From: Matthew Waters Date: Tue, 16 Nov 2021 07:11:49 +0000 (+1100) Subject: tests/webrtc: test for enabled bundled fec/rtx X-Git-Tag: 1.22.0~1979 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=75b23d646ad19e899db0a13ad616923c7fe07fc8;p=platform%2Fupstream%2Fgstreamer.git 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: --- diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c index d3043da..279496e 100644 --- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c +++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c @@ -29,6 +29,7 @@ #include #include #include +#include #include "../../../ext/webrtc/webrtcsdp.h" #include "../../../ext/webrtc/webrtcsdp.c" #include "../../../ext/webrtc/utils.h" @@ -4227,7 +4228,7 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element, GstPad * pad, gpointer user_data) { GstHarness *h; - GstHarness **sink_harness = user_data; + GList **sink_harness = user_data; if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC) return; @@ -4236,7 +4237,7 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element, t->harnesses = g_list_prepend (t->harnesses, h); if (sink_harness) { - *sink_harness = h; + *sink_harness = g_list_prepend (*sink_harness, h); g_cond_broadcast (&t->cond); } } @@ -4273,6 +4274,7 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad) VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media, GUINT_TO_POINTER (0), &count); GstHarness *h; + GList *sink_harnesses = NULL; GstHarness *sink_harness = NULL; guint i; GstElement *rtpbin2; @@ -4281,7 +4283,7 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad) t->on_negotiation_needed = NULL; t->on_ice_candidate = NULL; t->on_pad_added = _pad_added_harness; - t->pad_added_data = &sink_harness; + t->pad_added_data = &sink_harnesses; rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin"); fail_unless (rtpbin2 != NULL); @@ -4304,10 +4306,12 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad) gst_harness_push_from_src (h); g_mutex_lock (&t->lock); - while (sink_harness == NULL) { + while (sink_harnesses == NULL) { gst_harness_push_from_src (h); g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000); } + fail_unless_equals_int (1, g_list_length (sink_harnesses)); + sink_harness = (GstHarness *) sink_harnesses->data; g_mutex_unlock (&t->lock); fail_unless (sink_harness->element == t->webrtc2); @@ -4340,6 +4344,8 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad) } test_webrtc_free (t); + + g_list_free (sink_harnesses); } GST_END_TEST; @@ -4560,6 +4566,167 @@ GST_START_TEST (test_bundle_mid_header_extension) GST_END_TEST; +static void +on_new_transceiver_set_rtx_fec (GstElement * webrtcbin, GObject * trans, + gpointer user_data) +{ + g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, + "fec-percentage", 100, "do-nack", TRUE, NULL); +} + +GST_START_TEST (test_max_bundle_fec) +{ + struct test_webrtc *t = test_webrtc_new (); + guint media_format_count[] = { 5, 5, }; + VAL_SDP_INIT (media_formats, on_sdp_media_count_formats, + media_format_count, NULL); + VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, + &media_formats); + VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads); + VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media, + GUINT_TO_POINTER (1), &count); + VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media, + GUINT_TO_POINTER (2), &count); + const gchar *expected_offer_setup[] = { "actpass", "actpass", }; + VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, + &offer_non_reject); + const gchar *expected_answer_setup[] = { "active", "active", }; + VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, + &answer_non_reject); + const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", }; + VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction, + &offer_setup); + const gchar *expected_answer_direction[] = { "recvonly", "recvonly", }; + VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction, + &answer_setup); + GstHarness *src0, *src1; + GList *sink_harnesses = NULL; + guint i; + GstElement *rtpbin2; + GstBuffer *buf; + guint ssrcs[] = { 123456789, 987654321 }; + GArray *ssrcs_received; + + t->on_negotiation_needed = NULL; + t->on_ice_candidate = NULL; + t->on_pad_added = _pad_added_harness; + t->pad_added_data = &sink_harnesses; + + gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy", + "max-bundle"); + gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy", + "max-bundle"); + + rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin"); + fail_unless (rtpbin2 != NULL); + g_signal_connect (rtpbin2, "new-jitterbuffer", + G_CALLBACK (new_jitterbuffer_set_fast_start), NULL); + g_object_unref (rtpbin2); + + g_signal_connect (t->webrtc1, "on-new-transceiver", + G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL); + g_signal_connect (t->webrtc2, "on-new-transceiver", + G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL); + + src0 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL); + add_audio_test_src_harness (src0, ssrcs[0]); + t->harnesses = g_list_prepend (t->harnesses, src0); + + src1 = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL); + add_audio_test_src_harness (src1, ssrcs[1]); + t->harnesses = g_list_prepend (t->harnesses, src1); + + test_validate_sdp (t, &offer, &answer); + + fail_if (gst_element_set_state (t->webrtc1, + GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE); + fail_if (gst_element_set_state (t->webrtc2, + GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE); + + for (i = 0; i < 10; i++) { + gst_harness_push_from_src (src0); + gst_harness_push_from_src (src1); + } + + ssrcs_received = g_array_new (FALSE, TRUE, sizeof (guint32)); + + /* Get one buffer out for each ssrc sent. + */ + g_mutex_lock (&t->lock); + while (ssrcs_received->len < G_N_ELEMENTS (ssrcs)) { + GList *l; + guint i; + + gst_harness_push_from_src (src0); + gst_harness_push_from_src (src1); + if (g_list_length (sink_harnesses) < 2) { + g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000); + if (g_list_length (sink_harnesses) < 2) + continue; + } + + for (l = sink_harnesses; l; l = l->next) { + GstHarness *sink_harness = (GstHarness *) l->data; + GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; + GstWebRTCRTPTransceiver *rtp_trans; + char *trans_mid; + GstPad *srcpad; + guint ssrc; + guint mlineindex; + char *expected_mid; + + fail_unless (sink_harness->element == t->webrtc2); + + buf = gst_harness_try_pull (sink_harness); + if (!buf) + continue; + + /* ensure that the resulting pad has the correct mid set */ + srcpad = gst_pad_get_peer (sink_harness->sinkpad); + fail_unless (srcpad != NULL); + g_object_get (srcpad, "transceiver", &rtp_trans, NULL); + gst_clear_object (&srcpad); + fail_unless (rtp_trans); + g_object_get (rtp_trans, "mid", &trans_mid, "mlineindex", &mlineindex, + NULL); + gst_clear_object (&rtp_trans); + expected_mid = g_strdup_printf ("audio%u", mlineindex); + fail_unless (trans_mid != NULL); + fail_unless_equals_string (trans_mid, expected_mid); + g_clear_pointer (&trans_mid, g_free); + g_clear_pointer (&expected_mid, g_free); + + fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp)); + + ssrc = gst_rtp_buffer_get_ssrc (&rtp); + for (i = 0; i < ssrcs_received->len; i++) { + if (g_array_index (ssrcs_received, guint, i) == ssrc) + break; + } + if (i == ssrcs_received->len) { + g_array_append_val (ssrcs_received, ssrc); + } + + gst_rtp_buffer_unmap (&rtp); + + gst_buffer_unref (buf); + } + } + g_mutex_unlock (&t->lock); + + GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc1), GST_DEBUG_GRAPH_SHOW_ALL, + "webrtc1-fec-final"); + GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc2), GST_DEBUG_GRAPH_SHOW_ALL, + "webrtc2-fec-final"); + + test_webrtc_free (t); + g_list_free (sink_harnesses); + + g_array_unref (ssrcs_received); +} + +GST_END_TEST; + static Suite * webrtcbin_suite (void) { @@ -4616,6 +4783,7 @@ webrtcbin_suite (void) tcase_add_test (tc, test_codec_preferences_invalid_extmap); tcase_add_test (tc, test_renego_rtx); tcase_add_test (tc, test_bundle_mid_header_extension); + tcase_add_test (tc, test_max_bundle_fec); if (sctpenc && sctpdec) { tcase_add_test (tc, test_data_channel_create); tcase_add_test (tc, test_data_channel_remote_notify);