mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-03 05:59:44 +00:00
9920b88c3f
Original commit message from CVS: This is the audio/video sync release. Changed the mpegvideoparser to parse complete pictures. Added the PTS timestamps to the pictures. Added PTS timestamps to the MPEG audio frames. Made the clock a littlebit better. Gstplay now uses two more threads one for video, one for audio playback. Added the first QoS callbacks for the pads. hopefully fix an mmx compilation problem.
376 lines
12 KiB
C
376 lines
12 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 gboolean gst_audiosink_start(GstElement *element,
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GstElementState state);
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static gboolean gst_audiosink_stop(GstElement *element);
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static gboolean gst_audiosink_change_state(GstElement *element,
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GstElementState state);
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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->start = gst_audiosink_start;
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gstelement_class->stop = gst_audiosink_stop;
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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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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_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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gst_element_set_state(GST_ELEMENT(audiosink),GST_STATE_COMPLETE);
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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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if (gst_clock_current_diff(audiosink->clock, GST_BUFFER_TIMESTAMP(buf)) > 500000) {
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}
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else {
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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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//gst_clock_set(audiosink->clock, GST_BUFFER_TIMESTAMP(buf));
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}
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}
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}
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}
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end:
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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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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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g_print("audiosink: closed sound device\n");
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}
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static gboolean gst_audiosink_start(GstElement *element,
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GstElementState state) {
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g_return_val_if_fail(GST_IS_AUDIOSINK(element), FALSE);
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if (gst_audiosink_open_audio(GST_AUDIOSINK(element)) == TRUE) {
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gst_element_set_state(element,GST_STATE_RUNNING | state);
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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 gboolean gst_audiosink_stop(GstElement *element) {
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g_return_val_if_fail(GST_IS_AUDIOSINK(element), FALSE);
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gst_audiosink_close_audio(GST_AUDIOSINK(element));
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gst_element_set_state(element,~GST_STATE_RUNNING);
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return TRUE;
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}
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static gboolean gst_audiosink_change_state(GstElement *element,
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GstElementState state) {
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g_return_val_if_fail(GST_IS_AUDIOSINK(element), FALSE);
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switch (state) {
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case GST_STATE_RUNNING:
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if (!gst_audiosink_open_audio(GST_AUDIOSINK(element)))
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return FALSE;
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break;
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case ~GST_STATE_RUNNING:
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gst_audiosink_close_audio(GST_AUDIOSINK(element));
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break;
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default:
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break;
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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,state);
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return TRUE;
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}
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