mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 16:50:47 +00:00
9c6ea1f0c8
Original commit message from CVS: zaheer : * gst/tcp/gsttcp.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpclientsrc.h: * gst/tcp/gsttcpserversrc.c: - portability fix, to compile on OSX (fixes #143146) * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: - compilation warnings on OSX (fixes #143153) me : * ext/vorbis/vorbisdec.c : sign warning fixes * gst-libs/gst/mixer/mixertrack.c : forgoten include to define newly used G_MAXINT32, bad owen, bad
229 lines
6.5 KiB
C
229 lines
6.5 KiB
C
/* GStreamer
|
|
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
|
|
* 2000 Wim Taymans <wim.taymans@chello.be>
|
|
*
|
|
* gstosxaudiosink.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 <CoreAudio/CoreAudio.h>
|
|
#include <errno.h>
|
|
#include <unistd.h>
|
|
#include <string.h>
|
|
|
|
#include "gstosxaudiosink.h"
|
|
|
|
/* elementfactory information */
|
|
static GstElementDetails gst_osxaudiosink_details =
|
|
GST_ELEMENT_DETAILS ("Audio Sink (Mac OS X)",
|
|
"Sink/Audio",
|
|
"Output to a Mac OS X CoreAudio Sound Device",
|
|
"Zaheer Abbas Merali <zaheerabbas at merali.org>");
|
|
|
|
static void gst_osxaudiosink_base_init (gpointer g_class);
|
|
static void gst_osxaudiosink_class_init (GstOsxAudioSinkClass * klass);
|
|
static void gst_osxaudiosink_init (GstOsxAudioSink * osxaudiosink);
|
|
static void gst_osxaudiosink_dispose (GObject * object);
|
|
|
|
static GstElementStateReturn gst_osxaudiosink_change_state (GstElement *
|
|
element);
|
|
|
|
static void gst_osxaudiosink_chain (GstPad * pad, GstData * _data);
|
|
|
|
/* OssSink signals and args */
|
|
enum
|
|
{
|
|
SIGNAL_HANDOFF,
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
static GstStaticPadTemplate osxaudiosink_sink_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-float, "
|
|
"endianness = (int) BYTE_ORDER, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 32, " "rate = (int) 44100, " "channels = (int) 2")
|
|
);
|
|
|
|
static GstElementClass *parent_class = NULL;
|
|
static guint gst_osssink_signals[LAST_SIGNAL] = { 0 };
|
|
|
|
GType
|
|
gst_osxaudiosink_get_type (void)
|
|
{
|
|
static GType osxaudiosink_type = 0;
|
|
|
|
if (!osxaudiosink_type) {
|
|
static const GTypeInfo osxaudiosink_info = {
|
|
sizeof (GstOsxAudioSinkClass),
|
|
gst_osxaudiosink_base_init,
|
|
NULL,
|
|
(GClassInitFunc) gst_osxaudiosink_class_init,
|
|
NULL,
|
|
NULL,
|
|
sizeof (GstOsxAudioSink),
|
|
0,
|
|
(GInstanceInitFunc) gst_osxaudiosink_init,
|
|
};
|
|
|
|
osxaudiosink_type =
|
|
g_type_register_static (GST_TYPE_OSXAUDIOELEMENT, "GstOsxAudioSink",
|
|
&osxaudiosink_info, 0);
|
|
}
|
|
|
|
return osxaudiosink_type;
|
|
}
|
|
|
|
static void
|
|
gst_osxaudiosink_dispose (GObject * object)
|
|
{
|
|
/* GstOsxAudioSink *osxaudiosink = (GstOsxAudioSink *) object; */
|
|
|
|
/*gst_object_unparent (GST_OBJECT (osxaudiosink->provided_clock)); */
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_osxaudiosink_base_init (gpointer g_class)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_set_details (element_class, &gst_osxaudiosink_details);
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&osxaudiosink_sink_factory));
|
|
}
|
|
static void
|
|
gst_osxaudiosink_class_init (GstOsxAudioSinkClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
|
|
parent_class = g_type_class_ref (GST_TYPE_OSXAUDIOELEMENT);
|
|
|
|
gst_osssink_signals[SIGNAL_HANDOFF] =
|
|
g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
|
|
G_STRUCT_OFFSET (GstOsxAudioSinkClass, handoff), NULL, NULL,
|
|
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
|
|
|
|
gobject_class->dispose = gst_osxaudiosink_dispose;
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_osxaudiosink_change_state);
|
|
}
|
|
|
|
static void
|
|
gst_osxaudiosink_init (GstOsxAudioSink * osxaudiosink)
|
|
{
|
|
osxaudiosink->sinkpad =
|
|
gst_pad_new_from_template (gst_static_pad_template_get
|
|
(&osxaudiosink_sink_factory), "sink");
|
|
gst_element_add_pad (GST_ELEMENT (osxaudiosink), osxaudiosink->sinkpad);
|
|
|
|
gst_pad_set_chain_function (osxaudiosink->sinkpad, gst_osxaudiosink_chain);
|
|
|
|
GST_DEBUG ("initializing osxaudiosink");
|
|
|
|
GST_FLAG_SET (osxaudiosink, GST_ELEMENT_THREAD_SUGGESTED);
|
|
GST_FLAG_SET (osxaudiosink, GST_ELEMENT_EVENT_AWARE);
|
|
}
|
|
|
|
static void
|
|
gst_osxaudiosink_chain (GstPad * pad, GstData * _data)
|
|
{
|
|
GstBuffer *buf = GST_BUFFER (_data);
|
|
GstOsxAudioSink *osxaudiosink;
|
|
guchar *data;
|
|
guint to_write;
|
|
gint amount_written;
|
|
|
|
/* this has to be an audio buffer */
|
|
osxaudiosink = GST_OSXAUDIOSINK (gst_pad_get_parent (pad));
|
|
|
|
if (GST_IS_EVENT (buf)) {
|
|
GstEvent *event = GST_EVENT (buf);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
gst_pad_event_default (pad, event);
|
|
return;
|
|
case GST_EVENT_DISCONTINUOUS:
|
|
/* pass-through */
|
|
default:
|
|
gst_pad_event_default (pad, event);
|
|
return;
|
|
}
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
data = GST_BUFFER_DATA (buf);
|
|
to_write = GST_BUFFER_SIZE (buf);
|
|
amount_written = 0;
|
|
|
|
while (amount_written < to_write) {
|
|
data += amount_written;
|
|
to_write -= amount_written;
|
|
amount_written =
|
|
write_buffer (GST_OSXAUDIOELEMENT (osxaudiosink), data, to_write);
|
|
}
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_osxaudiosink_change_state (GstElement * element)
|
|
{
|
|
GstOsxAudioSink *osxaudiosink;
|
|
OSErr status;
|
|
|
|
osxaudiosink = GST_OSXAUDIOSINK (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
status =
|
|
AudioDeviceStart (GST_OSXAUDIOELEMENT (osxaudiosink)->device_id,
|
|
outputAudioDeviceIOProc);
|
|
if (status)
|
|
GST_DEBUG ("AudioDeviceStart returned %d\n", (int) status);
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
status =
|
|
AudioDeviceStop (GST_OSXAUDIOELEMENT (osxaudiosink)->device_id,
|
|
outputAudioDeviceIOProc);
|
|
if (status)
|
|
GST_DEBUG ("AudioDeviceStop returned %d\n", (int) status);
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|