mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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676 lines
25 KiB
C
676 lines
25 KiB
C
/* GStreamer
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*
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* Copyright (C) 2013 Collabora Ltd.
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* @author Julien Isorce <julien.isorce@collabora.co.uk>
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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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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#define verify_buf(buf, is_rtx, expected_ssrc, expted_pt, expected_seqnum) \
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G_STMT_START { \
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GstRTPBuffer _rtp = GST_RTP_BUFFER_INIT; \
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fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &_rtp)); \
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fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&_rtp), expected_ssrc); \
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fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&_rtp), expted_pt); \
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if (!(is_rtx)) { \
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fail_unless_equals_int (gst_rtp_buffer_get_seq (&_rtp), expected_seqnum); \
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} else { \
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fail_unless_equals_int (GST_READ_UINT16_BE (gst_rtp_buffer_get_payload \
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(&_rtp)), expected_seqnum); \
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} \
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gst_rtp_buffer_unmap (&_rtp); \
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} G_STMT_END
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#define pull_and_verify(h, is_rtx, expected_ssrc, expted_pt, expected_seqnum) \
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G_STMT_START { \
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GstBuffer *_buf = gst_harness_pull (h); \
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verify_buf (_buf, is_rtx, expected_ssrc, expted_pt, expected_seqnum); \
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gst_buffer_unref (_buf); \
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} G_STMT_END
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#define push_pull_and_verify(h, buf, is_rtx, expected_ssrc, expted_pt, expected_seqnum) \
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G_STMT_START { \
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gst_harness_push (h, buf); \
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pull_and_verify (h, is_rtx, expected_ssrc, expted_pt, expected_seqnum); \
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} G_STMT_END
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static GstEvent *
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create_rtx_event (guint32 ssrc, guint8 payload_type, guint16 seqnum)
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{
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return gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
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gst_structure_new ("GstRTPRetransmissionRequest",
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"seqnum", G_TYPE_UINT, seqnum,
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"ssrc", G_TYPE_UINT, ssrc,
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"payload-type", G_TYPE_UINT, payload_type, NULL));
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}
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static void
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compare_rtp_packets (GstBuffer * a, GstBuffer * b)
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{
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GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
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GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
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gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
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gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
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fail_unless_equals_int (gst_rtp_buffer_get_header_len (&rtp_a),
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gst_rtp_buffer_get_header_len (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_version (&rtp_a),
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gst_rtp_buffer_get_version (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp_a),
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gst_rtp_buffer_get_ssrc (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_seq (&rtp_a),
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gst_rtp_buffer_get_seq (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_csrc_count (&rtp_a),
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gst_rtp_buffer_get_csrc_count (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_marker (&rtp_a),
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gst_rtp_buffer_get_marker (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&rtp_a),
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gst_rtp_buffer_get_payload_type (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_timestamp (&rtp_a),
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gst_rtp_buffer_get_timestamp (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_extension (&rtp_a),
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gst_rtp_buffer_get_extension (&rtp_b));
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fail_unless_equals_int (gst_rtp_buffer_get_payload_len (&rtp_a),
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gst_rtp_buffer_get_payload_len (&rtp_b));
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fail_unless_equals_int (memcmp (gst_rtp_buffer_get_payload (&rtp_a),
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gst_rtp_buffer_get_payload (&rtp_b),
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gst_rtp_buffer_get_payload_len (&rtp_a)), 0);
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gst_rtp_buffer_unmap (&rtp_a);
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gst_rtp_buffer_unmap (&rtp_b);
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}
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static GstBuffer *
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create_rtp_buffer (guint32 ssrc, guint8 payload_type, guint16 seqnum)
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{
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GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
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guint payload_size = 29;
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guint64 timestamp = gst_util_uint64_scale_int (seqnum, 90000, 30);
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GstBuffer *buf = gst_rtp_buffer_new_allocate (payload_size, 0, 0);
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gst_rtp_buffer_map (buf, GST_MAP_WRITE, &rtpbuf);
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gst_rtp_buffer_set_ssrc (&rtpbuf, ssrc);
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gst_rtp_buffer_set_payload_type (&rtpbuf, payload_type);
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gst_rtp_buffer_set_seq (&rtpbuf, seqnum);
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gst_rtp_buffer_set_timestamp (&rtpbuf, (guint32) timestamp);
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memset (gst_rtp_buffer_get_payload (&rtpbuf), 0x29, payload_size);
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gst_rtp_buffer_unmap (&rtpbuf);
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return buf;
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}
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static GstBuffer *
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create_rtp_buffer_with_timestamp (guint32 ssrc, guint8 payload_type,
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guint16 seqnum, guint32 timestamp)
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{
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GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
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GstBuffer *buf = create_rtp_buffer (ssrc, payload_type, seqnum);
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gst_rtp_buffer_map (buf, GST_MAP_WRITE, &rtpbuf);
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gst_rtp_buffer_set_timestamp (&rtpbuf, timestamp);
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gst_rtp_buffer_unmap (&rtpbuf);
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return buf;
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}
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GST_START_TEST (test_rtxsend_rtxreceive)
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{
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const guint packets_num = 5;
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guint master_ssrc = 1234567;
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guint master_pt = 96;
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guint rtx_pt = 99;
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GstStructure *pt_map;
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GstBuffer *inbufs[5];
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GstHarness *hrecv = gst_harness_new ("rtprtxreceive");
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GstHarness *hsend = gst_harness_new ("rtprtxsend");
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gint i;
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pt_map = gst_structure_new ("application/x-rtp-pt-map",
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"96", G_TYPE_UINT, rtx_pt, NULL);
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g_object_set (hrecv->element, "payload-type-map", pt_map, NULL);
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g_object_set (hsend->element, "payload-type-map", pt_map, NULL);
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gst_harness_set_src_caps_str (hsend, "application/x-rtp, "
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"media = (string)video, payload = (int)96, "
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"ssrc = (uint)1234567, clock-rate = (int)90000, "
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"encoding-name = (string)RAW");
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gst_harness_set_src_caps_str (hrecv, "application/x-rtp, "
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"media = (string)video, payload = (int)96, "
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"ssrc = (uint)1234567, clock-rate = (int)90000, "
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"encoding-name = (string)RAW");
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/* Push 'packets_num' packets through rtxsend to rtxreceive */
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for (i = 0; i < packets_num; ++i) {
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inbufs[i] = create_rtp_buffer (master_ssrc, master_pt, 100 + i);
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gst_harness_push (hsend, gst_buffer_ref (inbufs[i]));
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gst_harness_push (hrecv, gst_harness_pull (hsend));
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pull_and_verify (hrecv, FALSE, master_ssrc, master_pt, 100 + i);
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}
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/* Getting rid of reconfigure event. Preparation before the next step */
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gst_event_unref (gst_harness_pull_upstream_event (hrecv));
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fail_unless_equals_int (gst_harness_upstream_events_in_queue (hrecv), 0);
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/* Push 'packets_num' RTX events through rtxreceive to rtxsend.
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Push RTX packets from rtxsend to rtxreceive and
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check that the packet produced out of RTX packet is the same
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as an original packet */
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for (i = 0; i < packets_num; ++i) {
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GstBuffer *outbuf;
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gst_harness_push_upstream_event (hrecv,
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create_rtx_event (master_ssrc, master_pt, 100 + i));
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gst_harness_push_upstream_event (hsend,
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gst_harness_pull_upstream_event (hrecv));
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gst_harness_push (hrecv, gst_harness_pull (hsend));
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outbuf = gst_harness_pull (hrecv);
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compare_rtp_packets (inbufs[i], outbuf);
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gst_buffer_unref (inbufs[i]);
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gst_buffer_unref (outbuf);
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}
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/* Check RTX stats */
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{
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guint rtx_requests;
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guint rtx_packets;
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guint rtx_assoc_packets;
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g_object_get (G_OBJECT (hsend->element),
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"num-rtx-requests", &rtx_requests,
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"num-rtx-packets", &rtx_packets, NULL);
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fail_unless_equals_int (rtx_packets, packets_num);
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fail_unless_equals_int (rtx_requests, packets_num);
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g_object_get (G_OBJECT (hrecv->element),
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"num-rtx-requests", &rtx_requests,
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"num-rtx-packets", &rtx_packets,
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"num-rtx-assoc-packets", &rtx_assoc_packets, NULL);
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fail_unless_equals_int (rtx_packets, packets_num);
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fail_unless_equals_int (rtx_requests, packets_num);
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fail_unless_equals_int (rtx_assoc_packets, packets_num);
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}
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gst_structure_free (pt_map);
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gst_harness_teardown (hrecv);
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gst_harness_teardown (hsend);
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}
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GST_END_TEST;
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GST_START_TEST (test_rtxsend_rtxreceive_with_packet_loss)
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{
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guint packets_num = 20;
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guint master_ssrc = 1234567;
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guint master_pt = 96;
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guint rtx_pt = 99;
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guint seqnum = 100;
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guint expected_rtx_packets = 0;
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GstStructure *pt_map;
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GstHarness *hrecv = gst_harness_new ("rtprtxreceive");
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GstHarness *hsend = gst_harness_new ("rtprtxsend");
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gint drop_nth_packet, i;
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pt_map = gst_structure_new ("application/x-rtp-pt-map",
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"96", G_TYPE_UINT, rtx_pt, NULL);
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g_object_set (hrecv->element, "payload-type-map", pt_map, NULL);
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g_object_set (hsend->element, "payload-type-map", pt_map, NULL);
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gst_harness_set_src_caps_str (hsend, "application/x-rtp, "
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"media = (string)video, payload = (int)96, "
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"ssrc = (uint)1234567, clock-rate = (int)90000, "
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"encoding-name = (string)RAW");
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gst_harness_set_src_caps_str (hrecv, "application/x-rtp, "
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"media = (string)video, payload = (int)96, "
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"ssrc = (uint)1234567, clock-rate = (int)90000, "
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"encoding-name = (string)RAW");
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/* Getting rid of reconfigure event. Making sure there is no upstream
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events in the queue. Preparation step before the test. */
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gst_event_unref (gst_harness_pull_upstream_event (hrecv));
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fail_unless_equals_int (gst_harness_upstream_events_in_queue (hrecv), 0);
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/* Push 'packets_num' packets through rtxsend to rtxreceive loosing every
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'drop_every_n_packets' packet. When we loose the packet we send RTX event
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through rtxreceive to rtxsend, and verify the packet was retransmitted */
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for (drop_nth_packet = 2; drop_nth_packet < 10; ++drop_nth_packet) {
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for (i = 0; i < packets_num; ++i, ++seqnum) {
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GstBuffer *outbuf;
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GstBuffer *inbuf = create_rtp_buffer (master_ssrc, master_pt, seqnum);
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gboolean drop_this_packet = ((i + 1) % drop_nth_packet) == 0;
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gst_harness_push (hsend, gst_buffer_ref (inbuf));
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if (drop_this_packet) {
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/* Dropping original packet */
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gst_buffer_unref (gst_harness_pull (hsend));
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/* Requesting retransmission */
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gst_harness_push_upstream_event (hrecv,
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create_rtx_event (master_ssrc, master_pt, seqnum));
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gst_harness_push_upstream_event (hsend,
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gst_harness_pull_upstream_event (hrecv));
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/* Pushing RTX packet to rtxreceive */
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gst_harness_push (hrecv, gst_harness_pull (hsend));
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++expected_rtx_packets;
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} else {
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gst_harness_push (hrecv, gst_harness_pull (hsend));
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}
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/* We making sure every buffer we pull is the same as original input
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buffer */
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outbuf = gst_harness_pull (hrecv);
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compare_rtp_packets (inbuf, outbuf);
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gst_buffer_unref (inbuf);
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gst_buffer_unref (outbuf);
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/*
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We should not have any packets in the harness queue by this point. It
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means rtxsend didn't send more packets than RTX events and rtxreceive
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didn't produce more than one packet per RTX packet.
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*/
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fail_unless_equals_int (gst_harness_buffers_in_queue (hsend), 0);
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fail_unless_equals_int (gst_harness_buffers_in_queue (hrecv), 0);
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}
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}
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/* Check RTX stats */
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{
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guint rtx_requests;
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guint rtx_packets;
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guint rtx_assoc_packets;
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g_object_get (G_OBJECT (hsend->element),
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"num-rtx-requests", &rtx_requests,
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"num-rtx-packets", &rtx_packets, NULL);
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fail_unless_equals_int (rtx_packets, expected_rtx_packets);
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fail_unless_equals_int (rtx_requests, expected_rtx_packets);
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g_object_get (G_OBJECT (hrecv->element),
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"num-rtx-requests", &rtx_requests,
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"num-rtx-packets", &rtx_packets,
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"num-rtx-assoc-packets", &rtx_assoc_packets, NULL);
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fail_unless_equals_int (rtx_packets, expected_rtx_packets);
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fail_unless_equals_int (rtx_requests, expected_rtx_packets);
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fail_unless_equals_int (rtx_assoc_packets, expected_rtx_packets);
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}
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gst_structure_free (pt_map);
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gst_harness_teardown (hrecv);
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gst_harness_teardown (hsend);
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}
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GST_END_TEST;
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typedef struct
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{
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GstHarness *h;
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guint master_ssrc;
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guint master_pt;
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guint rtx_ssrc;
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guint rtx_pt;
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guint seqnum;
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guint expected_rtx_packets;
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} RtxSender;
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static GstStructure *
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create_rtxsenders (RtxSender * senders, guint senders_num)
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{
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GstStructure *recv_pt_map =
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gst_structure_new_empty ("application/x-rtp-pt-map");
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gint i;
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for (i = 0; i < senders_num; ++i) {
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gchar *master_pt_str;
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gchar *master_caps_str;
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GstStructure *send_pt_map;
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senders[i].h = gst_harness_new ("rtprtxsend");
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senders[i].master_ssrc = 1234567 + i;
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senders[i].rtx_ssrc = 7654321 + i;
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senders[i].master_pt = 80 + i;
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senders[i].rtx_pt = 20 + i;
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senders[i].seqnum = i * 1000;
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senders[i].expected_rtx_packets = 0;
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master_pt_str = g_strdup_printf ("%u", senders[i].master_pt);
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master_caps_str = g_strdup_printf ("application/x-rtp, "
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"media = (string)video, payload = (int)%u, "
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"ssrc = (uint)%u, clock-rate = (int)90000, "
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"encoding-name = (string)RAW",
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senders[i].master_pt, senders[i].master_ssrc);
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send_pt_map = gst_structure_new ("application/x-rtp-pt-map",
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master_pt_str, G_TYPE_UINT, senders[i].rtx_pt, NULL);
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gst_structure_set (recv_pt_map,
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master_pt_str, G_TYPE_UINT, senders[i].rtx_pt, NULL);
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g_object_set (senders[i].h->element, "payload-type-map", send_pt_map, NULL);
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gst_harness_set_src_caps_str (senders[i].h, master_caps_str);
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gst_structure_free (send_pt_map);
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g_free (master_pt_str);
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g_free (master_caps_str);
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}
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return recv_pt_map;
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}
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static guint
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check_rtxsenders_stats_and_teardown (RtxSender * senders, guint senders_num)
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{
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guint total_pakets_num = 0;
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gint i;
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for (i = 0; i < senders_num; ++i) {
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guint rtx_requests;
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guint rtx_packets;
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g_object_get (G_OBJECT (senders[i].h->element),
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"num-rtx-requests", &rtx_requests,
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"num-rtx-packets", &rtx_packets, NULL);
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fail_unless_equals_int (rtx_packets, senders[i].expected_rtx_packets);
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fail_unless_equals_int (rtx_requests, senders[i].expected_rtx_packets);
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total_pakets_num += rtx_packets;
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gst_harness_teardown (senders[i].h);
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}
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return total_pakets_num;
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}
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GST_START_TEST (test_multi_rtxsend_rtxreceive_with_packet_loss)
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{
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guint senders_num = 5;
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guint packets_num = 10;
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guint total_pakets_num = senders_num * packets_num;
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guint total_dropped_packets = 0;
|
|
RtxSender senders[5];
|
|
GstStructure *pt_map;
|
|
GstHarness *hrecv = gst_harness_new ("rtprtxreceive");
|
|
gint drop_nth_packet, i, j;
|
|
|
|
pt_map = create_rtxsenders (senders, 5);
|
|
g_object_set (hrecv->element, "payload-type-map", pt_map, NULL);
|
|
gst_harness_set_src_caps_str (hrecv, "application/x-rtp, "
|
|
"media = (string)video, payload = (int)80, "
|
|
"ssrc = (uint)1234567, clock-rate = (int)90000, "
|
|
"encoding-name = (string)RAW");
|
|
|
|
/* Getting rid of reconfigure event. Making sure there is no upstream
|
|
events in the queue. Preparation step before the test. */
|
|
gst_event_unref (gst_harness_pull_upstream_event (hrecv));
|
|
fail_unless_equals_int (gst_harness_upstream_events_in_queue (hrecv), 0);
|
|
|
|
/* We are going to push the 1st packet from the 1st sender, 2nd from the 2nd,
|
|
3rd from the 3rd, etc. until all the senders will push 'packets_num' packets.
|
|
We will drop every 'drop_nth_packet' packet and request its retransmission
|
|
from all the senders. Because only one of them can produce RTX packet.
|
|
We need to make sure that all other senders will ignore the RTX event they
|
|
can't act upon.
|
|
*/
|
|
for (drop_nth_packet = 2; drop_nth_packet < 5; ++drop_nth_packet) {
|
|
for (i = 0; i < total_pakets_num; ++i) {
|
|
RtxSender *sender = &senders[i % senders_num];
|
|
gboolean drop_this_packet = ((i + 1) % drop_nth_packet) == 0;
|
|
GstBuffer *outbuf, *inbuf;
|
|
inbuf =
|
|
create_rtp_buffer (sender->master_ssrc, sender->master_pt,
|
|
sender->seqnum);
|
|
|
|
gst_harness_push (sender->h, gst_buffer_ref (inbuf));
|
|
if (drop_this_packet) {
|
|
GstEvent *rtxevent;
|
|
/* Dropping original packet */
|
|
gst_buffer_unref (gst_harness_pull (sender->h));
|
|
|
|
/* Pushing RTX event through rtxreceive to all the senders */
|
|
gst_harness_push_upstream_event (hrecv,
|
|
create_rtx_event (sender->master_ssrc, sender->master_pt,
|
|
sender->seqnum));
|
|
rtxevent = gst_harness_pull_upstream_event (hrecv);
|
|
|
|
/* ... to all the senders */
|
|
for (j = 0; j < senders_num; ++j)
|
|
gst_harness_push_upstream_event (senders[j].h,
|
|
gst_event_ref (rtxevent));
|
|
gst_event_unref (rtxevent);
|
|
|
|
/* Pushing RTX packet to rtxreceive */
|
|
gst_harness_push (hrecv, gst_harness_pull (sender->h));
|
|
++sender->expected_rtx_packets;
|
|
++total_dropped_packets;
|
|
} else {
|
|
gst_harness_push (hrecv, gst_harness_pull (sender->h));
|
|
}
|
|
|
|
/* It should not matter whether the buffer was dropped (and retransmitted)
|
|
or it went straight through rtxsend to rtxreceive. We should always pull
|
|
the same buffer that was pushed */
|
|
outbuf = gst_harness_pull (hrecv);
|
|
compare_rtp_packets (inbuf, outbuf);
|
|
gst_buffer_unref (inbuf);
|
|
gst_buffer_unref (outbuf);
|
|
|
|
/*
|
|
We should not have any packets in the harness queue by this point. It
|
|
means our senders didn't produce the packets for the unknown RTX event.
|
|
*/
|
|
for (j = 0; j < senders_num; ++j)
|
|
fail_unless_equals_int (gst_harness_buffers_in_queue (senders[j].h), 0);
|
|
|
|
++sender->seqnum;
|
|
}
|
|
}
|
|
|
|
/* Check RTX stats */
|
|
{
|
|
guint total_rtx_packets;
|
|
guint rtx_requests;
|
|
guint rtx_packets;
|
|
guint rtx_assoc_packets;
|
|
|
|
total_rtx_packets =
|
|
check_rtxsenders_stats_and_teardown (senders, senders_num);
|
|
fail_unless_equals_int (total_rtx_packets, total_dropped_packets);
|
|
|
|
g_object_get (G_OBJECT (hrecv->element),
|
|
"num-rtx-requests", &rtx_requests,
|
|
"num-rtx-packets", &rtx_packets,
|
|
"num-rtx-assoc-packets", &rtx_assoc_packets, NULL);
|
|
fail_unless_equals_int (rtx_packets, total_rtx_packets);
|
|
fail_unless_equals_int (rtx_requests, total_rtx_packets);
|
|
fail_unless_equals_int (rtx_assoc_packets, total_rtx_packets);
|
|
}
|
|
|
|
gst_structure_free (pt_map);
|
|
gst_harness_teardown (hrecv);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
test_rtxsender_packet_retention (gboolean test_with_time)
|
|
{
|
|
guint master_ssrc = 1234567;
|
|
guint master_pt = 96;
|
|
guint rtx_ssrc = 7654321;
|
|
guint rtx_pt = 99;
|
|
gint num_buffers = test_with_time ? 30 : 10;
|
|
gint half_buffers = num_buffers / 2;
|
|
guint timestamp_delta = 90000 / 30;
|
|
guint timestamp = G_MAXUINT32 - half_buffers * timestamp_delta;
|
|
GstHarness *h;
|
|
GstStructure *pt_map = gst_structure_new ("application/x-rtp-pt-map",
|
|
"96", G_TYPE_UINT, rtx_pt, NULL);
|
|
GstStructure *ssrc_map = gst_structure_new ("application/x-rtp-ssrc-map",
|
|
"1234567", G_TYPE_UINT, rtx_ssrc, NULL);
|
|
gint i, j;
|
|
|
|
h = gst_harness_new ("rtprtxsend");
|
|
|
|
/* In both cases we want the rtxsend queue to store 'half_buffers'
|
|
amount of buffers at most. In max-size-packets mode, it's trivial.
|
|
In max-size-time mode, we specify almost half a second, which is
|
|
the equivalent of 15 frames in a 30fps video stream.
|
|
*/
|
|
g_object_set (h->element,
|
|
"max-size-packets", test_with_time ? 0 : half_buffers,
|
|
"max-size-time", test_with_time ? 499 : 0,
|
|
"payload-type-map", pt_map, "ssrc-map", ssrc_map, NULL);
|
|
|
|
gst_harness_set_src_caps_str (h, "application/x-rtp, "
|
|
"media = (string)video, payload = (int)96, "
|
|
"ssrc = (uint)1234567, clock-rate = (int)90000, "
|
|
"encoding-name = (string)RAW");
|
|
|
|
/* Now push all buffers and request retransmission every time for all of them */
|
|
for (i = 0; i < num_buffers; ++i, timestamp += timestamp_delta) {
|
|
/* Request to retransmit all the previous ones */
|
|
for (j = 0; j < i; ++j) {
|
|
guint rtx_seqnum = 0x100 + j;
|
|
gst_harness_push_upstream_event (h,
|
|
create_rtx_event (master_ssrc, master_pt, rtx_seqnum));
|
|
|
|
/* Pull only the ones supposed to be retransmited */
|
|
if (j >= i - half_buffers)
|
|
pull_and_verify (h, TRUE, rtx_ssrc, rtx_pt, rtx_seqnum);
|
|
}
|
|
/* Check there no extra buffers in the harness queue */
|
|
fail_unless_equals_int (gst_harness_buffers_in_queue (h), 0);
|
|
|
|
/* We create RTP buffers with timestamps that will eventualy wrap around 0
|
|
to be sure, rtprtxsend can handle it properly */
|
|
push_pull_and_verify (h,
|
|
create_rtp_buffer_with_timestamp (master_ssrc, master_pt, 0x100 + i,
|
|
timestamp), FALSE, master_ssrc, master_pt, 0x100 + i);
|
|
}
|
|
|
|
gst_structure_free (pt_map);
|
|
gst_structure_free (ssrc_map);
|
|
gst_harness_teardown (h);
|
|
}
|
|
|
|
GST_START_TEST (test_rtxsender_max_size_packets)
|
|
{
|
|
test_rtxsender_packet_retention (FALSE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtxsender_max_size_time)
|
|
{
|
|
test_rtxsender_packet_retention (TRUE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
test_rtxqueue_packet_retention (gboolean test_with_time)
|
|
{
|
|
guint ssrc = 1234567;
|
|
guint pt = 96;
|
|
gint num_buffers = test_with_time ? 30 : 10;
|
|
gint half_buffers = num_buffers / 2;
|
|
GstClockTime timestamp_delta = GST_SECOND / 30;
|
|
GstClockTime timestamp = 0;
|
|
GstBuffer *buf;
|
|
GstHarness *h;
|
|
gint i, j;
|
|
|
|
h = gst_harness_new ("rtprtxqueue");
|
|
|
|
/* In both cases we want the rtxqueue to store 'half_buffers'
|
|
amount of buffers at most. In max-size-packets mode, it's trivial.
|
|
In max-size-time mode, we specify almost half a second, which is
|
|
the equivalent of 15 frames in a 30fps video stream.
|
|
*/
|
|
g_object_set (h->element,
|
|
"max-size-packets", test_with_time ? 0 : half_buffers,
|
|
"max-size-time", test_with_time ? 498 : 0, NULL);
|
|
|
|
gst_harness_set_src_caps_str (h, "application/x-rtp, "
|
|
"media = (string)video, payload = (int)96, "
|
|
"ssrc = (uint)1234567, clock-rate = (int)90000, "
|
|
"encoding-name = (string)RAW");
|
|
|
|
/* Now push all buffers and request retransmission every time for all of them.
|
|
* Note that rtprtxqueue sends retransmissions in chain(), just before
|
|
* pushing out the chained buffer, a differentiation from rtprtxsend above
|
|
*/
|
|
for (i = 0; i < num_buffers; ++i, timestamp += timestamp_delta) {
|
|
/* Request to retransmit all the previous ones */
|
|
for (j = 0; j < i; ++j) {
|
|
guint rtx_seqnum = 0x100 + j;
|
|
gst_harness_push_upstream_event (h,
|
|
create_rtx_event (ssrc, pt, rtx_seqnum));
|
|
}
|
|
|
|
/* push one packet */
|
|
buf = create_rtp_buffer (ssrc, pt, 0x100 + i);
|
|
GST_BUFFER_TIMESTAMP (buf) = timestamp;
|
|
gst_harness_push (h, buf);
|
|
|
|
/* Pull the ones supposed to be retransmitted */
|
|
for (j = 0; j < i; ++j) {
|
|
guint rtx_seqnum = 0x100 + j;
|
|
if (j >= i - half_buffers)
|
|
pull_and_verify (h, FALSE, ssrc, pt, rtx_seqnum);
|
|
}
|
|
|
|
/* There should be only one packet remaining in the queue now */
|
|
fail_unless_equals_int (gst_harness_buffers_in_queue (h), 1);
|
|
|
|
/* pull the one that we just pushed (comes after the retransmitted ones) */
|
|
pull_and_verify (h, FALSE, ssrc, pt, 0x100 + i);
|
|
|
|
/* Check there no extra buffers in the harness queue */
|
|
fail_unless_equals_int (gst_harness_buffers_in_queue (h), 0);
|
|
}
|
|
|
|
gst_harness_teardown (h);
|
|
}
|
|
|
|
GST_START_TEST (test_rtxqueue_max_size_packets)
|
|
{
|
|
test_rtxqueue_packet_retention (FALSE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtxqueue_max_size_time)
|
|
{
|
|
test_rtxqueue_packet_retention (TRUE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtprtx_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtprtx");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
tcase_set_timeout (tc_chain, 120);
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
|
|
tcase_add_test (tc_chain, test_rtxsend_rtxreceive);
|
|
tcase_add_test (tc_chain, test_rtxsend_rtxreceive_with_packet_loss);
|
|
tcase_add_test (tc_chain, test_multi_rtxsend_rtxreceive_with_packet_loss);
|
|
tcase_add_test (tc_chain, test_rtxsender_max_size_packets);
|
|
tcase_add_test (tc_chain, test_rtxsender_max_size_time);
|
|
tcase_add_test (tc_chain, test_rtxqueue_max_size_packets);
|
|
tcase_add_test (tc_chain, test_rtxqueue_max_size_time);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtprtx);
|