gstreamer/sys/osxaudio/gstosxaudiosrc.c
Stefan Kost b5af832d7b Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_class_init):
* ext/esd/esdsink.c: (gst_esdsink_class_init):
* ext/flac/gstflactag.c: (gst_flac_tag_class_init):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
* ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
* gst/alpha/gstalpha.c: (gst_alpha_class_init):
* gst/avi/gstavimux.c: (gst_avimux_class_init):
* gst/debug/efence.c: (gst_efence_class_init):
* gst/debug/negotiation.c: (gst_negotiation_class_init):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
* gst/goom/gstgoom.c: (gst_goom_class_init):
* gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
* gst/interleave/deinterleave.c: (deinterleave_class_init):
* gst/interleave/interleave.c: (interleave_class_init):
* gst/law/alaw-decode.c: (gst_alawdec_class_init):
* gst/law/alaw-encode.c: (gst_alawenc_class_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
* gst/median/gstmedian.c: (gst_median_class_init):
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
* gst/smpte/gstsmpte.c: (gst_smpte_class_init):
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
* gst/videomixer/videomixer.c: (gst_videomixer_class_init):
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
* sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
* sys/osxaudio/gstosxaudioelement.c:
(gst_osxaudioelement_class_init):
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00

220 lines
6 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstosxaudiosrc.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 <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <CoreAudio/CoreAudio.h>
#include <gstosxaudiosrc.h>
#include <gstosxaudioelement.h>
/* elementfactory information */
static GstElementDetails gst_osxaudiosrc_details =
GST_ELEMENT_DETAILS ("Audio Source (Mac OS X)",
"Source/Audio",
"Read from the sound card",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
/* Osxaudiosrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
static GstStaticPadTemplate osxaudiosrc_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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 void gst_osxaudiosrc_base_init (gpointer g_class);
static void gst_osxaudiosrc_class_init (GstOsxAudioSrcClass * klass);
static void gst_osxaudiosrc_init (GstOsxAudioSrc * osxaudiosrc);
static void gst_osxaudiosrc_dispose (GObject * object);
static GstStateChangeReturn gst_osxaudiosrc_change_state (GstElement * element);
static GstData *gst_osxaudiosrc_get (GstPad * pad);
static GstElementClass *parent_class = NULL;
/*static guint gst_osxaudiosrc_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_osxaudiosrc_get_type (void)
{
static GType osxaudiosrc_type = 0;
if (!osxaudiosrc_type) {
static const GTypeInfo osxaudiosrc_info = {
sizeof (GstOsxAudioSrcClass),
gst_osxaudiosrc_base_init,
NULL,
(GClassInitFunc) gst_osxaudiosrc_class_init,
NULL,
NULL,
sizeof (GstOsxAudioSrc),
0,
(GInstanceInitFunc) gst_osxaudiosrc_init,
};
osxaudiosrc_type =
g_type_register_static (GST_TYPE_OSXAUDIOELEMENT, "GstOsxAudioSrc",
&osxaudiosrc_info, 0);
}
return osxaudiosrc_type;
}
static void
gst_osxaudiosrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_osxaudiosrc_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osxaudiosrc_src_factory));
}
static void
gst_osxaudiosrc_class_init (GstOsxAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_osxaudiosrc_dispose;
gstelement_class->change_state = gst_osxaudiosrc_change_state;
}
static void
gst_osxaudiosrc_init (GstOsxAudioSrc * osxaudiosrc)
{
osxaudiosrc->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&osxaudiosrc_src_factory), "src");
gst_pad_set_get_function (osxaudiosrc->srcpad, gst_osxaudiosrc_get);
gst_element_add_pad (GST_ELEMENT (osxaudiosrc), osxaudiosrc->srcpad);
}
static void
gst_osxaudiosrc_dispose (GObject * object)
{
/* GstOsxAudioSrc *osxaudiosrc = (GstOsxAudioSrc *) object; */
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstData *
gst_osxaudiosrc_get (GstPad * pad)
{
GstOsxAudioSrc *src;
GstBuffer *buf;
glong readbytes;
src = GST_OSXAUDIOSRC (gst_pad_get_parent (pad));
buf = gst_buffer_new_and_alloc ((GST_OSXAUDIOELEMENT (src))->buffer_len);
readbytes =
read_buffer (GST_OSXAUDIOELEMENT (src), (char *) GST_BUFFER_DATA (buf));
if (readbytes < 0) {
gst_buffer_unref (buf);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
gst_object_unref (src);
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
}
if (readbytes == 0) {
gst_buffer_unref (buf);
gst_object_unref (src);
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
}
GST_BUFFER_SIZE (buf) = readbytes;
GST_BUFFER_OFFSET (buf) = src->curoffset;
src->curoffset += readbytes;
GST_DEBUG ("pushed buffer from soundcard of %ld bytes", readbytes);
gst_object_unref (src);
return GST_DATA (buf);
}
static GstStateChangeReturn
gst_osxaudiosrc_change_state (GstElement * element, GstStateChange transition)
{
GstOsxAudioSrc *osxaudiosrc = GST_OSXAUDIOSRC (element);
OSErr status;
GST_DEBUG ("osxaudiosrc: state change");
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
osxaudiosrc->curoffset = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
status =
AudioDeviceStart (GST_OSXAUDIOELEMENT (osxaudiosrc)->device_id,
inputAudioDeviceIOProc);
if (status)
GST_DEBUG ("AudioDeviceStart returned %d\n", (int) status);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
status =
AudioDeviceStop (GST_OSXAUDIOELEMENT (osxaudiosrc)->device_id,
inputAudioDeviceIOProc);
if (status)
GST_DEBUG ("AudioDeviceStop returned %d\n", (int) status);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}