mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
tests/webrtc: test for enabled bundled fec/rtx
Doesn't actually check that any fec/rtx happens, just that the pipeline is vaguely sane and doesn't error. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters
GStreamer Marge Bot
parent
699739c130
commit
75b23d646a
1 changed files with 172 additions and 4 deletions
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@ -29,6 +29,7 @@
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#include <gst/webrtc/webrtc.h>
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#include <gst/rtp/rtp.h>
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#include "../../../ext/webrtc/webrtcsdp.h"
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#include "../../../ext/webrtc/webrtcsdp.c"
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#include "../../../ext/webrtc/utils.h"
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@ -4227,7 +4228,7 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element,
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GstPad * pad, gpointer user_data)
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{
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GstHarness *h;
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GstHarness **sink_harness = user_data;
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GList **sink_harness = user_data;
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if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
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return;
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@ -4236,7 +4237,7 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element,
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t->harnesses = g_list_prepend (t->harnesses, h);
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if (sink_harness) {
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*sink_harness = h;
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*sink_harness = g_list_prepend (*sink_harness, h);
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g_cond_broadcast (&t->cond);
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}
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}
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@ -4273,6 +4274,7 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
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VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
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GUINT_TO_POINTER (0), &count);
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GstHarness *h;
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GList *sink_harnesses = NULL;
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GstHarness *sink_harness = NULL;
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guint i;
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GstElement *rtpbin2;
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@ -4281,7 +4283,7 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
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t->on_negotiation_needed = NULL;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_harness;
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t->pad_added_data = &sink_harness;
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t->pad_added_data = &sink_harnesses;
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rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
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fail_unless (rtpbin2 != NULL);
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@ -4304,10 +4306,12 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
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gst_harness_push_from_src (h);
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g_mutex_lock (&t->lock);
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while (sink_harness == NULL) {
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while (sink_harnesses == NULL) {
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gst_harness_push_from_src (h);
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g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
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}
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fail_unless_equals_int (1, g_list_length (sink_harnesses));
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sink_harness = (GstHarness *) sink_harnesses->data;
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g_mutex_unlock (&t->lock);
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fail_unless (sink_harness->element == t->webrtc2);
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@ -4340,6 +4344,8 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
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}
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test_webrtc_free (t);
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g_list_free (sink_harnesses);
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}
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GST_END_TEST;
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@ -4560,6 +4566,167 @@ GST_START_TEST (test_bundle_mid_header_extension)
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GST_END_TEST;
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static void
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on_new_transceiver_set_rtx_fec (GstElement * webrtcbin, GObject * trans,
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gpointer user_data)
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{
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g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
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"fec-percentage", 100, "do-nack", TRUE, NULL);
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}
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GST_START_TEST (test_max_bundle_fec)
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{
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struct test_webrtc *t = test_webrtc_new ();
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guint media_format_count[] = { 5, 5, };
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VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
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media_format_count, NULL);
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VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL,
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&media_formats);
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VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
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VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
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GUINT_TO_POINTER (1), &count);
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VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
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GUINT_TO_POINTER (2), &count);
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const gchar *expected_offer_setup[] = { "actpass", "actpass", };
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VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
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&offer_non_reject);
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const gchar *expected_answer_setup[] = { "active", "active", };
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VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
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&answer_non_reject);
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const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", };
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VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
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&offer_setup);
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const gchar *expected_answer_direction[] = { "recvonly", "recvonly", };
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VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
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&answer_setup);
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GstHarness *src0, *src1;
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GList *sink_harnesses = NULL;
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guint i;
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GstElement *rtpbin2;
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GstBuffer *buf;
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guint ssrcs[] = { 123456789, 987654321 };
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GArray *ssrcs_received;
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t->on_negotiation_needed = NULL;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_harness;
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t->pad_added_data = &sink_harnesses;
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gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
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"max-bundle");
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gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
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"max-bundle");
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rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
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fail_unless (rtpbin2 != NULL);
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g_signal_connect (rtpbin2, "new-jitterbuffer",
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G_CALLBACK (new_jitterbuffer_set_fast_start), NULL);
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g_object_unref (rtpbin2);
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g_signal_connect (t->webrtc1, "on-new-transceiver",
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G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
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g_signal_connect (t->webrtc2, "on-new-transceiver",
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G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
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src0 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
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add_audio_test_src_harness (src0, ssrcs[0]);
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t->harnesses = g_list_prepend (t->harnesses, src0);
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src1 = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
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add_audio_test_src_harness (src1, ssrcs[1]);
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t->harnesses = g_list_prepend (t->harnesses, src1);
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test_validate_sdp (t, &offer, &answer);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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for (i = 0; i < 10; i++) {
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gst_harness_push_from_src (src0);
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gst_harness_push_from_src (src1);
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}
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ssrcs_received = g_array_new (FALSE, TRUE, sizeof (guint32));
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/* Get one buffer out for each ssrc sent.
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*/
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g_mutex_lock (&t->lock);
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while (ssrcs_received->len < G_N_ELEMENTS (ssrcs)) {
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GList *l;
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guint i;
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gst_harness_push_from_src (src0);
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gst_harness_push_from_src (src1);
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if (g_list_length (sink_harnesses) < 2) {
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g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
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if (g_list_length (sink_harnesses) < 2)
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continue;
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}
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for (l = sink_harnesses; l; l = l->next) {
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GstHarness *sink_harness = (GstHarness *) l->data;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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GstWebRTCRTPTransceiver *rtp_trans;
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char *trans_mid;
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GstPad *srcpad;
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guint ssrc;
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guint mlineindex;
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char *expected_mid;
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fail_unless (sink_harness->element == t->webrtc2);
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buf = gst_harness_try_pull (sink_harness);
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if (!buf)
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continue;
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/* ensure that the resulting pad has the correct mid set */
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srcpad = gst_pad_get_peer (sink_harness->sinkpad);
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fail_unless (srcpad != NULL);
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g_object_get (srcpad, "transceiver", &rtp_trans, NULL);
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gst_clear_object (&srcpad);
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fail_unless (rtp_trans);
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g_object_get (rtp_trans, "mid", &trans_mid, "mlineindex", &mlineindex,
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NULL);
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gst_clear_object (&rtp_trans);
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expected_mid = g_strdup_printf ("audio%u", mlineindex);
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fail_unless (trans_mid != NULL);
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fail_unless_equals_string (trans_mid, expected_mid);
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g_clear_pointer (&trans_mid, g_free);
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g_clear_pointer (&expected_mid, g_free);
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fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp));
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ssrc = gst_rtp_buffer_get_ssrc (&rtp);
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for (i = 0; i < ssrcs_received->len; i++) {
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if (g_array_index (ssrcs_received, guint, i) == ssrc)
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break;
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}
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if (i == ssrcs_received->len) {
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g_array_append_val (ssrcs_received, ssrc);
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}
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gst_rtp_buffer_unmap (&rtp);
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gst_buffer_unref (buf);
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}
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}
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g_mutex_unlock (&t->lock);
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GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc1), GST_DEBUG_GRAPH_SHOW_ALL,
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"webrtc1-fec-final");
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GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc2), GST_DEBUG_GRAPH_SHOW_ALL,
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"webrtc2-fec-final");
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test_webrtc_free (t);
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g_list_free (sink_harnesses);
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g_array_unref (ssrcs_received);
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}
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GST_END_TEST;
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static Suite *
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webrtcbin_suite (void)
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{
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@ -4616,6 +4783,7 @@ webrtcbin_suite (void)
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tcase_add_test (tc, test_codec_preferences_invalid_extmap);
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tcase_add_test (tc, test_renego_rtx);
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tcase_add_test (tc, test_bundle_mid_header_extension);
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tcase_add_test (tc, test_max_bundle_fec);
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if (sctpenc && sctpdec) {
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tcase_add_test (tc, test_data_channel_create);
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tcase_add_test (tc, test_data_channel_remote_notify);
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