tests: Add test for rtpdtmfdepay and rtpdtmfsrc

2024-11-27 12:11:13 +00:00 · 2013-01-24 21:00:08 -05:00 · 2013-01-24 21:00:08 -05:00 · 6105510a7a
commit 6105510a7a
parent 92f9a9d9ff
1 changed files with 477 additions and 0 deletions
--- a/tests/check/elements/dtmf.c
+++ b/tests/check/elements/dtmf.c
@ -0,0 +1,477 @@
+/* GStreamer
+ *
+ * unit test for dtmf elements
+ * Copyright (C) 2013 Collabora Ltd
+ *   @author: Olivier Crete <olivier.crete@collabora.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/audio/audio.h>
+#include <gst/check/gstcheck.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+
+/* Include this from the plugin to get the defines */
+
+#include "gst/dtmf/gstdtmfcommon.h"
+
+#define END_BIT (1<<7)
+
+static GstStaticPadTemplate audio_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) \"" GST_AUDIO_NE (S16) "\", "
+        "rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
+    );
+
+static GstStaticPadTemplate rtp_dtmf_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) [ 0, MAX ], "
+        "encoding-name = (string) \"TELEPHONE-EVENT\"")
+    );
+
+
+static void
+check_get_dtmf_event_message (GstBus * bus, gint number, gint volume)
+{
+  GstMessage *message;
+  gboolean have_message = FALSE;
+
+  while (!have_message &&
+      (message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
+    if (gst_message_has_name (message, "dtmf-event")) {
+      const GstStructure *s = gst_message_get_structure (message);
+      gint stype, snumber, smethod, svolume;
+
+      fail_unless (gst_structure_get (s,
+              "type", G_TYPE_INT, &stype,
+              "number", G_TYPE_INT, &snumber,
+              "method", G_TYPE_INT, &smethod,
+              "volume", G_TYPE_INT, &svolume, NULL));
+
+      fail_unless (stype == 1);
+      fail_unless (smethod == 1);
+      fail_unless (snumber == number);
+      fail_unless (svolume == volume);
+      have_message = TRUE;
+    }
+    gst_message_unref (message);
+  }
+
+  fail_unless (have_message);
+}
+
+static void
+check_no_dtmf_event_message (GstBus * bus)
+{
+  GstMessage *message;
+  gboolean have_message = FALSE;
+
+  while (!have_message &&
+      (message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
+    if (gst_message_has_name (message, "dtmf-event") ||
+        gst_message_has_name (message, "dtmf-event-processed") ||
+        gst_message_has_name (message, "dtmf-event-dropped")) {
+      have_message = TRUE;
+    }
+    gst_message_unref (message);
+  }
+
+  fail_unless (!have_message);
+}
+
+static void
+check_buffers_duration (GstClockTime expected_duration)
+{
+  GstClockTime duration = 0;
+
+  while (buffers) {
+    GstBuffer *buf = buffers->data;
+
+    buffers = g_list_delete_link (buffers, buffers);
+
+    fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
+    duration += GST_BUFFER_DURATION (buf);
+  }
+
+  fail_unless (duration == expected_duration);
+}
+
+static void
+send_rtp_packet (GstPad * src, guint timestamp, gboolean marker, gboolean end,
+    guint number, guint volume, guint duration)
+{
+  GstBuffer *buf;
+  GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
+  gchar *payload;
+  static guint seqnum = 1;
+
+  buf = gst_rtp_buffer_new_allocate (4, 0, 0);
+  fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtpbuf));
+  gst_rtp_buffer_set_seq (&rtpbuf, seqnum++);
+  gst_rtp_buffer_set_timestamp (&rtpbuf, timestamp);
+  gst_rtp_buffer_set_marker (&rtpbuf, marker);
+  payload = gst_rtp_buffer_get_payload (&rtpbuf);
+  payload[0] = number;
+  payload[1] = volume | (end ? END_BIT : 0);
+  GST_WRITE_UINT16_BE (payload + 2, duration);
+  gst_rtp_buffer_unmap (&rtpbuf);
+  fail_unless (gst_pad_push (src, buf) == GST_FLOW_OK);
+}
+
+GST_START_TEST (test_rtpdtmfdepay)
+{
+  GstElement *dtmfdepay;
+  GstPad *src, *sink;
+  GstBus *bus;
+  GstCaps *caps_in;
+  GstCaps *expected_caps_out;
+  GstCaps *caps_out;
+
+  dtmfdepay = gst_check_setup_element ("rtpdtmfdepay");
+  sink = gst_check_setup_sink_pad (dtmfdepay, &audio_sink_template);
+  src = gst_check_setup_src_pad (dtmfdepay, &rtp_dtmf_src_template);
+
+  bus = gst_bus_new ();
+  gst_element_set_bus (dtmfdepay, bus);
+
+  gst_pad_set_active (src, TRUE);
+  gst_pad_set_active (sink, TRUE);
+  gst_element_set_state (dtmfdepay, GST_STATE_PLAYING);
+
+
+  caps_in = gst_caps_new_simple ("application/x-rtp",
+      "encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT",
+      "media", G_TYPE_STRING, "audio",
+      "clock-rate", G_TYPE_INT, 1000, "payload", G_TYPE_INT, 99, NULL);
+  fail_unless (gst_pad_set_caps (src, caps_in));
+  gst_caps_unref (caps_in);
+
+  caps_out = gst_pad_get_current_caps (sink);
+  expected_caps_out = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+      "rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
+  fail_unless (gst_caps_is_equal_fixed (caps_out, expected_caps_out));
+  gst_caps_unref (expected_caps_out);
+  gst_caps_unref (caps_out);
+
+  /* Single packet DTMF */
+  send_rtp_packet (src, 200, TRUE, TRUE, 1, 5, 250);
+  check_get_dtmf_event_message (bus, 1, 5);
+  check_buffers_duration (250 * GST_MSECOND);
+
+  /* Two packet DTMF */
+  send_rtp_packet (src, 800, TRUE, FALSE, 1, 5, 200);
+  send_rtp_packet (src, 800, FALSE, TRUE, 1, 5, 400);
+  check_buffers_duration (400 * GST_MSECOND);
+  check_get_dtmf_event_message (bus, 1, 5);
+
+  /* Long DTMF */
+  send_rtp_packet (src, 3000, TRUE, FALSE, 1, 5, 200);
+  check_get_dtmf_event_message (bus, 1, 5);
+  check_buffers_duration (200 * GST_MSECOND);
+  send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 400);
+  check_no_dtmf_event_message (bus);
+  check_buffers_duration (200 * GST_MSECOND);
+  send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 600);
+  check_no_dtmf_event_message (bus);
+  check_buffers_duration (200 * GST_MSECOND);
+
+  /* New without end to last */
+  send_rtp_packet (src, 4000, TRUE, TRUE, 1, 5, 250);
+  check_get_dtmf_event_message (bus, 1, 5);
+  check_buffers_duration (250 * GST_MSECOND);
+
+  check_no_dtmf_event_message (bus);
+  fail_unless (buffers == NULL);
+  gst_element_set_bus (dtmfdepay, NULL);
+  gst_object_unref (bus);
+
+  gst_pad_set_active (src, FALSE);
+  gst_pad_set_active (sink, FALSE);
+  gst_check_teardown_sink_pad (dtmfdepay);
+  gst_check_teardown_src_pad (dtmfdepay);
+  gst_check_teardown_element (dtmfdepay);
+}
+
+GST_END_TEST;
+
+
+static GstStaticPadTemplate rtp_dtmf_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) 99, "
+        "clock-rate = (int) 1000, "
+        "seqnum-base = (uint) 333, "
+        "clock-base = (uint) 666, "
+        "ssrc = (uint) 999, "
+        "maxptime = (uint) 20, encoding-name = (string) \"TELEPHONE-EVENT\"")
+    );
+
+GstElement *rtpdtmfsrc;
+GstPad *sink;
+GstClock *testclock;
+GstBus *bus;
+
+static void
+check_message_structure (GstStructure * expected_s)
+{
+  GstMessage *message;
+  gboolean have_message = FALSE;
+
+  while (!have_message &&
+      (message = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
+              GST_MESSAGE_ELEMENT)) != NULL) {
+    if (gst_message_has_name (message, gst_structure_get_name (expected_s))) {
+      const GstStructure *s = gst_message_get_structure (message);
+
+      fail_unless (gst_structure_is_equal (s, expected_s));
+      have_message = TRUE;
+    }
+    gst_message_unref (message);
+  }
+
+  fail_unless (have_message);
+
+  gst_structure_free (expected_s);
+}
+
+static void
+check_rtp_buffer (GstClockTime ts, GstClockTime duration, gboolean start,
+    gboolean end, guint rtpts, guint ssrc, guint volume, guint number,
+    guint rtpduration)
+{
+  GstBuffer *buffer;
+  GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
+  gchar *payload;
+
+  g_mutex_lock (&check_mutex);
+  while (buffers == NULL)
+    g_cond_wait (&check_cond, &check_mutex);
+  g_mutex_unlock (&check_mutex);
+  fail_unless (buffers != NULL);
+
+  buffer = buffers->data;
+  buffers = g_list_delete_link (buffers, buffers);
+
+  fail_unless (GST_BUFFER_PTS (buffer) == ts);
+  fail_unless (GST_BUFFER_DURATION (buffer) == duration);
+
+  fail_unless (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer));
+  fail_unless (gst_rtp_buffer_get_marker (&rtpbuffer) == start);
+  fail_unless (gst_rtp_buffer_get_timestamp (&rtpbuffer) == rtpts);
+  payload = gst_rtp_buffer_get_payload (&rtpbuffer);
+
+  fail_unless (payload[0] == number);
+  fail_unless ((payload[1] & 0x7F) == volume);
+  fail_unless (! !(payload[1] & 0x80) == end);
+  fail_unless (GST_READ_UINT16_BE (payload + 2) == rtpduration);
+
+  gst_rtp_buffer_unmap (&rtpbuffer);
+  gst_buffer_unref (buffer);
+}
+
+static void
+setup_rtpdtmfsrc (void)
+{
+  testclock = gst_test_clock_new ();
+  bus = gst_bus_new ();
+
+  rtpdtmfsrc = gst_check_setup_element ("rtpdtmfsrc");
+  sink = gst_check_setup_sink_pad (rtpdtmfsrc, &rtp_dtmf_sink_template);
+  gst_element_set_bus (rtpdtmfsrc, bus);
+  fail_unless (gst_element_set_clock (rtpdtmfsrc, testclock));
+
+  gst_pad_set_active (sink, TRUE);
+  fail_unless (gst_element_set_state (rtpdtmfsrc, GST_STATE_PLAYING) ==
+      GST_STATE_CHANGE_SUCCESS);
+}
+
+static void
+teardown_rtpdtmfsrc (void)
+{
+  gst_object_unref (testclock);
+  gst_pad_set_active (sink, FALSE);
+  gst_element_set_bus (rtpdtmfsrc, NULL);
+  gst_object_unref (bus);
+  gst_check_teardown_sink_pad (rtpdtmfsrc);
+  gst_check_teardown_element (rtpdtmfsrc);
+}
+
+GST_START_TEST (test_rtpdtmfsrc_invalid_events)
+{
+  GstStructure *s;
+
+  /* Missing start */
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
+      "method", G_TYPE_INT, 1, "volume", G_TYPE_INT, 8, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
+
+  /* Missing volume */
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
+      "method", G_TYPE_INT, 1, "start", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
+
+  /* Missing number */
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
+      "volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
+
+  /* Missing type */
+  s = gst_structure_new ("dtmf-event",
+      "number", G_TYPE_INT, 3, "method", G_TYPE_INT, 1,
+      "volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
+
+  /* Stop before start */
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
+      "method", G_TYPE_INT, 1, "volume", G_TYPE_INT, 8,
+      "start", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
+
+  gst_element_set_state (rtpdtmfsrc, GST_STATE_NULL);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtpdtmfsrc_min_duration)
+{
+  GstStructure *s;
+  GstClockID id;
+  guint timestamp = 0;
+  GstCaps *expected_caps, *caps;
+
+  /* Minimum duration dtmf */
+
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
+      "method", G_TYPE_INT, 1, "volume", G_TYPE_INT, 8,
+      "start", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
+              gst_structure_copy (s))));
+  gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
+  fail_unless (buffers == NULL);
+  id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
+  fail_unless (gst_clock_id_get_time (id) == 0);
+  gst_clock_id_unref (id);
+  gst_structure_set_name (s, "dtmf-event-processed");
+  check_message_structure (s);
+
+  s = gst_structure_new ("dtmf-event",
+      "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
+      "start", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_pad_push_event (sink,
+          gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
+              gst_structure_copy (s))));
+
+  check_rtp_buffer (0, 20 * GST_MSECOND, TRUE, FALSE, 666, 999, 8, 3, 20);
+
+  for (timestamp = 20; timestamp < MIN_PULSE_DURATION + 20; timestamp += 20) {
+    gst_test_clock_advance_time (GST_TEST_CLOCK (testclock),
+        20 * GST_MSECOND + 1);
+    gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
+    fail_unless (buffers == NULL);
+    id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
+    fail_unless (gst_clock_id_get_time (id) == timestamp * GST_MSECOND);
+    gst_clock_id_unref (id);
+
+    if (timestamp < MIN_PULSE_DURATION) {
+      check_rtp_buffer (timestamp * GST_MSECOND, 20 * GST_MSECOND, FALSE,
+          FALSE, 666, 999, 8, 3, timestamp + 20);
+      check_no_dtmf_event_message (bus);
+    } else {
+      gst_structure_set_name (s, "dtmf-event-processed");
+      check_message_structure (s);
+      check_rtp_buffer (timestamp * GST_MSECOND,
+          (20 + MIN_INTER_DIGIT_INTERVAL) * GST_MSECOND, FALSE, TRUE, 666,
+          999, 8, 3, timestamp + 20);
+    }
+
+    fail_unless (buffers == NULL);
+  }
+
+
+  fail_unless (gst_test_clock_peek_id_count (GST_TEST_CLOCK (testclock)) == 0);
+
+  /* caps check */
+
+  expected_caps = gst_caps_new_simple ("application/x-rtp",
+      "encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT",
+      "media", G_TYPE_STRING, "audio",
+      "clock-rate", G_TYPE_INT, 1000, "payload", G_TYPE_INT, 99,
+      "seqnum-base", G_TYPE_UINT, 333,
+      "clock-base", G_TYPE_UINT, 666,
+      "ssrc", G_TYPE_UINT, 999, "ptime", G_TYPE_UINT, 20, NULL);
+  caps = gst_pad_get_current_caps (sink);
+  fail_unless (gst_caps_can_intersect (caps, expected_caps));
+  gst_caps_unref (caps);
+  gst_caps_unref (expected_caps);
+
+  gst_element_set_state (rtpdtmfsrc, GST_STATE_NULL);
+
+  check_no_dtmf_event_message (bus);
+}
+
+GST_END_TEST;
+
+static Suite *
+dtmf_suite (void)
+{
+  Suite *s = suite_create ("dtmf");
+  TCase *tc;
+
+  tc = tcase_create ("rtpdtmfdepay");
+  tcase_add_test (tc, test_rtpdtmfdepay);
+  suite_add_tcase (s, tc);
+
+  tc = tcase_create ("rtpdtmfsrc");
+  tcase_add_checked_fixture (tc, setup_rtpdtmfsrc, teardown_rtpdtmfsrc);
+  tcase_add_test (tc, test_rtpdtmfsrc_invalid_events);
+  tcase_add_test (tc, test_rtpdtmfsrc_min_duration);
+  suite_add_tcase (s, tc);
+
+  return s;
+}
+
+
+GST_CHECK_MAIN (dtmf);