tests: Add test for rtpdtmfdepay and rtpdtmfsrc

This commit is contained in:
Olivier Crête 2013-01-24 21:00:08 -05:00
parent 92f9a9d9ff
commit 6105510a7a

477
tests/check/elements/dtmf.c Normal file
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/* 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);