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-rw-r--r--tests/check/pipelines/wavpack.c217
1 files changed, 217 insertions, 0 deletions
diff --git a/tests/check/pipelines/wavpack.c b/tests/check/pipelines/wavpack.c
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+++ b/tests/check/pipelines/wavpack.c
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+/* GStreamer
+ * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/base/gstadapter.h>
+
+static gboolean
+bus_handler (GstBus * bus, GstMessage * message, gpointer data)
+{
+ GMainLoop *loop = (GMainLoop *) data;
+
+ switch (message->type) {
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (loop);
+ break;
+ case GST_MESSAGE_WARNING:
+ case GST_MESSAGE_ERROR:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ gst_message_unref (message);
+ g_error_free (gerror);
+ g_free (debug);
+ g_main_loop_quit (loop);
+ break;
+ }
+ case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s = gst_message_get_structure (message);
+
+ const gchar *name = gst_structure_get_name (s);
+
+ fail_unless (strcmp (name, "imperfect-timestamp") != 0);
+ fail_unless (strcmp (name, "imperfect-offset") != 0);
+ break;
+ }
+ default:
+ break;
+ }
+
+ return TRUE;
+}
+
+static gboolean had_first_buffer = FALSE;
+
+static void
+identity_handoff (GstElement * object, GstBuffer * buffer, gpointer user_data)
+{
+ GstAdapter *adapter = GST_ADAPTER (user_data);
+
+ gst_adapter_push (adapter, gst_buffer_ref (buffer));
+}
+
+static void
+fakesink_handoff (GstElement * object, GstBuffer * buffer, GstPad * pad,
+ gpointer user_data)
+{
+ GstAdapter *adapter = GST_ADAPTER (user_data);
+
+ /* Don't allow the second buffer with offset=0 as it's the decoded
+ * rewrite of the first
+ */
+ if (had_first_buffer == FALSE && GST_BUFFER_OFFSET (buffer) == 0)
+ had_first_buffer = TRUE;
+ else if (GST_BUFFER_OFFSET (buffer) == 0)
+ return;
+
+ gst_adapter_push (adapter, gst_buffer_ref (buffer));
+}
+
+GST_START_TEST (test_encode_decode)
+{
+ GstElement *pipeline;
+ GstElement *audiotestsrc, *identity1, *wavpackenc, *identity2, *wavpackdec,
+ *identity3, *fakesink;
+ GstAdapter *srcadapter, *sinkadapter;
+ GstBus *bus;
+ GMainLoop *loop;
+ GstBuffer *in, *out;
+ guint bus_watch = 0;
+
+ srcadapter = gst_adapter_new ();
+ fail_unless (srcadapter != NULL);
+ sinkadapter = gst_adapter_new ();
+ fail_unless (sinkadapter != NULL);
+
+ pipeline = gst_pipeline_new ("pipeline");
+ fail_unless (pipeline != NULL);
+
+ audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
+ fail_unless (audiotestsrc != NULL);
+ g_object_set (G_OBJECT (audiotestsrc), "wave", 0, "freq", 440.0,
+ "num-buffers", 2000, NULL);
+
+ identity1 = gst_element_factory_make ("identity", "identity1");
+ fail_unless (identity1 != NULL);
+ g_object_set (G_OBJECT (identity1), "signal-handoffs", TRUE, NULL);
+ g_signal_connect (G_OBJECT (identity1), "handoff",
+ G_CALLBACK (identity_handoff), srcadapter);
+
+ wavpackenc = gst_element_factory_make ("wavpackenc", "enc");
+ fail_unless (wavpackenc != NULL);
+
+ identity2 = gst_element_factory_make ("identity", "identity2");
+ fail_unless (identity2 != NULL);
+ g_object_set (G_OBJECT (identity2), "check-imperfect-timestamp", TRUE,
+ "check-imperfect-offset", TRUE, NULL);
+
+ wavpackdec = gst_element_factory_make ("wavpackdec", "dec");
+ fail_unless (wavpackdec != NULL);
+
+ identity3 = gst_element_factory_make ("identity", "identity3");
+ fail_unless (identity3 != NULL);
+ g_object_set (G_OBJECT (identity3), "check-imperfect-timestamp", TRUE,
+ "check-imperfect-offset", TRUE, NULL);
+
+ fakesink = gst_element_factory_make ("fakesink", "sink");
+ fail_unless (fakesink != NULL);
+ g_object_set (G_OBJECT (fakesink), "signal-handoffs", TRUE, NULL);
+ g_signal_connect (G_OBJECT (fakesink), "handoff",
+ G_CALLBACK (fakesink_handoff), sinkadapter);
+
+ gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, identity1, wavpackenc,
+ identity2, wavpackdec, identity3, fakesink, NULL);
+
+ fail_unless (gst_element_link_many (audiotestsrc, identity1, wavpackenc,
+ identity2, wavpackdec, identity3, fakesink, NULL));
+
+ loop = g_main_loop_new (NULL, TRUE);
+ fail_unless (loop != NULL);
+
+ bus = gst_element_get_bus (pipeline);
+ fail_unless (bus != NULL);
+ bus_watch = gst_bus_add_watch (bus, bus_handler, loop);
+ gst_object_unref (bus);
+
+ had_first_buffer = FALSE;
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+ g_main_loop_run (loop);
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+
+ fail_unless (had_first_buffer == TRUE);
+ fail_unless (gst_adapter_available (srcadapter) > 0);
+ fail_unless (gst_adapter_available (sinkadapter) > 0);
+ fail_unless_equals_int (gst_adapter_available (srcadapter),
+ gst_adapter_available (sinkadapter));
+
+ in = gst_adapter_take_buffer (srcadapter, gst_adapter_available (srcadapter));
+ fail_unless (in != NULL);
+ out =
+ gst_adapter_take_buffer (sinkadapter,
+ gst_adapter_available (sinkadapter));
+ fail_unless (out != NULL);
+
+ fail_unless_equals_int (GST_BUFFER_SIZE (in), GST_BUFFER_SIZE (out));
+ fail_unless (memcmp (GST_BUFFER_DATA (in), GST_BUFFER_DATA (out),
+ GST_BUFFER_SIZE (in)) == 0);
+
+ gst_buffer_unref (in);
+ gst_buffer_unref (out);
+ g_object_unref (pipeline);
+ g_main_loop_unref (loop);
+ g_object_unref (srcadapter);
+ g_object_unref (sinkadapter);
+ g_source_remove (bus_watch);
+}
+
+GST_END_TEST;
+
+static Suite *
+wavpack_suite (void)
+{
+ Suite *s = suite_create ("Wavpack");
+ TCase *tc_chain = tcase_create ("linear");
+
+ /* time out after 60s, not the default 3 */
+ tcase_set_timeout (tc_chain, 60);
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_encode_decode);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+ Suite *s = wavpack_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}