mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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de1357a407
Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_finalize): * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init), (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init), (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose), (gst_gconf_audio_src_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init), (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init), (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose), (gst_gconf_video_src_finalize), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_set_child): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init), (gst_shout2send_init), (gst_shout2send_finalize): * gst/debug/testplugin.c: (gst_test_class_init), (gst_test_finalize): * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (gst_flxdec_dispose): * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context): * gst/rtsp/rtspextwms.h: * gst/smpte/gstsmpte.c: (gst_smpte_class_init), (gst_smpte_finalize): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize): * gst/udp/gstudpsink.c: (gst_udpsink_class_init), (gst_udpsink_finalize): * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose), (gst_wavparse_sink_activate): * sys/oss/gstosssink.c: (gst_oss_sink_finalise): * sys/oss/gstosssrc.c: (gst_oss_src_class_init), (gst_oss_src_finalize): * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_finalize): * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get): Fix a bunch of leaks shown by the newly-added states test.
566 lines
14 KiB
C
566 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* gstosssink.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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/**
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* SECTION:element-osssink
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* @short_description: output sound using OSS
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*
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* <refsect2>
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* <para>
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* This element lets you output sound using the Open Sound System (OSS).
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* </para>
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* <para>
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* Note that you should almost always use generic audio conversion elements
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* like audioconvert and audioresample in front of an audiosink to make sure
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* your pipeline works under all circumstances (those conversion elements will
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* act in passthrough-mode if no conversion is necessary).
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink
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* </programlisting>
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* will output a sine wave (continuous beep sound) to your sound card (with
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* a very low volume as precaution).
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* </para>
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* <para>
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* <programlisting>
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* gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink
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* </programlisting>
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* will play an Ogg/Vorbis audio file and output it using the Open Sound System.
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* </para>
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* </refsect2>
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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 <sys/ioctl.h>
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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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#ifdef HAVE_OSS_INCLUDE_IN_SYS
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# include <sys/soundcard.h>
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#else
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# ifdef HAVE_OSS_INCLUDE_IN_ROOT
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# include <soundcard.h>
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# else
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# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
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# include <machine/soundcard.h>
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# else
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# error "What to include?"
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# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
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# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
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#endif /* HAVE_OSS_INCLUDE_IN_SYS */
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#include "common.h"
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#include "gstosssink.h"
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GST_DEBUG_CATEGORY_EXTERN (oss_debug);
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#define GST_CAT_DEFAULT oss_debug
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/* elementfactory information */
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static const GstElementDetails gst_oss_sink_details =
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GST_ELEMENT_DETAILS ("Audio Sink (OSS)",
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"Sink/Audio",
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"Output to a sound card via OSS",
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"Erik Walthinsen <omega@cse.ogi.edu>, "
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"Wim Taymans <wim.taymans@chello.be>");
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static void gst_oss_sink_base_init (gpointer g_class);
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static void gst_oss_sink_class_init (GstOssSinkClass * klass);
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static void gst_oss_sink_init (GstOssSink * osssink);
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static void gst_oss_sink_dispose (GObject * object);
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static void gst_oss_sink_finalise (GObject * object);
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static void gst_oss_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_oss_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink);
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static gboolean gst_oss_sink_open (GstAudioSink * asink);
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static gboolean gst_oss_sink_close (GstAudioSink * asink);
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static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
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static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_oss_sink_delay (GstAudioSink * asink);
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static void gst_oss_sink_reset (GstAudioSink * asink);
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/* OssSink signals and args */
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enum
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{
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LAST_SIGNAL
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};
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#define DEFAULT_DEVICE "/dev/dsp"
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enum
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{
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PROP_0,
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PROP_DEVICE,
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};
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static GstStaticPadTemplate osssink_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
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);
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static GstElementClass *parent_class = NULL;
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/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_oss_sink_get_type (void)
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{
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static GType osssink_type = 0;
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if (!osssink_type) {
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static const GTypeInfo osssink_info = {
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sizeof (GstOssSinkClass),
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gst_oss_sink_base_init,
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NULL,
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(GClassInitFunc) gst_oss_sink_class_init,
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NULL,
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NULL,
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sizeof (GstOssSink),
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0,
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(GInstanceInitFunc) gst_oss_sink_init,
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};
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osssink_type =
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g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink",
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&osssink_info, 0);
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}
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return osssink_type;
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}
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static void
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gst_oss_sink_dispose (GObject * object)
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{
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GstOssSink *osssink = GST_OSSSINK (object);
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if (osssink->probed_caps) {
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gst_caps_unref (osssink->probed_caps);
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osssink->probed_caps = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_oss_sink_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_oss_sink_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&osssink_sink_factory));
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}
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static void
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gst_oss_sink_class_init (GstOssSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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GstBaseAudioSinkClass *gstbaseaudiosink_class;
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GstAudioSinkClass *gstaudiosink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
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gstaudiosink_class = (GstAudioSinkClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_sink_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_oss_sink_finalise);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_sink_get_property);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_sink_set_property);
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE));
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
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}
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static void
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gst_oss_sink_init (GstOssSink * osssink)
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{
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GST_DEBUG_OBJECT (osssink, "initializing osssink");
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osssink->device = g_strdup (DEFAULT_DEVICE);
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osssink->fd = -1;
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}
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static void
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gst_oss_sink_finalise (GObject * object)
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{
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GstOssSink *osssink = GST_OSSSINK (object);
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g_free (osssink->device);
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G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));
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}
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static void
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gst_oss_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOssSink *sink;
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sink = GST_OSSSINK (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_free (sink->device);
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sink->device = g_value_dup_string (value);
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if (sink->probed_caps) {
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gst_caps_unref (sink->probed_caps);
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sink->probed_caps = NULL;
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}
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_oss_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOssSink *sink;
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sink = GST_OSSSINK (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_value_set_string (value, sink->device);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstCaps *
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gst_oss_sink_getcaps (GstBaseSink * bsink)
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{
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GstOssSink *osssink;
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GstCaps *caps;
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osssink = GST_OSSSINK (bsink);
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if (osssink->fd == -1) {
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
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(bsink)));
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} else if (osssink->probed_caps) {
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caps = gst_caps_copy (osssink->probed_caps);
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} else {
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caps = gst_oss_helper_probe_caps (osssink->fd);
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if (caps && !gst_caps_is_empty (caps)) {
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osssink->probed_caps = gst_caps_copy (caps);
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}
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}
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return caps;
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}
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static gint
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ilog2 (gint x)
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{
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/* well... hacker's delight explains... */
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x = x | (x >> 1);
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x = x | (x >> 2);
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x = x | (x >> 4);
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x = x | (x >> 8);
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x = x | (x >> 16);
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x = x - ((x >> 1) & 0x55555555);
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x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
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x = (x + (x >> 4)) & 0x0f0f0f0f;
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x = x + (x >> 8);
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x = x + (x >> 16);
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return (x & 0x0000003f) - 1;
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}
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static gint
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gst_oss_sink_get_format (GstBufferFormat fmt)
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{
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gint result;
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switch (fmt) {
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case GST_MU_LAW:
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result = AFMT_MU_LAW;
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break;
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case GST_A_LAW:
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result = AFMT_A_LAW;
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break;
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case GST_IMA_ADPCM:
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result = AFMT_IMA_ADPCM;
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break;
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case GST_U8:
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result = AFMT_U8;
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break;
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case GST_S16_LE:
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result = AFMT_S16_LE;
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break;
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case GST_S16_BE:
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result = AFMT_S16_BE;
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break;
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case GST_S8:
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result = AFMT_S8;
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break;
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case GST_U16_LE:
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result = AFMT_U16_LE;
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break;
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case GST_U16_BE:
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result = AFMT_U16_BE;
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break;
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case GST_MPEG:
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result = AFMT_MPEG;
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break;
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default:
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result = 0;
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break;
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}
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return result;
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}
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static gboolean
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gst_oss_sink_open (GstAudioSink * asink)
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{
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GstOssSink *oss;
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int mode;
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oss = GST_OSSSINK (asink);
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mode = O_WRONLY;
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mode |= O_NONBLOCK;
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oss->fd = open (oss->device, mode, 0);
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if (oss->fd == -1) {
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switch (errno) {
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case EBUSY:
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goto busy;
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default:
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goto open_failed;
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}
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}
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return TRUE;
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/* ERRORS */
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busy:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, BUSY, (NULL), (NULL));
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return FALSE;
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}
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open_failed:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, (NULL), GST_ERROR_SYSTEM);
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return FALSE;
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}
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}
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static gboolean
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gst_oss_sink_close (GstAudioSink * asink)
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{
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close (GST_OSSSINK (asink)->fd);
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GST_OSSSINK (asink)->fd = -1;
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return TRUE;
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}
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static gboolean
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gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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{
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GstOssSink *oss;
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struct audio_buf_info info;
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int mode;
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int tmp;
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oss = GST_OSSSINK (asink);
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/* we opened non-blocking so that we can detect if the device is available
|
|
* without hanging forever. We now want to remove the non-blocking flag. */
|
|
mode = fcntl (oss->fd, F_GETFL);
|
|
mode &= ~O_NONBLOCK;
|
|
if (fcntl (oss->fd, F_SETFL, mode) == -1) {
|
|
/* some drivers do no support unsetting the non-blocking flag, try to
|
|
* close/open the device then. This is racy but we error out properly. */
|
|
gst_oss_sink_close (asink);
|
|
if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1)
|
|
goto non_block;
|
|
}
|
|
|
|
tmp = gst_oss_sink_get_format (spec->format);
|
|
if (tmp == 0)
|
|
goto wrong_format;
|
|
|
|
if (spec->width != 16 && spec->width != 8)
|
|
goto dodgy_width;
|
|
|
|
SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
|
|
if (spec->channels == 2)
|
|
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
|
|
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
|
|
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
|
|
|
|
tmp = ilog2 (spec->segsize);
|
|
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
|
|
GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
|
|
spec->segsize, spec->segtotal, tmp);
|
|
|
|
SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
|
|
GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");
|
|
|
|
spec->segsize = info.fragsize;
|
|
spec->segtotal = info.fragstotal;
|
|
|
|
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
|
|
oss->bytes_per_sample = (spec->width / 8) * spec->channels;
|
|
|
|
GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
|
|
spec->segsize, spec->segtotal, tmp);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
non_block:
|
|
{
|
|
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set device %s in non blocking mode: %s",
|
|
oss->device, g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
wrong_format:
|
|
{
|
|
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get format %d", spec->format));
|
|
return FALSE;
|
|
}
|
|
dodgy_width:
|
|
{
|
|
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
|
|
("unexpected width %d", spec->width));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_oss_sink_unprepare (GstAudioSink * asink)
|
|
{
|
|
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
|
|
|
|
if (!gst_oss_sink_close (asink))
|
|
goto couldnt_close;
|
|
|
|
if (!gst_oss_sink_open (asink))
|
|
goto couldnt_reopen;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
couldnt_close:
|
|
{
|
|
GST_DEBUG_OBJECT (asink, "Could not close the audio device");
|
|
return FALSE;
|
|
}
|
|
couldnt_reopen:
|
|
{
|
|
GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static guint
|
|
gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
return write (GST_OSSSINK (asink)->fd, data, length);
|
|
}
|
|
|
|
static guint
|
|
gst_oss_sink_delay (GstAudioSink * asink)
|
|
{
|
|
GstOssSink *oss;
|
|
gint delay = 0;
|
|
gint ret;
|
|
|
|
oss = GST_OSSSINK (asink);
|
|
|
|
#ifdef SNDCTL_DSP_GETODELAY
|
|
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
|
|
#else
|
|
ret = -1;
|
|
#endif
|
|
if (ret < 0) {
|
|
audio_buf_info info;
|
|
|
|
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
|
|
|
|
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
|
|
}
|
|
return delay / oss->bytes_per_sample;
|
|
}
|
|
|
|
static void
|
|
gst_oss_sink_reset (GstAudioSink * asink)
|
|
{
|
|
#if 0
|
|
GstOssSink *oss;
|
|
gint ret;
|
|
|
|
oss = GST_OSSSINK (asink);
|
|
|
|
/* deadlocks on my machine... */
|
|
ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
|
|
#endif
|
|
}
|