plugin: port audio filter example to 1.0

https://bugzilla.gnome.org/show_bug.cgi?id=764409

3 files changed, 165 insertions, 106 deletions

diff --git a/gst-plugin/configure.ac b/gst-plugin/configure.ac
index 1908b28..7f69f0a 100644
--- a/gst-plugin/configure.ac
+++ b/gst-plugin/configure.ac
@@ -46,6 +46,7 @@ PKG_CHECK_MODULES(GST, [
   gstreamer-1.0 >= $GST_REQUIRED
   gstreamer-base-1.0 >= $GST_REQUIRED
   gstreamer-controller-1.0 >= $GST_REQUIRED
+  gstreamer-audio-1.0 >= $GST_REQUIRED
 ], [
   AC_SUBST(GST_CFLAGS)
   AC_SUBST(GST_LIBS)
diff --git a/gst-plugin/src/Makefile.am b/gst-plugin/src/Makefile.am
index 8f2bc4b..9f778e5 100644
--- a/gst-plugin/src/Makefile.am
+++ b/gst-plugin/src/Makefile.am
@@ -3,7 +3,7 @@
 ##############################################################################
 # TODO: change libgstplugin.la to something else, e.g. libmysomething.la     #
 ##############################################################################
-plugin_LTLIBRARIES = libgstplugin.la
+plugin_LTLIBRARIES = libgstplugin.la libgstaudiofilterexample.la
 
 ##############################################################################
 # TODO: for the next set of variables, name the prefix if you named the .la, #
@@ -13,6 +13,8 @@ plugin_LTLIBRARIES = libgstplugin.la
 #                            libmysomething_la_LDFLAGS                       #
 ##############################################################################
 
+## Plugin 1
+
 # sources used to compile this plug-in
 libgstplugin_la_SOURCES = gstplugin.c gstplugin.h
 
@@ -22,5 +24,16 @@ libgstplugin_la_LIBADD = $(GST_LIBS)
 libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static
 
+## Plugin 2 (audio filter example)
+
+# sources used to compile this plug-in
+libgstaudiofilterexample_la_SOURCES = gstaudiofilter.c
+
+# compiler and linker flags used to compile this plugin, set in configure.ac
+libgstaudiofilterexample_la_CFLAGS = $(GST_CFLAGS)
+libgstaudiofilterexample_la_LIBADD = $(GST_LIBS)
+libgstaudiofilterexample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstaudiofilterexample_la_LIBTOOLFLAGS = --tag=disable-static
+
 # headers we need but don't want installed
 noinst_HEADERS = gstplugin.h
diff --git a/gst-plugin/src/gstaudiofilter.c b/gst-plugin/src/gstaudiofilter.c
index 462b763..4102a44 100644
--- a/gst-plugin/src/gstaudiofilter.c
+++ b/gst-plugin/src/gstaudiofilter.c
@@ -1,5 +1,4 @@
-/*
- * GStreamer
+/* GStreamer audio filter example class
  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
  * Copyright (C) <2003> David Schleef <ds@schleef.org>
  * Copyright (C) YEAR AUTHOR_NAME AUTHOR_EMAIL
@@ -65,31 +64,35 @@
 #include <gst/audio/gstaudiofilter.h>
 #include <string.h>
 
-GST_DEBUG_CATEGORY_STATIC (gst_plugin_template_debug);
-#define GST_CAT_DEFAULT gst_plugin_template_debug
-
-typedef struct _GstPluginTemplate GstPluginTemplate;
-typedef struct _GstPluginTemplateClass GstPluginTemplateClass;
-
-#define GST_TYPE_PLUGIN_TEMPLATE \
-  (gst_plugin_template_get_type())
-#define GST_PLUGIN_TEMPLATE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PLUGIN_TEMPLATE,GstPluginTemplate))
-#define GST_PLUGIN_TEMPLATE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PLUGIN_TEMPLATE,GstPluginTemplateClass))
-#define GST_IS_PLUGIN_TEMPLATE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PLUGIN_TEMPLATE))
-#define GST_IS_PLUGIN_TEMPLATE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PLUGIN_TEMPLATE))
-
-struct _GstPluginTemplate
+GST_DEBUG_CATEGORY_STATIC (audiofiltertemplate_debug);
+#define GST_CAT_DEFAULT audiofiltertemplate_debug
+
+typedef struct _GstAudioFilterTemplate GstAudioFilterTemplate;
+typedef struct _GstAudioFilterTemplateClass GstAudioFilterTemplateClass;
+
+/* These are boilerplate cast macros and type check macros */
+#define GST_TYPE_AUDIO_FILTER_TEMPLATE \
+  (gst_audio_filter_template_get_type())
+#define GST_AUDIO_FILTER_TEMPLATE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FILTER_TEMPLATE,GstAudioFilterTemplate))
+#define GST_AUDIO_FILTER_TEMPLATE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FILTER_TEMPLATE,GstAudioFilterTemplateClass))
+#define GST_IS_AUDIO_FILTER_TEMPLATE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FILTER_TEMPLATE))
+#define GST_IS_AUDIO_FILTER_TEMPLATE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FILTER_TEMPLATE))
+
+struct _GstAudioFilterTemplate
 {
   GstAudioFilter audiofilter;
+
+  /* here you can add additional per-instance
+   * data such as properties */
 };
 
-struct _GstPluginTemplateClass
+struct _GstAudioFilterTemplateClass
 {
-  GstAudioFilterClass parent_class;
+  GstAudioFilterClass audiofilter_class;
 };
 
 
@@ -105,60 +108,45 @@ enum
   /* FILL ME */
 };
 
-/* debug category for fltering log messages
- *
- * FIXME:exchange the string 'Template plugin' with your description
- */
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_plugin_template_debug, "plugin", 0, "Template plugin");
-
-GST_BOILERPLATE_FULL (GstPluginTemplate, gst_plugin_template,
-    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioFilterTemplate, gst_audio_filter_template,
+    GST_TYPE_AUDIO_FILTER);
 
-static void gst_plugin_template_set_property (GObject * object,
+static void gst_audio_filter_template_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_plugin_template_get_property (GObject * object,
+static void gst_audio_filter_template_get_property (GObject * object,
     guint prop_id, GValue * value, GParamSpec * pspec);
 
-static gboolean gst_plugin_template_setup (GstAudioFilter * filter,
-    GstRingBufferSpec * spec);
-static GstFlowReturn gst_plugin_template_filter (GstBaseTransform * bt,
+static gboolean gst_audio_filter_template_setup (GstAudioFilter * filter,
+    const GstAudioInfo * info);
+static GstFlowReturn gst_audio_filter_template_filter (GstBaseTransform * bt,
     GstBuffer * outbuf, GstBuffer * inbuf);
 static GstFlowReturn
-gst_plugin_template_filter_inplace (GstBaseTransform * base_transform,
+gst_audio_filter_template_filter_inplace (GstBaseTransform * base_transform,
     GstBuffer * buf);
 
-#define ALLOWED_CAPS_STRING \
-    GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS
-
-/* GObject vmethod implementations */
-
-static void
-gst_plugin_template_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
-  GstCaps *caps;
-  
-  gst_element_class_set_details_simple (element_class,
-    "Plugin",
-    "Filter/Effect/Audio",
-    "FIXME:Filters audio",
-    "AUTHOR_NAME AUTHOR_EMAIL");
+#if 0
+/* This means we support signed 16-bit pcm and signed 32-bit pcm in native
+ * endianness */
+#define SUPPORTED_CAPS_STRING \
+    GST_AUDIO_CAPS_MAKE("{ " GST_AUDIO_NE(S16) ", " GST_AUDIO_NE(S32) " }")
+#endif
 
-  caps = gst_caps_from_string (ALLOWED_CAPS_STRING);
-  gst_audio_filter_class_add_pad_templates (audiofilter_class, caps);
-  gst_caps_unref (caps);
-}
+/* For simplicity only support 16-bit pcm in native endianness for starters */
+#define SUPPORTED_CAPS_STRING \
+    GST_AUDIO_CAPS_MAKE(GST_AUDIO_NE(S16))
 
+/* GObject vmethod implementations */
 static void
-gst_plugin_template_class_init (GstPluginTemplateClass * klass)
+gst_audio_filter_template_class_init (GstAudioFilterTemplateClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *element_class;
   GstBaseTransformClass *btrans_class;
   GstAudioFilterClass *audio_filter_class;
+  GstCaps *caps;
 
   gobject_class = (GObjectClass *) klass;
+  element_class = (GstElementClass *) klass;
   btrans_class = (GstBaseTransformClass *) klass;
   audio_filter_class = (GstAudioFilterClass *) klass;
 
@@ -169,32 +157,42 @@ gst_plugin_template_class_init (GstPluginTemplateClass * klass)
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 #endif
 
-  gobject_class->set_property = gst_plugin_template_set_property;
-  gobject_class->get_property = gst_plugin_template_get_property;
+  gobject_class->set_property = gst_audio_filter_template_set_property;
+  gobject_class->get_property = gst_audio_filter_template_get_property;
 
-  /* this function will be called whenever the format changes */
-  audio_filter_class->setup = gst_plugin_template_setup;
+  /* this function will be called when the format is set before the
+   * first buffer comes in, and whenever the format changes */
+  audio_filter_class->setup = gst_audio_filter_template_setup;
 
   /* here you set up functions to process data (either in place, or from
    * one input buffer to another output buffer); only one is required */
-  btrans_class->transform = gst_plugin_template_filter;
-  btrans_class->transform_ip = gst_plugin_template_filter_inplace;
+  btrans_class->transform = gst_audio_filter_template_filter;
+  btrans_class->transform_ip = gst_audio_filter_template_filter_inplace;
+  /* Set some basic metadata about your new element */
+  gst_element_class_set_details_simple (element_class,
+    "Audio Filter Template", /* FIXME: short name */
+    "Filter/Effect/Audio",
+    "Filters audio", /* FIXME: short description*/
+    "Name <mail@example.com>"); /* FIXME: author */
+
+  caps = gst_caps_from_string (SUPPORTED_CAPS_STRING);
+  gst_audio_filter_class_add_pad_templates (audio_filter_class, caps);
+  gst_caps_unref (caps);
 }
 
 static void
-gst_plugin_template_init (GstPluginTemplate * plugin_template,
-    GstPluginTemplateClass * g_class)
+gst_audio_filter_template_init (GstAudioFilterTemplate * filter)
 {
-  GST_DEBUG ("init");
-
-  /* do stuff if you need to */
+  /* This function is called when a new filter object is created. You
+   * would typically do things like initialise properties to their
+   * default values here if needed. */
 }
 
 static void
-gst_plugin_template_set_property (GObject * object, guint prop_id,
+gst_audio_filter_template_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
-  GstPluginTemplate *filter = GST_PLUGIN_TEMPLATE (object);
+  GstAudioFilterTemplate *filter = GST_AUDIO_FILTER_TEMPLATE (object);
 
   GST_OBJECT_LOCK (filter);
   switch (prop_id) {
@@ -206,10 +204,10 @@ gst_plugin_template_set_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_plugin_template_get_property (GObject * object, guint prop_id,
+gst_audio_filter_template_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
-  GstPluginTemplate *filter = GST_PLUGIN_TEMPLATE (object);
+  GstAudioFilterTemplate *filter = GST_AUDIO_FILTER_TEMPLATE (object);
 
   GST_OBJECT_LOCK (filter);
   switch (prop_id) {
@@ -221,16 +219,29 @@ gst_plugin_template_get_property (GObject * object, guint prop_id,
 }
 
 static gboolean
-gst_plugin_template_setup (GstAudioFilter * filter,
-    GstRingBufferSpec * spec)
+gst_audio_filter_template_setup (GstAudioFilter * filter,
+    const GstAudioInfo * info)
 {
-  GstPluginTemplate *plugin_template;
+  GstAudioFilterTemplate *filter_template;
+  GstAudioFormat fmt;
+  gint chans, rate;
+
+  filter_template = GST_AUDIO_FILTER_TEMPLATE (filter);
+
+  rate = GST_AUDIO_INFO_RATE (info);
+  chans = GST_AUDIO_INFO_CHANNELS (info);
+  fmt = GST_AUDIO_INFO_FORMAT (info);
 
-  plugin_template = GST_PLUGIN_TEMPLATE (filter);
+  GST_INFO_OBJECT (filter_template, "format %d (%s), rate %d, %d channels",
+      fmt, GST_AUDIO_INFO_NAME (info), rate, chans);
 
-  /* if any setup needs to be done, do it here */
+  /* if any setup needs to be done (like memory allocated), do it here */
 
-  return TRUE;                  /* it's all good */
+  /* The audio filter base class also saves the audio info in
+   * GST_AUDIO_FILTER_INFO(filter) so it's automatically available
+   * later from there as well */
+
+  return TRUE;
 }
 
 /* You may choose to implement either a copying filter or an
@@ -240,45 +251,79 @@ gst_plugin_template_setup (GstAudioFilter * filter,
  * with a minimum of memory copies. */
 
 static GstFlowReturn
-gst_plugin_template_filter (GstBaseTransform * base_transform,
+gst_audio_filter_template_filter (GstBaseTransform * base_transform,
     GstBuffer * inbuf, GstBuffer * outbuf)
 {
-  GstPluginTemplate *plugin_template;
-  GstAudioFilter *audiofilter;
-
-  audiofilter = GST_AUDIO_FILTER (base_transform);
-  plugin_template = GST_PLUGIN_TEMPLATE (base_transform);
-
-  /* FIXME: do something interesting here.  This simply copies the source
-   * to the destination. */
-
-  memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf),
-      GST_BUFFER_SIZE (inbuf));
+  GstAudioFilterTemplate *filter = GST_AUDIO_FILTER_TEMPLATE (base_transform);
+  GstMapInfo map_in;
+  GstMapInfo map_out;
+
+  GST_LOG_OBJECT (filter, "transform buffer");
+
+  /* FIXME: do something interesting here.  We simply copy the input data
+   * to the output buffer for now. */
+  if (gst_buffer_map (inbuf, &map_in, GST_MAP_READ)) {
+    if (gst_buffer_map (outbuf, &map_out, GST_MAP_WRITE)) {
+      g_assert (map_out.size == map_in.size);
+      memcpy (map_out.data, map_in.data, map_out.size);
+      gst_buffer_unmap (outbuf, &map_out);
+    }
+    gst_buffer_unmap (inbuf, &map_in);
+  }
 
   return GST_FLOW_OK;
 }
 
 static GstFlowReturn
-gst_plugin_template_filter_inplace (GstBaseTransform * base_transform,
+gst_audio_filter_template_filter_inplace (GstBaseTransform * base_transform,
     GstBuffer * buf)
 {
-  GstPluginTemplate *plugin_template;
-  GstAudioFilter *audiofilter;
+  GstAudioFilterTemplate *filter = GST_AUDIO_FILTER_TEMPLATE (base_transform);
+  GstFlowReturn flow = GST_FLOW_OK;
+  GstMapInfo map;
 
-  audiofilter = GST_AUDIO_FILTER (base_transform);
-  plugin_template = GST_PLUGIN_TEMPLATE (base_transform);
+  GST_LOG_OBJECT (filter, "transform buffer in place");
 
-  /* FIXME: do something interesting here.  This simply copies the source
-   * to the destination. */
+  /* FIXME: do something interesting here.  Doing nothing means the input
+   * buffer is simply pushed out as is without any modification */
+  if (gst_buffer_map (buf, &map, GST_MAP_READWRITE)) {
+#if 0
+    switch (GST_AUDIO_FILTER_FORMAT (filter)) {
+      case GST_AUDIO_FORMAT_S16LE:
+      case GST_AUDIO_FORMAT_S16BE: {
+        gint16 *samples = map.data;
+        guint n_samples = map.size / sizeof (gint16);
+        guint i;
+
+        for (i = 0; i < n; ++n) {
+          samples[i] = samples[i];
+        }
+        break;
+      }
+      default:
+        g_warning ("Unexpected audio format %s!",
+            GST_AUDIO_INFO_NAME (GST_AUDIO_FILTER_INFO(filter)));
+        flow = GST_FLOW_ERROR;
+        break;
+    }
+#endif
+    gst_buffer_unmap (buf, &map);
+  }
 
-  return GST_FLOW_OK;
+  return flow;
 }
 
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  return gst_element_register (plugin, "plugin", GST_RANK_NONE,
-      GST_TYPE_PLUGIN_TEMPLATE);
+  /* Register debug category for filtering log messages
+   * FIXME:exchange the string 'Template plugin' with your description */
+  GST_DEBUG_CATEGORY_INIT (audiofiltertemplate_debug, "audiofiltertemplate", 0,
+      "Audio filter template example");
+
+  /* This is the name used in gst-launch-1.0 and gst_element_factory_make() */
+  return gst_element_register (plugin, "audiofiltertemplate", GST_RANK_NONE,
+      GST_TYPE_AUDIO_FILTER_TEMPLATE);
 }
 
 /* gstreamer looks for this structure to register plugins
@@ -288,10 +333,10 @@ plugin_init (GstPlugin * plugin)
 GST_PLUGIN_DEFINE (
     GST_VERSION_MAJOR,
     GST_VERSION_MINOR,
-    "plugin",
-    "Template plugin",
+    audiofiltertemplateplugin,
+    "Audio filter template plugin",
     plugin_init,
     VERSION, "LGPL",
     "GStreamer",
-    "http://gstreamer.net/"
+    "http://gstreamer.freedesktop.org"
 );