mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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687ccc01c7
Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
420 lines
13 KiB
C
420 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
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*
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* gstdirectsoundsink.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 "gstdirectsoundsink.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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GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
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#define GST_CAT_DEFAULT directsoundsink_debug
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/* elementfactory information */
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static GstElementDetails gst_directsoundsink_details =
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GST_ELEMENT_DETAILS ("Audio Sink (DIRECTSOUND)",
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"Sink/Audio",
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"Output to a sound card via DIRECTSOUND",
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"Sebastien Moutte <sebastien@moutte.net>");
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static void gst_directsoundsink_base_init (gpointer g_class);
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static void gst_directsoundsink_class_init (GstDirectSoundSinkClass * klass);
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static void gst_directsoundsink_init (GstDirectSoundSink * alsasink);
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static void gst_directsoundsink_dispose (GObject * object);
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static void gst_directsoundsink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_directsoundsink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstCaps *gst_directsoundsink_getcaps (GstBaseSink * bsink);
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static gboolean gst_directsoundsink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static gboolean gst_directsoundsink_unprepare (GstAudioSink * asink);
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static gboolean gst_directsoundsink_open (GstAudioSink * asink);
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static gboolean gst_directsoundsink_close (GstAudioSink * asink);
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static guint gst_directsoundsink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_directsoundsink_delay (GstAudioSink * asink);
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static void gst_directsoundsink_reset (GstAudioSink * asink);
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static GstStaticPadTemplate directsoundsink_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) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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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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static void
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_do_init (GType directsoundsink_type)
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{
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GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
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"DirectSound sink");
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}
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GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsoundsink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, _do_init);
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static void
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gst_directsoundsink_dispose (GObject * object)
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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_directsoundsink_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_directsoundsink_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&directsoundsink_sink_factory));
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}
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static void
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gst_directsoundsink_class_init (GstDirectSoundSinkClass * 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_directsoundsink_dispose);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_directsoundsink_get_property);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_directsoundsink_set_property);
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_directsoundsink_getcaps);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_directsoundsink_prepare);
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gstaudiosink_class->unprepare =
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GST_DEBUG_FUNCPTR (gst_directsoundsink_unprepare);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_directsoundsink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_directsoundsink_close);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsoundsink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsoundsink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsoundsink_reset);
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}
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static void
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gst_directsoundsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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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_directsoundsink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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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_directsoundsink_init (GstDirectSoundSink * dsoundsink,
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GstDirectSoundSinkClass g_class)
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{
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GST_DEBUG ("initializing directsoundsink");
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dsoundsink->pDS = NULL;
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dsoundsink->pDSBSecondary = NULL;
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dsoundsink->current_circular_offset = 0;
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dsoundsink->buffer_size = DSBSIZE_MIN;
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}
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static GstCaps *
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gst_directsoundsink_getcaps (GstBaseSink * bsink)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (bsink);
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return
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gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
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(dsoundsink)));
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}
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static gboolean
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gst_directsoundsink_open (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink;
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HRESULT hRes;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* create and initialize a DirecSound object */
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if (FAILED (hRes = DirectSoundCreate (NULL, &dsoundsink->pDS, NULL))) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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("gst_directsoundsink_open: DirectSoundCreate: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS,
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GetDesktopWindow (), DSSCL_PRIORITY))) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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("gst_directsoundsink_open: IDirectSound_SetCooperativeLevel: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_directsoundsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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{
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GstDirectSoundSink *dsoundsink;
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HRESULT hRes;
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DSBUFFERDESC descSecondary;
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WAVEFORMATEX wfx;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/*save number of bytes per sample */
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dsoundsink->bytes_per_sample = spec->bytes_per_sample;
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/* fill the WAVEFORMATEX struture with spec params */
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memset (&wfx, 0, sizeof (wfx));
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wfx.cbSize = sizeof (wfx);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = spec->channels;
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wfx.nSamplesPerSec = spec->rate;
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wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels;
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wfx.nBlockAlign = spec->bytes_per_sample;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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GST_DEBUG
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("GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
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"WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n",
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spec->channels, spec->rate, spec->bytes_per_sample, wfx.nSamplesPerSec,
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wfx.wBitsPerSample, wfx.nBlockAlign, wfx.nAvgBytesPerSec);
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/* directsound buffer size can handle 2 secs of the stream */
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dsoundsink->buffer_size = wfx.nAvgBytesPerSec / 2;
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/* create a secondary directsound buffer */
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memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
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descSecondary.dwSize = sizeof (DSBUFFERDESC);
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descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 |
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DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME;
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descSecondary.dwBufferBytes = dsoundsink->buffer_size;
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descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx;
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hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
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&dsoundsink->pDSBSecondary, NULL);
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if (FAILED (hRes)) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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("gst_directsoundsink_prepare: IDirectSound_CreateSoundBuffer: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_directsoundsink_unprepare (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* release secondary DirectSound buffer */
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if (dsoundsink->pDSBSecondary)
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IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
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return TRUE;
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}
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static gboolean
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gst_directsoundsink_close (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink = NULL;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* release DirectSound object */
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g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE);
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IDirectSound_Release (dsoundsink->pDS);
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return TRUE;
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}
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static guint
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gst_directsoundsink_write (GstAudioSink * asink, gpointer data, guint length)
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{
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GstDirectSoundSink *dsoundsink;
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DWORD dwStatus;
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HRESULT hRes;
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LPVOID pLockedBuffer1 = NULL, pLockedBuffer2 = NULL;
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DWORD dwSizeBuffer1, dwSizeBuffer2;
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DWORD dwCurrentPlayCursor;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* get current buffer status */
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hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
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/* get current play cursor position */
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hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
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&dwCurrentPlayCursor, NULL);
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if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) {
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DWORD dwFreeBufferSize;
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calculate_freesize:
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/* calculate the free size of the circular buffer */
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if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
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dwFreeBufferSize =
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dsoundsink->buffer_size - (dsoundsink->current_circular_offset -
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dwCurrentPlayCursor);
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else
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dwFreeBufferSize =
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dwCurrentPlayCursor - dsoundsink->current_circular_offset;
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if (length >= dwFreeBufferSize) {
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Sleep (100);
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hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
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&dwCurrentPlayCursor, NULL);
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goto calculate_freesize;
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}
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}
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if (dwStatus & DSBSTATUS_BUFFERLOST) {
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hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary); /*need a loop waiting the buffer is restored?? */
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dsoundsink->current_circular_offset = 0;
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}
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hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
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dsoundsink->current_circular_offset, length, &pLockedBuffer1,
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&dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
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if (SUCCEEDED (hRes)) {
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// Write to pointers without reordering.
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|
memcpy (pLockedBuffer1, data, dwSizeBuffer1);
|
|
if (pLockedBuffer2 != NULL)
|
|
memcpy (pLockedBuffer2, (LPBYTE) data + dwSizeBuffer1, dwSizeBuffer2);
|
|
|
|
// Update where the buffer will lock (for next time)
|
|
dsoundsink->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
|
|
dsoundsink->current_circular_offset %= dsoundsink->buffer_size; /* Circular buffer */
|
|
|
|
hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1,
|
|
dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
|
|
}
|
|
|
|
/* if the buffer was not in playing state yet, call play on the buffer */
|
|
if (!(dwStatus & DSBSTATUS_PLAYING)) {
|
|
hRes = IDirectSoundBuffer_Play (dsoundsink->pDSBSecondary, 0, 0,
|
|
DSBPLAY_LOOPING);
|
|
}
|
|
|
|
return length;
|
|
}
|
|
|
|
static guint
|
|
gst_directsoundsink_delay (GstAudioSink * asink)
|
|
{
|
|
GstDirectSoundSink *dsoundsink;
|
|
HRESULT hRes;
|
|
DWORD dwCurrentPlayCursor;
|
|
DWORD dwBytesInQueue = 0;
|
|
gint nNbSamplesInQueue = 0;
|
|
|
|
dsoundsink = GST_DIRECTSOUND_SINK (asink);
|
|
|
|
/*evaluate the number of samples in queue in the circular buffer */
|
|
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
|
|
&dwCurrentPlayCursor, NULL);
|
|
|
|
if (hRes == S_OK) {
|
|
if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
|
|
dwBytesInQueue =
|
|
dsoundsink->current_circular_offset - dwCurrentPlayCursor;
|
|
else
|
|
dwBytesInQueue =
|
|
dsoundsink->current_circular_offset + (dsoundsink->buffer_size -
|
|
dwCurrentPlayCursor);
|
|
|
|
nNbSamplesInQueue = dwBytesInQueue / dsoundsink->bytes_per_sample;
|
|
}
|
|
|
|
return nNbSamplesInQueue;
|
|
}
|
|
|
|
static void
|
|
gst_directsoundsink_reset (GstAudioSink * asink)
|
|
{
|
|
/*not tested for seeking */
|
|
GstDirectSoundSink *dsoundsink;
|
|
|
|
dsoundsink = GST_DIRECTSOUND_SINK (asink);
|
|
|
|
IDirectSoundBuffer_Stop (dsoundsink->pDSBSecondary);
|
|
}
|