Merge branch 'master' into 0.11

[platform/upstream/gstreamer.git] / gst / audiotestsrc / gstaudiotestsrc.c

diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c
index 030322e..1ad7372 100644 (file)
--- a/gst/audiotestsrc/gstaudiotestsrc.c
+++ b/gst/audiotestsrc/gstaudiotestsrc.c
@@ -109,9 +109,8 @@ static GstStaticPadTemplate gst_audio_test_src_src_template =
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 
-
-GST_BOILERPLATE (GstAudioTestSrc, gst_audio_test_src, GstBaseSrc,
-    GST_TYPE_BASE_SRC);
+#define gst_audio_test_src_parent_class parent_class
+G_DEFINE_TYPE (GstAudioTestSrc, gst_audio_test_src, GST_TYPE_BASE_SRC);
 
 #define GST_TYPE_AUDIO_TEST_SRC_WAVE (gst_audiostestsrc_wave_get_type())
 static GType
@@ -155,7 +154,6 @@ static gboolean gst_audio_test_src_setcaps (GstBaseSrc * basesrc,
 static void gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps);
 
 static gboolean gst_audio_test_src_is_seekable (GstBaseSrc * basesrc);
-static gboolean gst_audio_test_src_check_get_range (GstBaseSrc * basesrc);
 static gboolean gst_audio_test_src_do_seek (GstBaseSrc * basesrc,
     GstSegment * segment);
 static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc,
@@ -170,27 +168,15 @@ static gboolean gst_audio_test_src_stop (GstBaseSrc * basesrc);
 static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
     guint64 offset, guint length, GstBuffer ** buffer);
 
-
-static void
-gst_audio_test_src_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_audio_test_src_src_template));
-  gst_element_class_set_details_simple (element_class,
-      "Audio test source", "Source/Audio",
-      "Creates audio test signals of given frequency and volume",
-      "Stefan Kost <ensonic@users.sf.net>");
-}
-
 static void
 gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
   GstBaseSrcClass *gstbasesrc_class;
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
   gstbasesrc_class = (GstBaseSrcClass *) klass;
 
   gobject_class->set_property = gst_audio_test_src_set_property;
@@ -233,11 +219,16 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
           "Can activate in pull mode", DEFAULT_CAN_ACTIVATE_PULL,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_audio_test_src_src_template));
+  gst_element_class_set_details_simple (gstelement_class,
+      "Audio test source", "Source/Audio",
+      "Creates audio test signals of given frequency and volume",
+      "Stefan Kost <ensonic@users.sf.net>");
+
   gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_test_src_setcaps);
   gstbasesrc_class->is_seekable =
       GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
-  gstbasesrc_class->check_get_range =
-      GST_DEBUG_FUNCPTR (gst_audio_test_src_check_get_range);
   gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
   gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_query);
   gstbasesrc_class->get_times =
@@ -248,7 +239,7 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
 }
 
 static void
-gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
+gst_audio_test_src_init (GstAudioTestSrc * src)
 {
   GstPad *pad = GST_BASE_SRC_PAD (src);
 
@@ -414,6 +405,14 @@ gst_audio_test_src_query (GstBaseSrc * basesrc, GstQuery * query)
       res = TRUE;
       break;
     }
+    case GST_QUERY_SCHEDULING:
+    {
+      /* if we can operate in pull mode */
+      gst_query_set_scheduling (query, src->can_activate_pull, TRUE, FALSE, 1,
+          -1, 1);
+      res = TRUE;
+      break;
+    }
     default:
       res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
       break;
@@ -1089,7 +1088,7 @@ gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
 
   GST_DEBUG_OBJECT (src, "seeking %" GST_SEGMENT_FORMAT, segment);
 
-  time = segment->last_stop;
+  time = segment->position;
   src->reverse = (segment->rate < 0.0);
 
   /* now move to the time indicated */
@@ -1135,28 +1134,17 @@ gst_audio_test_src_is_seekable (GstBaseSrc * basesrc)
   return TRUE;
 }
 
-static gboolean
-gst_audio_test_src_check_get_range (GstBaseSrc * basesrc)
-{
-  GstAudioTestSrc *src;
-
-  src = GST_AUDIO_TEST_SRC (basesrc);
-
-  /* if we can operate in pull mode */
-  return src->can_activate_pull;
-}
-
 static GstFlowReturn
 gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
     guint length, GstBuffer ** buffer)
 {
-  GstFlowReturn res;
   GstAudioTestSrc *src;
   GstBuffer *buf;
   GstClockTime next_time;
   gint64 next_sample, next_byte;
   gint bytes, samples;
   GstElementClass *eclass;
+  guint8 *data;
 
   src = GST_AUDIO_TEST_SRC (basesrc);
 
@@ -1220,10 +1208,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
 
   bytes = src->generate_samples_per_buffer * src->sample_size * src->channels;
 
-  if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->next_sample,
-              bytes, GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) {
-    return res;
-  }
+  buf = gst_buffer_new_and_alloc (bytes);
 
   next_byte = src->next_byte + (src->reverse ? (-bytes) : bytes);
   next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND,
@@ -1253,7 +1238,9 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
       src->generate_samples_per_buffer,
       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
 
-  src->process (src, GST_BUFFER_DATA (buf));
+  data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
+  src->process (src, data);
+  gst_buffer_unmap (buf, data, bytes);
 
   if (G_UNLIKELY ((src->wave == GST_AUDIO_TEST_SRC_WAVE_SILENCE)
           || (src->volume == 0.0))) {