER,GTK_TYPE_NONE,1,
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GST_TYPE_AUDIOSINK);
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gtk_object_class_add_signals(gtkobject_class,gst_audiosink_signals,
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LAST_SIGNAL);
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gstelement_class->change_state = gst_audiosink_change_state;
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}
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static void gst_audiosink_init(GstAudioSink *audiosink) {
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audiosink->sinkpad = gst_pad_new("sink",GST_PAD_SINK);
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gst_element_add_pad(GST_ELEMENT(audiosink),audiosink->sinkpad);
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gst_pad_set_type_id(audiosink->sinkpad,gst_audiosink_type_audio);
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gst_pad_set_chain_function(audiosink->sinkpad,gst_audiosink_chain);
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audiosink->fd = -1;
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audiosink->clock = gst_clock_get_system();
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gst_clock_register(audiosink->clock, GST_OBJECT(audiosink));
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//audiosink->clocktime = 0LL;
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}
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void gst_audiosink_sync_parms(GstAudioSink *audiosink) {
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audio_buf_info ospace;
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int frag;
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g_return_if_fail(audiosink != NULL);
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g_return_if_fail(GST_IS_AUDIOSINK(audiosink));
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g_return_if_fail(audiosink->fd > 0);
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ioctl(audiosink->fd,SNDCTL_DSP_RESET,0);
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ioctl(audiosink->fd,SNDCTL_DSP_SETFMT,&audiosink->format);
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ioctl(audiosink->fd,SNDCTL_DSP_CHANNELS,&audiosink->channels);
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ioctl(audiosink->fd,SNDCTL_DSP_SPEED,&audiosink->frequency);
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ioctl(audiosink->fd,SNDCTL_DSP_GETBLKSIZE, &frag);
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ioctl(audiosink->fd,SNDCTL_DSP_GETOSPACE,&ospace);
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g_print("audiosink: setting sound card to %dKHz %d bit %s (%d bytes buffer, %d fragment)\n",
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audiosink->frequency,audiosink->format,
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(audiosink->channels == 2) ? "stereo" : "mono",ospace.bytes, frag);
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}
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GstElement *gst_audiosink_new(gchar *name) {
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GstElement *audiosink = GST_ELEMENT(gtk_type_new(GST_TYPE_AUDIOSINK));
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gst_element_set_name(GST_ELEMENT(audiosink),name);
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return audiosink;
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}
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void gst_audiosink_chain(GstPad *pad,GstBuffer *buf) {
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GstAudioSink *audiosink;
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MetaAudioRaw *meta;
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gboolean in_flush;
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audio_buf_info ospace;
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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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/* this has to be an audio buffer */
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// g_return_if_fail(((GstMeta *)buf->meta)->type !=
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//gst_audiosink_type_audio);
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audiosink = GST_AUDIOSINK(pad->parent);
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// g_return_if_fail(GST_FLAG_IS_SET(audiosink,GST_STATE_RUNNING));
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if ((in_flush = GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLUSH))) {
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DEBUG("audiosink: flush\n");
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ioctl(audiosink->fd,SNDCTL_DSP_RESET,0);
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}
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meta = (MetaAudioRaw *)gst_buffer_get_first_meta(buf);
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if (meta != NULL) {
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if ((meta->format != audiosink->format) ||
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(meta->channels != audiosink->channels) ||
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(meta->frequency != audiosink->frequency)) {
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audiosink->format = meta->format;
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audiosink->channels = meta->channels;
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audiosink->frequency = meta->frequency;
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gst_audiosink_sync_parms(audiosink);
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g_print("audiosink: sound device set to format %d, %d channels, %dHz\n",
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audiosink->format,audiosink->channels,audiosink->frequency);
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}
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}
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gtk_signal_emit(GTK_OBJECT(audiosink),gst_audiosink_signals[SIGNAL_HANDOFF],
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audiosink);
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if (GST_BUFFER_DATA(buf) != NULL) {
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gst_trace_add_entry(NULL,0,buf,"audiosink: writing to soundcard");
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//g_print("audiosink: writing to soundcard\n");
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if (audiosink->fd > 2) {
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if (!audiosink->mute) {
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gst_clock_wait(audiosink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(audiosink));
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ioctl(audiosink->fd,SNDCTL_DSP_GETOSPACE,&ospace);
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DEBUG("audiosink: (%d bytes buffer)\n", ospace.bytes);
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write(audiosink->fd,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
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}
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}
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}
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//g_print("a unref\n");
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gst_buffer_unref(buf);
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//g_print("a done\n");
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}
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static void gst_audiosink_set_arg(GtkObject *object,GtkArg *arg,guint id) {
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GstAudioSink *audiosink;
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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_AUDIOSINK(object));
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audiosink = GST_AUDIOSINK(object);
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switch(id) {
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case ARG_MUTE:
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audiosink->mute = GTK_VALUE_BOOL(*arg);
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break;
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case ARG_FORMAT:
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audiosink->format = GTK_VALUE_INT(*arg);
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gst_audiosink_sync_parms(audiosink);
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break;
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case ARG_CHANNELS:
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audiosink->channels = GTK_VALUE_INT(*arg);
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gst_audiosink_sync_parms(audiosink);
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break;
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case ARG_FREQUENCY:
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audiosink->frequency = GTK_VALUE_INT(*arg);
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gst_audiosink_sync_parms(audiosink);
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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_audiosink_get_arg(GtkObject *object,GtkArg *arg,guint id) {
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GstAudioSink *audiosink;
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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_AUDIOSINK(object));
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audiosink = GST_AUDIOSINK(object);
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switch(id) {
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case ARG_MUTE:
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GTK_VALUE_BOOL(*arg) = audiosink->mute;
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break;
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case ARG_FORMAT:
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GTK_VALUE_INT(*arg) = audiosink->format;
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break;
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case ARG_CHANNELS:
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GTK_VALUE_INT(*arg) = audiosink->channels;
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break;
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case ARG_FREQUENCY:
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GTK_VALUE_INT(*arg) = audiosink->frequency;
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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 gboolean gst_audiosink_open_audio(GstAudioSink *sink) {
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g_return_val_if_fail(sink->fd == -1, FALSE);
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g_print("audiosink: attempting to open sound device\n");
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/* first try to open the sound card */
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sink->fd = open("/dev/dsp",O_RDWR);
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/* if we have it, set the default parameters and go have fun */
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if (sink->fd > 0) {
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/* set card state */
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sink->format = AFMT_S16_LE;
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sink->channels = 2; /* stereo */
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sink->frequency = 44100;
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gst_audiosink_sync_parms(sink);
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ioctl(sink->fd,SNDCTL_DSP_GETCAPS,&sink->caps);
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g_print("audiosink: Capabilities\n");
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if (sink->caps & DSP_CAP_DUPLEX) g_print("audiosink: Full duplex\n");
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if (sink->caps & DSP_CAP_REALTIME) g_print("audiosink: Realtime\n");
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if (sink->caps & DSP_CAP_BATCH) g_print("audiosink: Batch\n");
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if (sink->caps & DSP_CAP_COPROC) g_print("audiosink: Has coprocessor\n");
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if (sink->caps & DSP_CAP_TRIGGER) g_print("audiosink: Trigger\n");
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if (sink->caps & DSP_CAP_MMAP) g_print("audiosink: Direct access\n");
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g_print("audiosink: opened audio\n");
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GST_FLAG_SET(sink,GST_AUDIOSINK_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_audiosink_close_audio(GstAudioSink *sink) {
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if (sink->fd < 0) return;
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close(sink->fd);
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sink->fd = -1;
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GST_FLAG_UNSET(sink,GST_AUDIOSINK_OPEN);
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g_print("audiosink: closed sound device\n");
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}
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static GstElementStateReturn gst_audiosink_change_state(GstElement *element) {
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g_return_val_if_fail(GST_IS_AUDIOSINK(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_AUDIOSINK_OPEN))
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gst_audiosink_close_audio(GST_AUDIOSINK(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_AUDIOSINK_OPEN)) {
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if (!gst_audiosink_open_audio(GST_AUDIOSINK(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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gboolean gst_audiosink_factory_init(GstElementFactory *factory) {
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if (!gst_audiosink_type_audio)
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gst_audiosink_type_audio = gst_type_find_by_mime("audio/raw");
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gst_type_add_sink(gst_audiosink_type_audio, factory);
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return TRUE;
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}
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