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Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
220 lines
6 KiB
C
220 lines
6 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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*
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* gstosxaudiosrc.c:
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <fcntl.h>
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#include <errno.h>
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#include <unistd.h>
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#include <string.h>
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#include <CoreAudio/CoreAudio.h>
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#include <gstosxaudiosrc.h>
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#include <gstosxaudioelement.h>
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/* elementfactory information */
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static GstElementDetails gst_osxaudiosrc_details =
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GST_ELEMENT_DETAILS ("Audio Source (Mac OS X)",
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"Source/Audio",
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"Read from the sound card",
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"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
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/* Osxaudiosrc signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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static GstStaticPadTemplate osxaudiosrc_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-float, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) TRUE , "
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"width = (int) 32, " "rate = (int) 44100, " "channels = (int) 2")
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);
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static void gst_osxaudiosrc_base_init (gpointer g_class);
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static void gst_osxaudiosrc_class_init (GstOsxAudioSrcClass * klass);
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static void gst_osxaudiosrc_init (GstOsxAudioSrc * osxaudiosrc);
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static void gst_osxaudiosrc_dispose (GObject * object);
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static GstStateChangeReturn gst_osxaudiosrc_change_state (GstElement * element);
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static GstData *gst_osxaudiosrc_get (GstPad * pad);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_osxaudiosrc_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_osxaudiosrc_get_type (void)
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{
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static GType osxaudiosrc_type = 0;
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if (!osxaudiosrc_type) {
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static const GTypeInfo osxaudiosrc_info = {
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sizeof (GstOsxAudioSrcClass),
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gst_osxaudiosrc_base_init,
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NULL,
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(GClassInitFunc) gst_osxaudiosrc_class_init,
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NULL,
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NULL,
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sizeof (GstOsxAudioSrc),
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0,
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(GInstanceInitFunc) gst_osxaudiosrc_init,
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};
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osxaudiosrc_type =
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g_type_register_static (GST_TYPE_OSXAUDIOELEMENT, "GstOsxAudioSrc",
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&osxaudiosrc_info, 0);
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}
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return osxaudiosrc_type;
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}
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static void
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gst_osxaudiosrc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details (element_class, &gst_osxaudiosrc_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&osxaudiosrc_src_factory));
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}
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static void
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gst_osxaudiosrc_class_init (GstOsxAudioSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_OSXAUDIOELEMENT);
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gobject_class->dispose = gst_osxaudiosrc_dispose;
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gstelement_class->change_state = gst_osxaudiosrc_change_state;
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}
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static void
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gst_osxaudiosrc_init (GstOsxAudioSrc * osxaudiosrc)
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{
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osxaudiosrc->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&osxaudiosrc_src_factory), "src");
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gst_pad_set_get_function (osxaudiosrc->srcpad, gst_osxaudiosrc_get);
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gst_element_add_pad (GST_ELEMENT (osxaudiosrc), osxaudiosrc->srcpad);
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}
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static void
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gst_osxaudiosrc_dispose (GObject * object)
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{
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/* GstOsxAudioSrc *osxaudiosrc = (GstOsxAudioSrc *) object; */
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static GstData *
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gst_osxaudiosrc_get (GstPad * pad)
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{
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GstOsxAudioSrc *src;
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GstBuffer *buf;
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glong readbytes;
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src = GST_OSXAUDIOSRC (gst_pad_get_parent (pad));
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buf = gst_buffer_new_and_alloc ((GST_OSXAUDIOELEMENT (src))->buffer_len);
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readbytes =
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read_buffer (GST_OSXAUDIOELEMENT (src), (char *) GST_BUFFER_DATA (buf));
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if (readbytes < 0) {
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gst_buffer_unref (buf);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
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gst_object_unref (src);
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return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
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}
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if (readbytes == 0) {
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gst_buffer_unref (buf);
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gst_object_unref (src);
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return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
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}
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GST_BUFFER_SIZE (buf) = readbytes;
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GST_BUFFER_OFFSET (buf) = src->curoffset;
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src->curoffset += readbytes;
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GST_DEBUG ("pushed buffer from soundcard of %ld bytes", readbytes);
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gst_object_unref (src);
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return GST_DATA (buf);
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}
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static GstStateChangeReturn
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gst_osxaudiosrc_change_state (GstElement * element, GstStateChange transition)
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{
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GstOsxAudioSrc *osxaudiosrc = GST_OSXAUDIOSRC (element);
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OSErr status;
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GST_DEBUG ("osxaudiosrc: state change");
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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osxaudiosrc->curoffset = 0;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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status =
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AudioDeviceStart (GST_OSXAUDIOELEMENT (osxaudiosrc)->device_id,
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inputAudioDeviceIOProc);
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if (status)
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GST_DEBUG ("AudioDeviceStart returned %d\n", (int) status);
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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status =
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AudioDeviceStop (GST_OSXAUDIOELEMENT (osxaudiosrc)->device_id,
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inputAudioDeviceIOProc);
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if (status)
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GST_DEBUG ("AudioDeviceStop returned %d\n", (int) status);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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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, transition);
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return GST_STATE_CHANGE_SUCCESS;
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}
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