--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2014 Samsung Electronics. All rights reserved.
+ * Author: Thiago Santos <ts.santos@sisa.samsung.com>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <gst/gst.h>
+#include <gst/check/gstcheck.h>
+#include <gst/audio/audio.h>
+#include <gst/app/app.h>
+
+static GstPad *mysrcpad, *mysinkpad;
+static GstElement *dec;
+static GList *events = NULL;
+
+#define TEST_MSECS_PER_SAMPLE 44100
+
+#define GST_AUDIO_DECODER_TESTER_TYPE gst_audio_decoder_tester_get_type()
+static GType gst_audio_decoder_tester_get_type (void);
+
+typedef struct _GstAudioDecoderTester GstAudioDecoderTester;
+typedef struct _GstAudioDecoderTesterClass GstAudioDecoderTesterClass;
+
+struct _GstAudioDecoderTester
+{
+ GstAudioDecoder parent;
+};
+
+struct _GstAudioDecoderTesterClass
+{
+ GstAudioDecoderClass parent_class;
+};
+
+G_DEFINE_TYPE (GstAudioDecoderTester, gst_audio_decoder_tester,
+ GST_TYPE_AUDIO_DECODER);
+
+static gboolean
+gst_audio_decoder_tester_start (GstAudioDecoder * dec)
+{
+ return TRUE;
+}
+
+static gboolean
+gst_audio_decoder_tester_stop (GstAudioDecoder * dec)
+{
+ return TRUE;
+}
+
+static void
+gst_audio_decoder_tester_flush (GstAudioDecoder * dec, gboolean hard)
+{
+}
+
+static gboolean
+gst_audio_decoder_tester_set_format (GstAudioDecoder * dec, GstCaps * caps)
+{
+ GstAudioInfo info;
+ gst_caps_unref (caps);
+
+ caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16LE",
+ "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100,
+ "layout", G_TYPE_STRING, "interleaved", NULL);
+ gst_audio_info_from_caps (&info, caps);
+ gst_caps_unref (caps);
+
+ gst_audio_decoder_set_output_format (dec, &info);
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_audio_decoder_tester_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer)
+{
+ guint8 *data;
+ gint size;
+ GstMapInfo map;
+ GstBuffer *output_buffer;
+ guint64 samples;
+
+ if (buffer == NULL)
+ return GST_FLOW_OK;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+ /* the output is SE16LE stereo 44100 Hz */
+ samples =
+ gst_util_uint64_scale_round (44100, GST_BUFFER_DURATION (buffer),
+ GST_SECOND);
+ size = 2 * 2 * samples;
+ data = g_malloc0 (size);
+
+ memcpy (data, map.data, sizeof (guint64));
+
+ output_buffer = gst_buffer_new_wrapped (data, size);
+
+ gst_buffer_unmap (buffer, &map);
+
+ return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
+}
+
+static void
+gst_audio_decoder_tester_class_init (GstAudioDecoderTesterClass * klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstAudioDecoderClass *audiosink_class = GST_AUDIO_DECODER_CLASS (klass);
+
+ static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK, GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-test-custom"));
+
+ static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC, GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw"));
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_templ));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_templ));
+
+ gst_element_class_set_metadata (element_class,
+ "AudioDecoderTester", "Decoder/Audio", "yep", "me");
+
+ audiosink_class->start = gst_audio_decoder_tester_start;
+ audiosink_class->stop = gst_audio_decoder_tester_stop;
+ audiosink_class->flush = gst_audio_decoder_tester_flush;
+ audiosink_class->handle_frame = gst_audio_decoder_tester_handle_frame;
+ audiosink_class->set_format = gst_audio_decoder_tester_set_format;
+}
+
+static void
+gst_audio_decoder_tester_init (GstAudioDecoderTester * tester)
+{
+}
+
+static gboolean
+_mysinkpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
+{
+ events = g_list_append (events, event);
+ return TRUE;
+}
+
+static void
+setup_audiodecodertester (void)
+{
+ GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw")
+ );
+ GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-test-custom")
+ );
+
+ dec = g_object_new (GST_AUDIO_DECODER_TESTER_TYPE, NULL);
+ mysrcpad = gst_check_setup_src_pad (dec, &srctemplate);
+ mysinkpad = gst_check_setup_sink_pad (dec, &sinktemplate);
+
+ gst_pad_set_event_function (mysinkpad, _mysinkpad_event);
+}
+
+static void
+cleanup_audiodecodertest (void)
+{
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (dec);
+ gst_check_teardown_sink_pad (dec);
+ gst_check_teardown_element (dec);
+}
+
+static GstBuffer *
+create_test_buffer (guint64 num)
+{
+ GstBuffer *buffer;
+ guint64 *data = g_malloc (sizeof (guint64));
+
+ *data = num;
+
+ buffer = gst_buffer_new_wrapped (data, sizeof (guint64));
+
+ GST_BUFFER_PTS (buffer) =
+ gst_util_uint64_scale_round (num, GST_SECOND, TEST_MSECS_PER_SAMPLE);
+ GST_BUFFER_DURATION (buffer) =
+ gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
+
+ return buffer;
+}
+
+static void
+send_startup_events (void)
+{
+ GstCaps *caps;
+
+ fail_unless (gst_pad_push_event (mysrcpad,
+ gst_event_new_stream_start ("randomvalue")));
+
+ /* push caps */
+ caps =
+ gst_caps_new_simple ("audio/x-test-custom", "channels", G_TYPE_INT, 2,
+ "rate", G_TYPE_INT, 44100, NULL);
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_caps (caps)));
+
+}
+
+#define NUM_BUFFERS 1000
+GST_START_TEST (audiodecoder_playback)
+{
+ GstSegment segment;
+ GstBuffer *buffer;
+ guint64 i;
+
+ setup_audiodecodertester ();
+
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_element_set_state (dec, GST_STATE_PLAYING);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ send_startup_events ();
+
+ /* push a new segment */
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
+
+ /* push buffers, the data is actually a number so we can track them */
+ for (i = 0; i < NUM_BUFFERS; i++) {
+ GstMapInfo map;
+ guint64 num;
+
+ buffer = create_test_buffer (i);
+
+ fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
+
+ /* check that buffer was received by our source pad */
+ buffer = buffers->data;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+ num = *(guint64 *) map.data;
+ fail_unless (i == num);
+ fail_unless (GST_BUFFER_PTS (buffer) == gst_util_uint64_scale_round (i,
+ GST_SECOND, TEST_MSECS_PER_SAMPLE));
+ fail_unless (GST_BUFFER_DURATION (buffer) == gst_util_uint64_scale_round (1,
+ GST_SECOND, TEST_MSECS_PER_SAMPLE));
+
+ gst_buffer_unmap (buffer, &map);
+
+ gst_buffer_unref (buffer);
+ buffers = g_list_delete_link (buffers, buffers);
+ }
+
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
+
+ fail_unless (buffers == NULL);
+
+ cleanup_audiodecodertest ();
+}
+
+GST_END_TEST;
+
+
+static Suite *
+gst_audiodecoder_suite (void)
+{
+ Suite *s = suite_create ("GstAudioDecoder");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_add_test (tc, audiodecoder_playback);
+
+ return s;
+}
+
+GST_CHECK_MAIN (gst_audiodecoder);
projects / platform / upstream / gst-plugins-base.git / commitdiff