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tests/check: Add rtprtx::test_rtxsender_packet_retention
This unit test verifies that the rtxsend element correctly maintains a buffer of already transmitted rtp packets and that it can re-transmit all of them correctly on demand. It also verifies that the limit of this buffer (max-size-packets property) is respected.
This commit is contained in:
George Kiagiadakis
Wim Taymans
parent
71bdb5e088
commit
f7277db9e4
1 changed files with 230 additions and 1 deletions
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@ -1058,19 +1058,248 @@ GST_START_TEST (test_drop_multiple_sender)
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GST_END_TEST;
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struct GenerateTestBuffersData
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{
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GstElement *src, *payloader, *sink;
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GMutex mutex;
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GCond cond;
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GList *buffers;
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gint num_buffers;
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guint last_seqnum;
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};
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static void
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fakesink_handoff (GstElement * sink, GstBuffer * buf, GstPad * pad,
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gpointer user_data)
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{
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struct GenerateTestBuffersData *data = user_data;
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g_mutex_lock (&data->mutex);
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if (data->num_buffers > 0)
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data->buffers = g_list_append (data->buffers, gst_buffer_ref (buf));
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/* if we have collected enough buffers, unblock the main thread to stop */
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if (--data->num_buffers <= 0)
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g_cond_signal (&data->cond);
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if (data->num_buffers == 0)
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g_object_get (data->payloader, "seqnum", &data->last_seqnum, NULL);
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g_mutex_unlock (&data->mutex);
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}
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static GList *
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generate_test_buffers (const gint num_buffers, guint ssrc, guint * payload_type)
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{
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GstElement *bin;
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gboolean res;
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struct GenerateTestBuffersData data;
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fail_unless (num_buffers > 0);
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g_mutex_init (&data.mutex);
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g_cond_init (&data.cond);
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data.buffers = NULL;
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data.num_buffers = num_buffers;
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bin = gst_pipeline_new (NULL);
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data.src = gst_element_factory_make ("videotestsrc", NULL);
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data.payloader = gst_element_factory_make ("rtpvrawpay", NULL);
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data.sink = gst_element_factory_make ("fakesink", NULL);
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g_object_set (data.payloader, "seqnum-offset", 1, "ssrc", ssrc, NULL);
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g_object_set (data.sink, "signal-handoffs", TRUE, NULL);
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g_signal_connect (data.sink, "handoff", (GCallback) fakesink_handoff, &data);
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gst_bin_add_many (GST_BIN (bin), data.src, data.payloader, data.sink, NULL);
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res = gst_element_link_many (data.src, data.payloader, data.sink, NULL);
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fail_unless_equals_int (res, TRUE);
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g_mutex_lock (&data.mutex);
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ASSERT_SET_STATE (bin, GST_STATE_PLAYING, GST_STATE_CHANGE_ASYNC);
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while (data.num_buffers > 0)
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g_cond_wait (&data.cond, &data.mutex);
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g_mutex_unlock (&data.mutex);
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g_object_get (data.payloader, "pt", payload_type, NULL);
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ASSERT_SET_STATE (bin, GST_STATE_NULL, GST_STATE_CHANGE_SUCCESS);
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fail_unless_equals_int (g_list_length (data.buffers), num_buffers);
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fail_unless_equals_int (num_buffers, data.last_seqnum);
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g_mutex_clear (&data.mutex);
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g_cond_clear (&data.cond);
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gst_object_unref (bin);
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return data.buffers;
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}
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static GstEvent *
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create_rtx_event (guint seqnum, guint ssrc, guint payload_type)
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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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GST_START_TEST (test_rtxsender_packet_retention)
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{
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const gint num_buffers = 10;
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const gint half_buffers = num_buffers / 2;
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const guint ssrc = 1234567;
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const guint rtx_payload_type = 99;
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GList *in_buffers, *node;
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guint payload_type;
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GstElement *rtxsend;
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GstPad *srcpad, *sinkpad;
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GstCaps *caps;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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gint i, j;
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gboolean res;
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/* generate test data */
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in_buffers = generate_test_buffers (num_buffers, ssrc, &payload_type);
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/* clear the global buffers list, which we are going to use later */
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gst_check_drop_buffers ();
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/* setup element & pads */
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rtxsend = gst_check_setup_element ("rtprtxsend");
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g_object_set (rtxsend, "max-size-packets", half_buffers,
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"rtx-payload-type", rtx_payload_type, NULL);
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srcpad = gst_check_setup_src_pad (rtxsend, &srctemplate);
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fail_unless_equals_int (gst_pad_set_active (srcpad, TRUE), TRUE);
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sinkpad = gst_check_setup_sink_pad (rtxsend, &sinktemplate);
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fail_unless_equals_int (gst_pad_set_active (sinkpad, TRUE), TRUE);
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ASSERT_SET_STATE (rtxsend, GST_STATE_PLAYING, GST_STATE_CHANGE_SUCCESS);
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caps = gst_caps_from_string ("application/x-rtp, "
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"media = (string)video, payload = (int)96, "
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"clock-rate = (int)90000, encoding-name = (string)RAW");
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gst_check_setup_events (srcpad, rtxsend, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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/* now push all buffers and request retransmission every time for all of them */
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node = in_buffers;
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for (i = 1; i <= num_buffers; i++) {
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GstBuffer *buffer = GST_BUFFER (node->data);
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/* verify that the original packets are correct */
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res = gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
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fail_unless_equals_int (res, TRUE);
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fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
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fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&rtp),
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payload_type);
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fail_unless_equals_int (gst_rtp_buffer_get_seq (&rtp), i);
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gst_rtp_buffer_unmap (&rtp);
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/* retransmit all the previous ones */
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for (j = 1; j < i; j++) {
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res = gst_pad_push_event (sinkpad,
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create_rtx_event (j, ssrc, payload_type));
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fail_unless_equals_int (res, TRUE);
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}
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/* push this one, triggering the retransmit in rtxsend's chain() function */
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gst_pad_push (srcpad, gst_buffer_ref (buffer));
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node = g_list_next (node);
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}
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/* verify the result. buffers should be in this order (numbers are seqnums):
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* 1, 1rtx, 2, 2rtx, 1rtx, 3, ... , 9, 9rtx, 8rtx, 7rtx, 6rtx, 5rtx, 10 */
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{
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GstRTPBuffer orig_rtp = GST_RTP_BUFFER_INIT;
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gint expected_rtx_requests, expected_rtx_packets;
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gint real_rtx_requests, real_rtx_packets;
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/* verify statistics first */
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expected_rtx_packets = half_buffers * half_buffers +
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((half_buffers - 1) / 2.0f) * half_buffers;
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for (i = 1, expected_rtx_requests = 0; i < num_buffers; i++)
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expected_rtx_requests += i;
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g_object_get (rtxsend, "num-rtx-requests", &real_rtx_requests,
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"num-rtx-packets", &real_rtx_packets, NULL);
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fail_unless_equals_int (expected_rtx_requests, real_rtx_requests);
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fail_unless_equals_int (expected_rtx_packets, real_rtx_packets);
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/* and the number of actual buffers that we were pushed out of rtxsend */
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fail_unless_equals_int (g_list_length (buffers),
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num_buffers + expected_rtx_packets);
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node = buffers;
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for (i = 1; i <= num_buffers; i++) {
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/* verify the normal rtp flow packet */
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res = gst_rtp_buffer_map (GST_BUFFER (node->data), GST_MAP_READ, &rtp);
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fail_unless_equals_int (res, TRUE);
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fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
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fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&rtp),
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payload_type);
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fail_unless_equals_int (gst_rtp_buffer_get_seq (&rtp), i);
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gst_rtp_buffer_unmap (&rtp);
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node = g_list_next (node);
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/* there are no rtx packets after the last normal one */
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if (i == num_buffers)
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break;
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/* now verify the retransmission packets */
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for (j = i; j > MAX (i - half_buffers, 0); j--) {
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GST_INFO ("checking %d, %d", i, j);
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res = gst_rtp_buffer_map (GST_BUFFER (node->data), GST_MAP_READ, &rtp);
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fail_unless_equals_int (res, TRUE);
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fail_if (gst_rtp_buffer_get_ssrc (&rtp) == ssrc);
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fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&rtp),
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rtx_payload_type);
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fail_unless_equals_int (GST_READ_UINT16_BE (gst_rtp_buffer_get_payload (&rtp)), j); /* j == rtx seqnum */
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/* open the original packet for this rtx packet and verify timestamps */
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res = gst_rtp_buffer_map (GST_BUFFER (g_list_nth_data (in_buffers, j)),
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GST_MAP_READ, &orig_rtp);
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fail_unless_equals_int (res, TRUE);
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fail_unless_equals_int (gst_rtp_buffer_get_timestamp (&orig_rtp),
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gst_rtp_buffer_get_timestamp (&rtp));
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gst_rtp_buffer_unmap (&orig_rtp);
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gst_rtp_buffer_unmap (&rtp);
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node = g_list_next (node);
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}
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}
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}
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g_list_free_full (in_buffers, (GDestroyNotify) gst_buffer_unref);
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gst_check_drop_buffers ();
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gst_check_teardown_src_pad (rtxsend);
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gst_check_teardown_sink_pad (rtxsend);
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gst_check_teardown_element (rtxsend);
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}
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GST_END_TEST;
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static Suite *
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rtprtx_suite (void)
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{
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Suite *s = suite_create ("rtprtx");
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TCase *tc_chain = tcase_create ("general");
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tcase_set_timeout (tc_chain, 10000);
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tcase_set_timeout (tc_chain, 10);
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_push_forward_seq);
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tcase_add_test (tc_chain, test_drop_one_sender);
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tcase_add_test (tc_chain, test_drop_multiple_sender);
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tcase_add_test (tc_chain, test_rtxsender_packet_retention);
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return s;
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}
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