Merge branch 'master' of git+ssh://ligo-vcs.phys.uwm.edu/usr/local/git/gstlal

configure.ac

@@ -88,13 +88,21 @@ AC_SUBST([FFTW_LIBS])
 #
 
 
-PKG_CHECK_MODULES([NDS2], [nds2-client], [HAVE_NDS2="yes"], [AC_MSG_WARN([Not found!]) ; HAVE_NDS2="no"])
-AM_CONDITIONAL([COND_NDS2], [test "${HAVE_NDS2}" == "yes"])
-if test "${HAVE_NDS2}" == "yes" ; then
-	AC_DEFINE([HAVE_NDS2], 1)
+AC_ARG_WITH([nds],
+    [AS_HELP_STRING([--with-nds],
+        [include NDS source element @<:@default=check@:>@])],
+    [],
+    [with_nds=check])
+
+
+AS_IF([test "x$with_nds" != xno],
+    [PKG_CHECK_MODULES([NDS], [nds2-client], [HAVE_NDS="yes"], [AC_MSG_WARN([Not found!]) ; HAVE_NDS="no"])], [])
+AM_CONDITIONAL([COND_NDS], [test "${HAVE_NDS}" == "yes"])
+if test "${HAVE_NDS}" == "yes" ; then
+	AC_DEFINE([HAVE_NDS], 1)
 fi
-AC_SUBST([NDS2_CFLAGS])
-AC_SUBST([NDS2_LIBS])
+AC_SUBST([NDS_CFLAGS])
+AC_SUBST([NDS_LIBS])
 
 
 #

examples/play_nds_hoft.sh

 # A gst-launch pipeline to encode h(t) from NDS2 into a OGG/Vorbis file
+# The 'adder' element is included as a hack to throw away GPS time stamps; otherwise
+# the autoaudiosink won't play
 
 HOST="marble.ligo-wa.caltech.edu"
 REQUESTED_CHANNEL_NAME="H1:DMT-STRAIN"
@@ -6,7 +8,7 @@ REQUESTED_CHANNEL_NAME="H1:DMT-STRAIN"
 gst-launch \
 	ndssrc \
         host="${HOST}" \
-        requested-channel-name="${REQUESTED_CHANNEL_NAME}" \
+        channel-name="${REQUESTED_CHANNEL_NAME}" \
 	! audiochebband \
 		lower-frequency=45 \
 		upper-frequency=2500 \
@@ -16,6 +18,7 @@ gst-launch \
 		amplification=2e+17 \
 	! progressreport \
 		update-freq=2 \
-    ! audiorate \
+	! adder \
+	! audiorate \
 	! audioconvert \
 	! autoaudiosink

src/plugins/Makefile.am

@@ -2,7 +2,7 @@ INCLUDES = -I$(top_srcdir)/src/lib
 
 plugin_LTLIBRARIES = libgstlal.la
 
-if COND_NDS2
+if COND_NDS
 NDSSRC_SOURCES = gstlal_ndssrc.h gstlal_ndssrc.c
 else
 NDSSRC_SOURCES =
@@ -29,8 +29,8 @@ libgstlal_la_SOURCES = \
 	gstlal_reblock.h gstlal_reblock.c \
 	gstlal_delay.h gstlal_delay.c \
 	$(NDSSRC_SOURCES)
-libgstlal_la_CFLAGS = -O2 $(AM_CFLAGS) $(CFLAGS) ${gstreamer_CFLAGS} ${LAL_CFLAGS} ${GSL_CFLAGS} ${FFTW_CFLAGS} ${NDS2_CFLAGS}
-libgstlal_la_LIBADD = ${gstreamer_LIBS} ${LAL_LIBS} ${GSL_LIBS} ${FFTW_LIBS} ${NDS2_LIBS} $(top_srcdir)/src/lib/libgstlal.la
+libgstlal_la_CFLAGS = -O2 $(AM_CFLAGS) $(CFLAGS) ${gstreamer_CFLAGS} ${LAL_CFLAGS} ${GSL_CFLAGS} ${FFTW_CFLAGS} ${NDS_CFLAGS}
+libgstlal_la_LIBADD = ${gstreamer_LIBS} ${LAL_LIBS} ${GSL_LIBS} ${FFTW_LIBS} ${NDS_LIBS} $(top_srcdir)/src/lib/libgstlal.la
 libgstlal_la_LDFLAGS = ${GSTREAMER_PLUGIN_LDFLAGS}
 
 noinst_HEADERS = gstlal_plugins.h

src/plugins/gstlal.c

@@ -91,7 +91,7 @@
 #include <gstlal_nofakedisconts.h>
 #include <gstlal_reblock.h>
 #include <gstlal_delay.h>
-#ifdef HAVE_NDS2
+#ifdef HAVE_NDS
 #include <gstlal_ndssrc.h>
 #endif
 
@@ -140,7 +140,7 @@ static gboolean plugin_init(GstPlugin *plugin)
 		{"lal_nofakedisconts", GSTLAL_NOFAKEDISCONTS_TYPE},
 		{"lal_reblock", GSTLAL_REBLOCK_TYPE},
 		{"lal_delay", GSTLAL_DELAY_TYPE},
-#ifdef HAVE_NDS2
+#ifdef HAVE_NDS
 		{"ndssrc", GSTLAL_NDSSRC_TYPE},
 #endif
 		{NULL, 0},

src/plugins/gstlal_delay.c

@@ -55,7 +55,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE(
 	GST_STATIC_CAPS(
 		"audio/x-raw-float, " \
 		"rate = (int) [1, MAX], " \
-		"channels = (int) [1, MAX], " \
+		"channels = (int) [1, MAX], "\
 		"endianness = (int) BYTE_ORDER, " \
 		"width = (int) {32,64}; " \
 		"audio/x-raw-int, " \
@@ -106,13 +106,6 @@ GST_BOILERPLATE(
 );
 
 
-enum property {
-   ARG_DELAY = 1,
-   ARG_SILENT,
-};
-#define DEFAULT_DELAY 0
-
-
 /*
  * ============================================================================
  *
@@ -122,8 +115,18 @@ enum property {
  */
 
 
+enum property {
+   ARG_DELAY = 1,
+   ARG_SILENT,
+};
+
+#define DEFAULT_DELAY 0
+
 /*
- * get_unit_size()
+ * get_unit_size() stores the size (in bytes) of a single sample
+ * from a single channel in the buffer.
+ * The "width" of a buffer is equal to the total number of channels
+ * times the number of bits per channel.
  */
 static gboolean get_unit_size(GstBaseTransform *trans, GstCaps *caps, guint *size)
 {
@@ -146,7 +149,11 @@ static gboolean get_unit_size(GstBaseTransform *trans, GstCaps *caps, guint *siz
 }
 
 
-
+/*
+ * When the caps on an element's pads a finally set, this function is called.
+ * We use this opportunity to record the chosen sampling rate and the unit
+ * size.
+ */
 static gboolean set_caps(GstBaseTransform *trans,
 			 GstCaps *incaps,
 			 GstCaps *outcaps)
@@ -159,19 +166,45 @@ static gboolean set_caps(GstBaseTransform *trans,
       /* size of unit sample */
       get_unit_size(trans,incaps,&element->unit_size);
 
+      if ( !element->silent)
+      {
+	    fprintf(stderr,"lal_delay incaps set = %s\n",gst_caps_to_string(incaps));
+	    fprintf(stderr,"lal_delay outcaps set = %s\n",gst_caps_to_string(outcaps));
+      }
+
+
       return TRUE;
 }
 
 
+/*
+ * In addition to handling buffers, elements also handle events.
+ * These events change how the element behaves, e.g., an EOS event
+ * tells the element to stop reading buffers.  The event() function
+ * is called whenever an event is passed to an element.  Return
+ * FALSE to override the parent class event handling.
+ */
 static gboolean event(GstBaseTransform *trans, GstEvent *event)
 {
+
+      GSTLALDelay *element = GSTLAL_DELAY( trans );
+
+      if ( !element->silent )
+      {
+	    fprintf(stderr,"lal_delay received signal %s\n",
+		    gst_event_type_get_name(GST_EVENT_TYPE(event)));
+      }
+
       return TRUE; //FIXME
 }
 
 
 
 /*
- * set up output buffer
+ * When an input buffer is received, prepare_output_buffer is called.
+ * This function allows you to map an output buffer to a given
+ * input buffer.  In this case, we use this function to set the
+ * size of the output buffer.
  */
 static GstFlowReturn prepare_output_buffer(GstBaseTransform *trans,
 					   GstBuffer *inbuf,
@@ -212,6 +245,12 @@ static GstFlowReturn prepare_output_buffer(GstBaseTransform *trans,
 }
 
 
+/*
+ * The transform function actually does the heavy lifting on buffers.
+ * Given an input buffer and an output buffer (the latter of which is
+ * set in prepare_output_buffer), determine what data actually gets put
+ * into the output buffer.
+ */
 static GstFlowReturn transform( GstBaseTransform *trans, GstBuffer *inbuf, GstBuffer *outbuf)
 {
       GSTLALDelay *element = GSTLAL_DELAY(trans);
@@ -219,12 +258,12 @@ static GstFlowReturn transform( GstBaseTransform *trans, GstBuffer *inbuf, GstBu
       guint delaysize = element->unit_size*element->delay;
       guint64 delaytime;
 
-      if ( GST_BUFFER_SIZE(inbuf) < delaysize )
+      if ( GST_BUFFER_SIZE(inbuf) <= delaysize )
 	 /* drop entire buffer */
       {
 	    if ( !element->silent )
 	    {
-		  fprintf(stderr,"skip %ld to %ld, pass nothing (rate %d)\n",
+		  fprintf(stderr,"skip %ld to %ld, let nothing through (rate %d)\n",
 			  GST_BUFFER_TIMESTAMP(inbuf),
 			  GST_BUFFER_TIMESTAMP(inbuf)+GST_BUFFER_DURATION(inbuf),
 			  element->rate);
@@ -251,7 +290,7 @@ static GstFlowReturn transform( GstBaseTransform *trans, GstBuffer *inbuf, GstBu
 
 	    if ( !element->silent )
 	    {
-		  fprintf(stderr,"skip %ld to %ld, pass %ld to %ld (rate %d)\n",
+		  fprintf(stderr,"skip %ld to %ld, let %ld to %ld through (rate %d)\n",
 			  GST_BUFFER_TIMESTAMP(inbuf),
 			  GST_BUFFER_TIMESTAMP(inbuf) + delaytime,
 			  GST_BUFFER_TIMESTAMP(outbuf),
@@ -267,6 +306,13 @@ static GstFlowReturn transform( GstBaseTransform *trans, GstBuffer *inbuf, GstBu
       else
 	 /* pass entire buffer */
       {
+	    if ( !element->silent )
+	    {
+		  fprintf(stderr,"let everything through %ld to %ld (rate %d)\n",
+			  GST_BUFFER_TIMESTAMP(inbuf),
+			  GST_BUFFER_TIMESTAMP(inbuf) + GST_BUFFER_DURATION(inbuf),
+			  element->rate);
+	    }
 	    result = GST_FLOW_OK;
       }
 

src/plugins/gstlal_ndssrc.h

 /*
  *  NDS-based src element
+ *  see https://www.lsc-group.phys.uwm.edu/daswg/wiki/NetworkDataServer2
  *
  *  Copyright (C) 2008  Leo Singer
  *
@@ -54,10 +55,15 @@ typedef struct {
 
     char* host;
     int port;
-	char* requested_channel_name;
-    
+
     daq_t* daq;
-    daq_channel_t* daq_channel;
+    gboolean needs_seek;
+
+    char* channelName;
+    enum chantype channelType;
+    daq_channel_t* availableChannels;
+    int countAvailableChannels;
+
 } GSTLALNDSSrc;