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.
This commit is contained in:
Michael Smith 2008-10-01 21:22:26 +00:00
parent 44cb25a342
commit e2dbf108e6
3 changed files with 105 additions and 41 deletions

View file

@ -1,3 +1,11 @@
2008-10-01 Michael Smith <msmith@songbirdnest.com>
* configure.ac:
Fix libs for linking directsound.
* sys/directsound/gstdirectsoundsink.c:
Fix buffer sizing to prevent racing the ringbuffer at startup.
Add volume property.
2008-09-27 Jan Schmidt <jan.schmidt@sun.com>
* ext/pulse/pulsesink.c:

View file

@ -394,7 +394,7 @@ AG_GST_CHECK_FEATURE(DIRECTSOUND, [DirectSound plug-in], directsoundsink, [
save_LIBS="$LIBS"
CFLAGS="$CFLAGS $DIRECTSOUND_CFLAGS"
LDFLAGS="$LDFLAGS $DIRECTSOUND_LDFLAGS"
LIBS="$LIBS -ldsound -ldxerr9"
LIBS="$LIBS -ldsound -ldxerr9 -luser32"
AC_MSG_CHECKING(for DirectSound LDFLAGS)
AC_LINK_IFELSE([
#include <windows.h>
@ -418,7 +418,7 @@ int main ()
if test "x$HAVE_DIRECTSOUND" = "xyes"; then
dnl this is much more than we want
DIRECTSOUND_LIBS="-ldsound -ldxerr9"
DIRECTSOUND_LIBS="-ldsound -ldxerr9 -luser32"
AC_SUBST(DIRECTSOUND_CFLAGS)
AC_SUBST(DIRECTSOUND_LDFLAGS)
AC_SUBST(DIRECTSOUND_LIBS)
@ -673,38 +673,9 @@ AG_GST_CHECK_FEATURE(ESD, [ESounD sound daemon], esdsink, [
dnl *** FLAC ***
translit(dnm, m, l) AM_CONDITIONAL(USE_FLAC, true)
AC_TRY_COMPILE([#include <FLAC/export.h>], [
#if FLAC_API_VERSION_CURRENT < 8
#error "legacy flac API"
#endif
], [ legacy_flac=no ], [ legacy_flac=yes ], [ legacy_flac=no ])
if test "x$legacy_flac" = "xyes"; then
AG_GST_CHECK_FEATURE(FLAC, [FLAC lossless audio], flac, [
AG_GST_CHECK_LIBHEADER(FLAC, FLAC, FLAC__seekable_stream_encoder_new, -lm, FLAC/all.h, FLAC_LIBS="-lFLAC -lm")
dnl API change in FLAC 1.1.1, so require that...
dnl (this check will also fail with FLAC 1.1.3 which changed API again)
if test x$HAVE_FLAC = xyes; then
AC_CHECK_DECL(FLAC__SEEKABLE_STREAM_ENCODER_TELL_ERROR,
HAVE_FLAC="yes", HAVE_FLAC="no", [
#include <FLAC/seekable_stream_encoder.h>
])
fi
AC_SUBST(FLAC_LIBS)
AG_GST_PKG_CHECK_MODULES(FLAC, flac >= 1.1.3)
])
else
AG_GST_CHECK_FEATURE(FLAC, [FLAC lossless audio], flac, [
AG_GST_CHECK_LIBHEADER(FLAC, FLAC, FLAC__stream_encoder_new, -lm, FLAC/all.h, FLAC_LIBS="-lFLAC -lm")
dnl API change in FLAC 1.1.3, so require that...
if test x$HAVE_FLAC = xyes; then
AC_CHECK_DECL(FLAC__STREAM_ENCODER_TELL_STATUS_ERROR,
HAVE_FLAC="yes", HAVE_FLAC="no", [
#include <FLAC/stream_encoder.h>
])
fi
AC_SUBST(FLAC_LIBS)
])
fi
dnl *** GConf ***
translit(dnm, m, l) AM_CONDITIONAL(USE_GCONF, true)

View file

@ -62,6 +62,8 @@
#include "gstdirectsoundsink.h"
#include <math.h>
GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
/* elementfactory information */
@ -77,6 +79,11 @@ static void gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
GstDirectSoundSinkClass * g_class);
static void gst_directsound_sink_finalise (GObject * object);
static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
@ -111,6 +118,12 @@ static GstStaticPadTemplate directsoundsink_sink_factory =
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
enum
{
PROP_0,
PROP_VOLUME
};
GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsound_sink, GstAudioSink,
GST_TYPE_AUDIO_SINK, gst_directsound_sink_interfaces_init);
@ -167,6 +180,28 @@ gst_directsound_sink_mixer_list_tracks (GstMixer * mixer)
return dsoundsink->tracks;
}
static void
gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink)
{
if (dsoundsink->pDSBSecondary) {
/* DirectSound controls volume using units of 100th of a decibel,
* ranging from -10000 to 0. We use a linear scale of 0 - 100
* here, so remap.
*/
long dsVolume;
if (dsoundsink->volume == 0)
dsVolume = -10000;
else
dsVolume = 100 * (long) (20 * log10 ((double) dsoundsink->volume / 100.));
dsVolume = CLAMP (dsVolume, -10000, 0);
GST_DEBUG_OBJECT (dsoundsink,
"Setting volume on secondary buffer to %d from %d", (int) dsVolume,
(int) dsoundsink->volume);
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
}
}
/*
* Set volume. volumes is an array of size track->num_channels, and
* each value in the array gives the wanted volume for one channel
@ -182,13 +217,7 @@ gst_directsound_sink_mixer_set_volume (GstMixer * mixer,
if (volumes[0] != dsoundsink->volume) {
dsoundsink->volume = volumes[0];
if (dsoundsink->pDSBSecondary) {
/* DirectSound is using attenuation in the following range
* (DSBVOLUME_MIN=-10000, DSBVOLUME_MAX=0) */
glong ds_attenuation = DSBVOLUME_MIN + (dsoundsink->volume * 100);
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, ds_attenuation);
}
gst_directsound_sink_set_volume (dsoundsink);
}
}
@ -261,6 +290,10 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_sink_finalise);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_directsound_sink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_directsound_sink_get_property);
gstbasesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
@ -274,6 +307,12 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsound_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_sink_reset);
g_object_class_install_property (gobject_class,
PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Volume of this stream", 0.0, 1.0, 1.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
@ -300,6 +339,39 @@ gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
dsoundsink->first_buffer_after_reset = FALSE;
}
static void
gst_directsound_sink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
sink->volume = (int) (g_value_get_double (value) * 100);
gst_directsound_sink_set_volume (sink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_sink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
g_value_set_double (value, (double) sink->volume / 100.);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_directsound_sink_getcaps (GstBaseSink * bsink)
{
@ -358,8 +430,19 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
wfx.nBlockAlign = spec->bytes_per_sample;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* directsound buffer size can handle 1/2 sec of the stream */
dsoundsink->buffer_size = wfx.nAvgBytesPerSec / 2;
/* Create directsound buffer with size based on our configured
* buffer_size (which is 200 ms by default) */
dsoundsink->buffer_size =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
GST_MSECOND);
spec->segsize =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
GST_MSECOND);
spec->segtotal = dsoundsink->buffer_size / spec->segsize;
// Make the final buffer size be an integer number of segments
dsoundsink->buffer_size = spec->segsize * spec->segtotal;
GST_INFO_OBJECT (dsoundsink,
"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
@ -386,6 +469,8 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
return FALSE;
}
gst_directsound_sink_set_volume (dsoundsink);
return TRUE;
}