From 6105510a7a9ab4116a687bf874b4df0ee9e2400a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Thu, 24 Jan 2013 21:00:08 -0500 Subject: [PATCH] tests: Add test for rtpdtmfdepay and rtpdtmfsrc --- tests/check/elements/dtmf.c | 477 ++++++++++++++++++++++++++++++++++++ 1 file changed, 477 insertions(+) create mode 100644 tests/check/elements/dtmf.c diff --git a/tests/check/elements/dtmf.c b/tests/check/elements/dtmf.c new file mode 100644 index 0000000000..aeaf1b8dd0 --- /dev/null +++ b/tests/check/elements/dtmf.c @@ -0,0 +1,477 @@ +/* GStreamer + * + * unit test for dtmf elements + * Copyright (C) 2013 Collabora Ltd + * @author: Olivier Crete + * + * 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 +#include +#include +#include +#include + + +/* 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);