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.

diff --git a/ChangeLog b/ChangeLog
index f5cef90..ce9d992 100644 (file)
--- 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 2b3c4f1..785d40d 100644 (file)
--- 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 97127bd..bb9d38c 100644 (file)
--- 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;
 }