summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
-rw-r--r--ChangeLog8
-rw-r--r--configure.ac35
-rw-r--r--sys/directsound/gstdirectsoundsink.c103
3 files changed, 105 insertions, 41 deletions
diff --git a/ChangeLog b/ChangeLog
index f5cef900..ce9d9920 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -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:
diff --git a/configure.ac b/configure.ac
index 2b3c4f18..785d40de 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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)
diff --git a/sys/directsound/gstdirectsoundsink.c b/sys/directsound/gstdirectsoundsink.c
index 97127bdb..bb9d38c6 100644
--- a/sys/directsound/gstdirectsoundsink.c
+++ b/sys/directsound/gstdirectsoundsink.c
@@ -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;
}