Added gst-plugins-base-subtitles0.10-0.10.34 for Meego Harmattan 1.2

[mafwsubrenderer] / gst-plugins-base-subtitles0.10 / tests / examples / playrec / playrec.c

diff --git a/gst-plugins-base-subtitles0.10/tests/examples/playrec/playrec.c b/gst-plugins-base-subtitles0.10/tests/examples/playrec/playrec.c
new file mode 100644 (file)
index 0000000..352f65a
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+++ b/gst-plugins-base-subtitles0.10/tests/examples/playrec/playrec.c
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+/* GStreamer
+ *
+ * Copyright (C) 2010 Wim Taymans <wim.taymans@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/* An example of synchronized playback and recording.
+ * The trick is to wait for the playback pipeline to preroll before starting
+ * playback and recording.
+ */
+#include <string.h>
+
+#include <gst/gst.h>
+
+/* Define to run the asynchronous version. This requires 0.10.31 of the
+ * GStreamer core. The async version has the benefit that it doesn't block the
+ * main thread but it produces slightly less clear code. */
+#define ASYNC_VERSION
+
+static GMainLoop *loop;
+static GstElement *pipeline = NULL;
+static GstElement *play_bin, *play_source, *play_sink;
+static GstElement *rec_bin, *rec_source, *rec_sink;
+
+static gboolean
+message_handler (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+
+  switch (GST_MESSAGE_TYPE (message)) {
+#ifdef ASYNC_VERSION
+    case GST_MESSAGE_ELEMENT:{
+      const GstStructure *str;
+
+      str = gst_message_get_structure (message);
+
+      if (gst_structure_has_name (str, "GstBinForwarded")) {
+        GstMessage *orig;
+
+        /* unwrap the element message */
+        gst_structure_get (str, "message", GST_TYPE_MESSAGE, &orig, NULL);
+        g_assert (orig);
+
+        switch (GST_MESSAGE_TYPE (orig)) {
+          case GST_MESSAGE_ASYNC_DONE:
+            g_print ("ASYNC done %s\n", GST_MESSAGE_SRC_NAME (orig));
+            if (GST_MESSAGE_SRC (orig) == GST_OBJECT_CAST (play_bin)) {
+              g_print
+                  ("prerolled, starting synchronized playback and recording\n");
+              /* returns ASYNC because the sink linked to the live source is not
+               * prerolled */
+              g_assert (gst_element_set_state (pipeline,
+                      GST_STATE_PLAYING) == GST_STATE_CHANGE_ASYNC);
+            }
+            break;
+          default:
+            break;
+        }
+      }
+      break;
+    }
+#endif
+    case GST_MESSAGE_EOS:
+      g_print ("EOS\n");
+      g_main_loop_quit (loop);
+      break;
+    case GST_MESSAGE_ERROR:{
+      GError *err = NULL;
+
+      gst_message_parse_error (message, &err, NULL);
+      g_print ("error: %s\n", err->message);
+      g_clear_error (&err);
+
+      g_main_loop_quit (loop);
+      break;
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GstBus *bus;
+  gint watch_id;
+
+  gst_init (NULL, NULL);
+
+  loop = g_main_loop_new (NULL, TRUE);
+
+  pipeline = gst_pipeline_new ("pipeline");
+#ifdef ASYNC_VERSION
+  /* this enables messages of individual elements inside the pipeline */
+  g_object_set (pipeline, "message-forward", TRUE, NULL);
+#endif
+
+  /* make a bin with the playback elements this is a non-live pipeline */
+  play_bin = gst_bin_new ("play_bin");
+  play_source = gst_element_factory_make ("audiotestsrc", "play_source");
+  play_sink = gst_element_factory_make ("autoaudiosink", "play_sink");
+
+  gst_bin_add (GST_BIN (play_bin), play_source);
+  gst_bin_add (GST_BIN (play_bin), play_sink);
+
+  gst_element_link (play_source, play_sink);
+
+  /* make bin with the record elements, this is a live pipeline */
+  rec_bin = gst_bin_new ("rec_bin");
+  rec_source = gst_element_factory_make ("autoaudiosrc", "rec_source");
+  rec_sink = gst_element_factory_make ("fakesink", "rec_sink");
+
+  gst_bin_add (GST_BIN (rec_bin), rec_source);
+  gst_bin_add (GST_BIN (rec_bin), rec_sink);
+
+  gst_element_link (rec_source, rec_sink);
+
+  gst_bin_add (GST_BIN (pipeline), play_bin);
+  gst_bin_add (GST_BIN (pipeline), rec_bin);
+
+  bus = gst_element_get_bus (pipeline);
+  watch_id = gst_bus_add_watch (bus, message_handler, NULL);
+  gst_object_unref (bus);
+
+  g_print ("going to PAUSED\n");
+  /* returns NO_PREROLL because we have a live element */
+  g_assert (gst_element_set_state (pipeline,
+          GST_STATE_PAUSED) == GST_STATE_CHANGE_NO_PREROLL);
+
+  g_print ("waiting for playback preroll\n");
+#ifndef ASYNC_VERSION
+  /* sync wait for preroll on the playback bin and then go to PLAYING */
+  g_assert (gst_element_get_state (play_bin, NULL, NULL,
+          GST_CLOCK_TIME_NONE) == GST_STATE_CHANGE_SUCCESS);
+  g_print ("prerolled, starting synchronized playback and recording\n");
+  /* returns ASYNC because the sink linked to the live source is not
+   * prerolled */
+  g_assert (gst_element_set_state (pipeline,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_ASYNC);
+#endif
+
+  g_main_loop_run (loop);
+
+  g_source_remove (watch_id);
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+
+  g_main_loop_unref (loop);
+
+  return 0;
+}