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configure.ac: Fix libs for linking directsound.
Original commit message from CVS: * configure.ac: Fix libs for linking directsound. * sys/directsound/gstdirectsoundsink.c: Fix buffer sizing to prevent racing the ringbuffer at startup. Add volume property.
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3 changed files with 105 additions and 41 deletions
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@ -1,3 +1,11 @@
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2008-10-01 Michael Smith <msmith@songbirdnest.com>
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* configure.ac:
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Fix libs for linking directsound.
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* sys/directsound/gstdirectsoundsink.c:
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Fix buffer sizing to prevent racing the ringbuffer at startup.
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Add volume property.
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2008-09-27 Jan Schmidt <jan.schmidt@sun.com>
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* ext/pulse/pulsesink.c:
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35
configure.ac
35
configure.ac
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@ -394,7 +394,7 @@ AG_GST_CHECK_FEATURE(DIRECTSOUND, [DirectSound plug-in], directsoundsink, [
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save_LIBS="$LIBS"
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CFLAGS="$CFLAGS $DIRECTSOUND_CFLAGS"
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LDFLAGS="$LDFLAGS $DIRECTSOUND_LDFLAGS"
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LIBS="$LIBS -ldsound -ldxerr9"
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LIBS="$LIBS -ldsound -ldxerr9 -luser32"
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AC_MSG_CHECKING(for DirectSound LDFLAGS)
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AC_LINK_IFELSE([
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#include <windows.h>
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@ -418,7 +418,7 @@ int main ()
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if test "x$HAVE_DIRECTSOUND" = "xyes"; then
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dnl this is much more than we want
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DIRECTSOUND_LIBS="-ldsound -ldxerr9"
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DIRECTSOUND_LIBS="-ldsound -ldxerr9 -luser32"
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AC_SUBST(DIRECTSOUND_CFLAGS)
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AC_SUBST(DIRECTSOUND_LDFLAGS)
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AC_SUBST(DIRECTSOUND_LIBS)
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@ -673,38 +673,9 @@ AG_GST_CHECK_FEATURE(ESD, [ESounD sound daemon], esdsink, [
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dnl *** FLAC ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_FLAC, true)
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AC_TRY_COMPILE([#include <FLAC/export.h>], [
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#if FLAC_API_VERSION_CURRENT < 8
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#error "legacy flac API"
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#endif
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], [ legacy_flac=no ], [ legacy_flac=yes ], [ legacy_flac=no ])
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if test "x$legacy_flac" = "xyes"; then
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AG_GST_CHECK_FEATURE(FLAC, [FLAC lossless audio], flac, [
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AG_GST_CHECK_LIBHEADER(FLAC, FLAC, FLAC__seekable_stream_encoder_new, -lm, FLAC/all.h, FLAC_LIBS="-lFLAC -lm")
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dnl API change in FLAC 1.1.1, so require that...
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dnl (this check will also fail with FLAC 1.1.3 which changed API again)
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if test x$HAVE_FLAC = xyes; then
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AC_CHECK_DECL(FLAC__SEEKABLE_STREAM_ENCODER_TELL_ERROR,
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HAVE_FLAC="yes", HAVE_FLAC="no", [
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#include <FLAC/seekable_stream_encoder.h>
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])
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fi
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AC_SUBST(FLAC_LIBS)
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AG_GST_PKG_CHECK_MODULES(FLAC, flac >= 1.1.3)
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])
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else
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AG_GST_CHECK_FEATURE(FLAC, [FLAC lossless audio], flac, [
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AG_GST_CHECK_LIBHEADER(FLAC, FLAC, FLAC__stream_encoder_new, -lm, FLAC/all.h, FLAC_LIBS="-lFLAC -lm")
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dnl API change in FLAC 1.1.3, so require that...
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if test x$HAVE_FLAC = xyes; then
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AC_CHECK_DECL(FLAC__STREAM_ENCODER_TELL_STATUS_ERROR,
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HAVE_FLAC="yes", HAVE_FLAC="no", [
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#include <FLAC/stream_encoder.h>
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])
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fi
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AC_SUBST(FLAC_LIBS)
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])
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fi
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dnl *** GConf ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_GCONF, true)
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@ -62,6 +62,8 @@
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#include "gstdirectsoundsink.h"
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#include <math.h>
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GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
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/* elementfactory information */
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@ -77,6 +79,11 @@ static void gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
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GstDirectSoundSinkClass * g_class);
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static void gst_directsound_sink_finalise (GObject * object);
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static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink);
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static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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@ -111,6 +118,12 @@ static GstStaticPadTemplate directsoundsink_sink_factory =
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
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enum
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{
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PROP_0,
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PROP_VOLUME
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};
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GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsound_sink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, gst_directsound_sink_interfaces_init);
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@ -167,6 +180,28 @@ gst_directsound_sink_mixer_list_tracks (GstMixer * mixer)
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return dsoundsink->tracks;
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}
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static void
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gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink)
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{
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if (dsoundsink->pDSBSecondary) {
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/* DirectSound controls volume using units of 100th of a decibel,
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* ranging from -10000 to 0. We use a linear scale of 0 - 100
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* here, so remap.
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*/
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long dsVolume;
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if (dsoundsink->volume == 0)
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dsVolume = -10000;
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else
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dsVolume = 100 * (long) (20 * log10 ((double) dsoundsink->volume / 100.));
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dsVolume = CLAMP (dsVolume, -10000, 0);
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GST_DEBUG_OBJECT (dsoundsink,
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"Setting volume on secondary buffer to %d from %d", (int) dsVolume,
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(int) dsoundsink->volume);
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IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
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}
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}
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/*
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* Set volume. volumes is an array of size track->num_channels, and
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* each value in the array gives the wanted volume for one channel
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@ -182,13 +217,7 @@ gst_directsound_sink_mixer_set_volume (GstMixer * mixer,
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if (volumes[0] != dsoundsink->volume) {
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dsoundsink->volume = volumes[0];
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if (dsoundsink->pDSBSecondary) {
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/* DirectSound is using attenuation in the following range
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* (DSBVOLUME_MIN=-10000, DSBVOLUME_MAX=0) */
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glong ds_attenuation = DSBVOLUME_MIN + (dsoundsink->volume * 100);
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IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, ds_attenuation);
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}
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gst_directsound_sink_set_volume (dsoundsink);
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}
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}
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@ -261,6 +290,10 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_sink_finalise);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_get_property);
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gstbasesink_class->get_caps =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
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@ -274,6 +307,12 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsound_sink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_sink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_sink_reset);
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g_object_class_install_property (gobject_class,
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PROP_VOLUME,
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g_param_spec_double ("volume", "Volume",
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"Volume of this stream", 0.0, 1.0, 1.0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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dsoundsink->first_buffer_after_reset = FALSE;
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}
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static void
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gst_directsound_sink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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case PROP_VOLUME:
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sink->volume = (int) (g_value_get_double (value) * 100);
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gst_directsound_sink_set_volume (sink);
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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_directsound_sink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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case PROP_VOLUME:
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g_value_set_double (value, (double) sink->volume / 100.);
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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_directsound_sink_getcaps (GstBaseSink * bsink)
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{
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@ -358,8 +430,19 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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wfx.nBlockAlign = spec->bytes_per_sample;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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/* directsound buffer size can handle 1/2 sec of the stream */
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dsoundsink->buffer_size = wfx.nAvgBytesPerSec / 2;
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/* Create directsound buffer with size based on our configured
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* buffer_size (which is 200 ms by default) */
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dsoundsink->buffer_size =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
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GST_MSECOND);
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spec->segsize =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
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GST_MSECOND);
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spec->segtotal = dsoundsink->buffer_size / spec->segsize;
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// Make the final buffer size be an integer number of segments
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dsoundsink->buffer_size = spec->segsize * spec->segtotal;
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GST_INFO_OBJECT (dsoundsink,
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"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
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@ -386,6 +469,8 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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return FALSE;
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}
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gst_directsound_sink_set_volume (dsoundsink);
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return TRUE;
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}
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