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02447fa0b2
.. and update rtpdtmfdepay unit test to trigger the potential leak more reliably (without the fix). Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3633 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7062>
624 lines
19 KiB
C
624 lines
19 KiB
C
/* GStreamer
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*
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* unit test for dtmf elements
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* Copyright (C) 2013 Collabora Ltd
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* @author: Olivier Crete <olivier.crete@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/check/gstcheck.h>
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#include <gst/check/gsttestclock.h>
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#include <gst/rtp/gstrtpbuffer.h>
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/* Include this from the plugin to get the defines */
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#include "../../gst/dtmf/gstdtmfcommon.h"
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#define END_BIT (1<<7)
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static GstStaticPadTemplate audio_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) \"" GST_AUDIO_NE (S16) "\", "
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"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
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);
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static GstStaticPadTemplate rtp_dtmf_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) [ 0, MAX ], "
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"encoding-name = (string) \"TELEPHONE-EVENT\"")
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);
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static void
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check_get_dtmf_event_message (GstBus * bus, gint number, gint volume)
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{
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GstMessage *message;
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gboolean have_message = FALSE;
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while (!have_message &&
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(message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
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if (gst_message_has_name (message, "dtmf-event")) {
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const GstStructure *s = gst_message_get_structure (message);
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gint stype, snumber, smethod, svolume;
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fail_unless (gst_structure_get (s,
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"type", G_TYPE_INT, &stype,
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"number", G_TYPE_INT, &snumber,
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"method", G_TYPE_INT, &smethod,
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"volume", G_TYPE_INT, &svolume, NULL));
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fail_unless (stype == 1);
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fail_unless (smethod == 1);
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fail_unless (snumber == number);
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fail_unless (svolume == volume);
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have_message = TRUE;
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}
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gst_message_unref (message);
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}
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fail_unless (have_message);
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}
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static void
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check_no_dtmf_event_message (GstBus * bus)
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{
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GstMessage *message;
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gboolean have_message = FALSE;
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while (!have_message &&
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(message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
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if (gst_message_has_name (message, "dtmf-event") ||
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gst_message_has_name (message, "dtmf-event-processed") ||
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gst_message_has_name (message, "dtmf-event-dropped")) {
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have_message = TRUE;
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}
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gst_message_unref (message);
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}
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fail_unless (!have_message);
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}
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static void
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check_buffers_duration (GstClockTime expected_duration)
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{
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GstClockTime duration = 0;
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while (buffers) {
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GstBuffer *buf = buffers->data;
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buffers = g_list_delete_link (buffers, buffers);
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fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
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duration += GST_BUFFER_DURATION (buf);
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gst_buffer_unref (buf);
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}
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fail_unless (duration == expected_duration);
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}
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static void
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send_rtp_packet (GstPad * src, guint timestamp, gboolean marker, gboolean end,
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guint number, guint volume, guint duration)
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{
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GstBuffer *buf;
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GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
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gchar *payload;
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static guint seqnum = 1;
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buf = gst_rtp_buffer_new_allocate (4, 0, 0);
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fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtpbuf));
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gst_rtp_buffer_set_seq (&rtpbuf, seqnum++);
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gst_rtp_buffer_set_timestamp (&rtpbuf, timestamp);
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gst_rtp_buffer_set_marker (&rtpbuf, marker);
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payload = gst_rtp_buffer_get_payload (&rtpbuf);
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payload[0] = number;
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payload[1] = volume | (end ? END_BIT : 0);
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GST_WRITE_UINT16_BE (payload + 2, duration);
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gst_rtp_buffer_unmap (&rtpbuf);
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fail_unless (gst_pad_push (src, buf) == GST_FLOW_OK);
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}
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GST_START_TEST (test_rtpdtmfdepay)
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{
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GstElement *dtmfdepay;
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GstPad *src, *sink;
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GstBus *bus;
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GstCaps *caps_in;
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GstCaps *expected_caps_out;
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GstCaps *caps_out;
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dtmfdepay = gst_check_setup_element ("rtpdtmfdepay");
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sink = gst_check_setup_sink_pad (dtmfdepay, &audio_sink_template);
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src = gst_check_setup_src_pad (dtmfdepay, &rtp_dtmf_src_template);
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bus = gst_bus_new ();
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gst_element_set_bus (dtmfdepay, bus);
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gst_pad_set_active (src, TRUE);
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gst_pad_set_active (sink, TRUE);
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gst_element_set_state (dtmfdepay, GST_STATE_PLAYING);
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caps_in = gst_caps_new_simple ("application/x-rtp",
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"encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT",
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"media", G_TYPE_STRING, "audio",
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"clock-rate", G_TYPE_INT, 1000, "payload", G_TYPE_INT, 99, NULL);
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gst_check_setup_events (src, dtmfdepay, caps_in, GST_FORMAT_TIME);
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gst_caps_unref (caps_in);
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caps_out = gst_pad_get_current_caps (sink);
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expected_caps_out = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1,
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"layout", G_TYPE_STRING, "interleaved", NULL);
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fail_unless (gst_caps_is_equal_fixed (caps_out, expected_caps_out));
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gst_caps_unref (expected_caps_out);
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gst_caps_unref (caps_out);
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/* Single packet DTMF */
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send_rtp_packet (src, 200, TRUE, TRUE, 1, 5, 250);
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check_get_dtmf_event_message (bus, 1, 5);
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check_buffers_duration (250 * GST_MSECOND);
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/* Two packet DTMF */
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send_rtp_packet (src, 800, TRUE, FALSE, 1, 5, 200);
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send_rtp_packet (src, 800, FALSE, TRUE, 1, 5, 400);
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check_buffers_duration (400 * GST_MSECOND);
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check_get_dtmf_event_message (bus, 1, 5);
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/* Long DTMF */
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send_rtp_packet (src, 3000, TRUE, FALSE, 1, 5, 200);
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check_get_dtmf_event_message (bus, 1, 5);
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check_buffers_duration (200 * GST_MSECOND);
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send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 400);
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check_no_dtmf_event_message (bus);
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check_buffers_duration (200 * GST_MSECOND);
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send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 600);
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check_no_dtmf_event_message (bus);
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check_buffers_duration (200 * GST_MSECOND);
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/* New without end to last */
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send_rtp_packet (src, 4000, TRUE, TRUE, 1, 5, 250);
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check_get_dtmf_event_message (bus, 1, 5);
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check_buffers_duration (250 * GST_MSECOND);
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check_no_dtmf_event_message (bus);
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fail_unless (buffers == NULL);
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gst_element_set_bus (dtmfdepay, NULL);
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gst_object_unref (bus);
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gst_pad_set_active (src, FALSE);
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gst_pad_set_active (sink, FALSE);
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gst_check_teardown_sink_pad (dtmfdepay);
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gst_check_teardown_src_pad (dtmfdepay);
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gst_check_teardown_element (dtmfdepay);
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}
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GST_END_TEST;
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static GstStaticPadTemplate rtp_dtmf_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) 99, "
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"clock-rate = (int) 1000, "
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"seqnum-offset = (uint) 333, "
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"timestamp-offset = (uint) 666, "
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"ssrc = (uint) 999, "
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"maxptime = (uint) 20, encoding-name = (string) \"TELEPHONE-EVENT\"")
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);
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GstElement *dtmfsrc;
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GstPad *sink;
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GstClock *testclock;
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GstBus *bus;
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static void
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check_message_structure (GstStructure * expected_s)
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{
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GstMessage *message;
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gboolean have_message = FALSE;
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while (!have_message &&
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(message = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
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GST_MESSAGE_ELEMENT)) != NULL) {
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if (gst_message_has_name (message, gst_structure_get_name (expected_s))) {
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const GstStructure *s = gst_message_get_structure (message);
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fail_unless (gst_structure_is_equal (s, expected_s));
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have_message = TRUE;
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}
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gst_message_unref (message);
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}
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fail_unless (have_message);
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gst_structure_free (expected_s);
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}
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static void
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check_rtp_buffer (GstClockTime ts, GstClockTime duration, gboolean start,
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gboolean end, guint rtpts, guint ssrc, guint volume, guint number,
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guint rtpduration)
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{
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GstBuffer *buffer;
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GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
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gchar *payload;
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g_mutex_lock (&check_mutex);
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while (buffers == NULL)
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g_cond_wait (&check_cond, &check_mutex);
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g_mutex_unlock (&check_mutex);
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fail_unless (buffers != NULL);
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buffer = buffers->data;
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buffers = g_list_delete_link (buffers, buffers);
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fail_unless (GST_BUFFER_PTS (buffer) == ts);
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fail_unless (GST_BUFFER_DURATION (buffer) == duration);
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fail_unless (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer));
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fail_unless (gst_rtp_buffer_get_marker (&rtpbuffer) == start);
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fail_unless (gst_rtp_buffer_get_timestamp (&rtpbuffer) == rtpts);
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payload = gst_rtp_buffer_get_payload (&rtpbuffer);
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fail_unless (payload[0] == number);
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fail_unless ((payload[1] & 0x7F) == volume);
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fail_unless (!!(payload[1] & 0x80) == end);
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fail_unless (GST_READ_UINT16_BE (payload + 2) == rtpduration);
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gst_rtp_buffer_unmap (&rtpbuffer);
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gst_buffer_unref (buffer);
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}
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gint method;
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static void
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setup_rtpdtmfsrc (void)
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{
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testclock = gst_test_clock_new ();
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bus = gst_bus_new ();
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method = 1;
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dtmfsrc = gst_check_setup_element ("rtpdtmfsrc");
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sink = gst_check_setup_sink_pad (dtmfsrc, &rtp_dtmf_sink_template);
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gst_element_set_bus (dtmfsrc, bus);
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fail_unless (gst_element_set_clock (dtmfsrc, testclock));
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gst_pad_set_active (sink, TRUE);
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fail_unless (gst_element_set_state (dtmfsrc, GST_STATE_PLAYING) ==
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GST_STATE_CHANGE_SUCCESS);
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}
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static void
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teardown_dtmfsrc (void)
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{
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gst_object_unref (testclock);
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gst_pad_set_active (sink, FALSE);
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gst_element_set_bus (dtmfsrc, NULL);
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gst_object_unref (bus);
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gst_check_teardown_sink_pad (dtmfsrc);
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gst_check_teardown_element (dtmfsrc);
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}
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GST_START_TEST (test_dtmfsrc_invalid_events)
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{
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GstStructure *s;
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/* Missing start */
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
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"method", G_TYPE_INT, method, "volume", G_TYPE_INT, 8, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
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/* Missing volume */
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
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"method", G_TYPE_INT, method, "start", G_TYPE_BOOLEAN, TRUE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
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/* Missing number */
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "method", G_TYPE_INT, method,
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"volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
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/* Missing type */
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s = gst_structure_new ("dtmf-event",
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"number", G_TYPE_INT, 3, "method", G_TYPE_INT, method,
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"volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
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/* Stop before start */
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
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"method", G_TYPE_INT, method, "volume", G_TYPE_INT, 8,
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"start", G_TYPE_BOOLEAN, FALSE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
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gst_element_set_state (dtmfsrc, GST_STATE_NULL);
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}
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GST_END_TEST;
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GST_START_TEST (test_rtpdtmfsrc_min_duration)
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{
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GstStructure *s;
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GstClockID id;
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guint timestamp = 0;
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GstCaps *expected_caps, *caps;
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/* Minimum duration dtmf */
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
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"method", G_TYPE_INT, 1, "volume", G_TYPE_INT, 8,
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"start", G_TYPE_BOOLEAN, TRUE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
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gst_structure_copy (s))));
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check_no_dtmf_event_message (bus);
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
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fail_unless (buffers == NULL);
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id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
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fail_unless (gst_clock_id_get_time (id) == 0);
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gst_clock_id_unref (id);
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gst_structure_set_name (s, "dtmf-event-processed");
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check_message_structure (s);
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s = gst_structure_new ("dtmf-event",
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"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
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"start", G_TYPE_BOOLEAN, FALSE, NULL);
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fail_unless (gst_pad_push_event (sink,
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gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
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gst_structure_copy (s))));
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check_rtp_buffer (0, 20 * GST_MSECOND, TRUE, FALSE, 666, 999, 8, 3, 20);
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for (timestamp = 20; timestamp < MIN_PULSE_DURATION + 20; timestamp += 20) {
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gst_test_clock_advance_time (GST_TEST_CLOCK (testclock),
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20 * GST_MSECOND + 1);
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
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fail_unless (buffers == NULL);
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id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
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fail_unless (gst_clock_id_get_time (id) == timestamp * GST_MSECOND);
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gst_clock_id_unref (id);
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if (timestamp < MIN_PULSE_DURATION) {
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check_rtp_buffer (timestamp * GST_MSECOND, 20 * GST_MSECOND, FALSE,
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FALSE, 666, 999, 8, 3, timestamp + 20);
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check_no_dtmf_event_message (bus);
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} else {
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gst_structure_set_name (s, "dtmf-event-processed");
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check_message_structure (s);
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check_rtp_buffer (timestamp * GST_MSECOND,
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(20 + MIN_INTER_DIGIT_INTERVAL) * GST_MSECOND, FALSE, TRUE, 666,
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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-offset", G_TYPE_UINT, 333,
|
|
"timestamp-offset", 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 (dtmfsrc, GST_STATE_NULL);
|
|
|
|
check_no_dtmf_event_message (bus);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstStaticPadTemplate audio_dtmfsrc_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 = (int) 8003, " "channels = (int) 1")
|
|
);
|
|
static void
|
|
setup_dtmfsrc (void)
|
|
{
|
|
testclock = gst_test_clock_new ();
|
|
bus = gst_bus_new ();
|
|
|
|
method = 2;
|
|
dtmfsrc = gst_check_setup_element ("dtmfsrc");
|
|
sink = gst_check_setup_sink_pad (dtmfsrc, &audio_dtmfsrc_sink_template);
|
|
gst_element_set_bus (dtmfsrc, bus);
|
|
fail_unless (gst_element_set_clock (dtmfsrc, testclock));
|
|
|
|
gst_pad_set_active (sink, TRUE);
|
|
fail_unless (gst_element_set_state (dtmfsrc, GST_STATE_PLAYING) ==
|
|
GST_STATE_CHANGE_SUCCESS);
|
|
}
|
|
|
|
|
|
GST_START_TEST (test_dtmfsrc_min_duration)
|
|
{
|
|
GstStructure *s;
|
|
GstClockID id;
|
|
GstClockTime timestamp = 0;
|
|
GstCaps *expected_caps, *caps;
|
|
guint interval;
|
|
|
|
g_object_get (dtmfsrc, "interval", &interval, NULL);
|
|
fail_unless (interval == 50);
|
|
|
|
/* Minimum duration dtmf */
|
|
gst_test_clock_set_time (GST_TEST_CLOCK (testclock), 0);
|
|
|
|
s = gst_structure_new ("dtmf-event",
|
|
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
|
|
"method", G_TYPE_INT, 2, "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);
|
|
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, 2,
|
|
"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))));
|
|
|
|
for (timestamp = interval * GST_MSECOND;
|
|
timestamp < (MIN_PULSE_DURATION + MIN_INTER_DIGIT_INTERVAL) *
|
|
GST_MSECOND; timestamp += GST_MSECOND * interval) {
|
|
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
|
|
gst_test_clock_advance_time (GST_TEST_CLOCK (testclock),
|
|
interval * GST_MSECOND);
|
|
|
|
id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
|
|
fail_unless (gst_clock_id_get_time (id) == timestamp);
|
|
gst_clock_id_unref (id);
|
|
}
|
|
|
|
gst_structure_set_name (s, "dtmf-event-processed");
|
|
check_message_structure (s);
|
|
|
|
check_buffers_duration ((MIN_PULSE_DURATION + MIN_INTER_DIGIT_INTERVAL) *
|
|
GST_MSECOND);
|
|
|
|
fail_unless (gst_test_clock_peek_id_count (GST_TEST_CLOCK (testclock)) == 0);
|
|
|
|
/* caps check */
|
|
|
|
expected_caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
|
|
"rate", G_TYPE_INT, 8003, "channels", G_TYPE_INT, 1, 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 (dtmfsrc, GST_STATE_NULL);
|
|
|
|
check_no_dtmf_event_message (bus);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtpdtmfdepay_src_caps_fixated)
|
|
{
|
|
// Loop a few times, additional test for rtpdtmfsrc leak on early shutdown
|
|
for (int i = 0; i < 50; ++i) {
|
|
GstElement *pipeline;
|
|
GstStructure *s;
|
|
|
|
pipeline =
|
|
gst_parse_launch ("rtpdtmfsrc ! rtpdtmfdepay ! audioconvert ! fakesink",
|
|
NULL);
|
|
|
|
fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_PLAYING),
|
|
GST_STATE_CHANGE_ASYNC);
|
|
|
|
// Send an event so there's some data flow and the pipeline prerolls
|
|
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_element_send_event (pipeline,
|
|
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)));
|
|
|
|
// Wait for preroll
|
|
fail_unless_equals_int (gst_element_get_state (pipeline, NULL, NULL, -1),
|
|
GST_STATE_CHANGE_SUCCESS);
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
gst_object_unref (pipeline);
|
|
}
|
|
}
|
|
|
|
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);
|
|
tcase_add_test (tc, test_rtpdtmfdepay_src_caps_fixated);
|
|
suite_add_tcase (s, tc);
|
|
|
|
tc = tcase_create ("rtpdtmfsrc");
|
|
tcase_add_checked_fixture (tc, setup_rtpdtmfsrc, teardown_dtmfsrc);
|
|
tcase_add_test (tc, test_dtmfsrc_invalid_events);
|
|
tcase_add_test (tc, test_rtpdtmfsrc_min_duration);
|
|
suite_add_tcase (s, tc);
|
|
|
|
tc = tcase_create ("dtmfsrc");
|
|
tcase_add_checked_fixture (tc, setup_dtmfsrc, teardown_dtmfsrc);
|
|
tcase_add_test (tc, test_dtmfsrc_invalid_events);
|
|
tcase_add_test (tc, test_dtmfsrc_min_duration);
|
|
suite_add_tcase (s, tc);
|
|
|
|
return s;
|
|
}
|
|
|
|
|
|
GST_CHECK_MAIN (dtmf);
|