mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-10-05 10:12:20 +00:00
Code Issues 1 Projects Releases Wiki Activity

webrtcbin test: Add for the case where a second m-line is renegotiated

Browse source 
This is for the case where there answerer forces a specific media type
for a m-line, but he origin offer only has the other media type. In this
case, we will create a second transceiver on receiving the offer and add
the desired media type using renegotiation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-31 11:41:45 -04:00
parent c7107fd940
commit 4a48e291ff
1 changed files with 102 additions and 9 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

111
tests/check/elements/webrtcbin.c
View file

@ -37,7 +37,7 @@
#define OPUS_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=OPUS,media=audio,clock-rate=48000,ssrc=(uint)3384078950"
#define VP8_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=VP8,media=video,clock-rate=90000,ssrc=(uint)3484078950"
#define TEST_IS_OFFER_ELEMENT(t, e) ((t)->offerror == 1 && (e) == (t)->webrtc1 ? TRUE : FALSE)
#define TEST_IS_OFFER_ELEMENT(t, e) ((((t)->offerror == 1 && (e) == (t)->webrtc1) || ((t)->offerror == 2 && (e) == (t)->webrtc2)) ? TRUE : FALSE)
#define TEST_GET_OFFEROR(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc1 : t->webrtc2)
#define TEST_GET_ANSWERER(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc2 : t->webrtc1)
@ -531,6 +531,7 @@ test_webrtc_new (void)
ret->on_answer_created = _offer_answer_not_reached;
ret->on_data_channel = _on_data_channel_not_reached;
ret->bus_message = _bus_no_errors;
ret->offerror = 1;
g_mutex_init (&ret->lock);
g_cond_init (&ret->cond);
@ -661,13 +662,13 @@ test_webrtc_free (struct test_webrtc *t)
}
static void
test_webrtc_create_offer (struct test_webrtc *t, GstElement * webrtc)
test_webrtc_create_offer (struct test_webrtc *t)
{
GstPromise *promise;
GstElement *offeror = TEST_GET_OFFEROR (t);
t->offerror = webrtc == t->webrtc1 ? 1 : 2;
promise = gst_promise_new_with_change_func (_on_offer_received, t, NULL);
g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
g_signal_emit_by_name (offeror, "create-offer", NULL, promise);
}
static void
@ -748,7 +749,7 @@ on_negotiation_needed_hit (struct test_webrtc *t, GstElement * element,
{
guint *flag = (guint *) user_data;
*flag = 1;
*flag |= 1 << ((element == t->webrtc1) ? 1 : 2);
}
typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
@ -772,7 +773,7 @@ _check_validate_sdp (struct test_webrtc *t, GstElement * element,
struct validate_sdp *validate = user_data;
GstWebRTCSessionDescription *desc = NULL;
if (t->offerror == 1 && t->webrtc1 == element)
if (TEST_IS_OFFER_ELEMENT (t, element))
desc = t->offer_desc;
else
desc = t->answer_desc;
@ -810,7 +811,7 @@ test_validate_sdp_full (struct test_webrtc *t, struct validate_sdp *offer,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
}
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_create_offer (t);
if (wait_mask == 0) {
test_webrtc_wait_for_answer_error_eos (t);
@ -2564,7 +2565,7 @@ GST_START_TEST (test_duplicate_nego)
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
fail_unless_equals_int (negotiation_flag, 1);
fail_unless (negotiation_flag & (1 << 2));
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
@ -3297,7 +3298,7 @@ GST_START_TEST (test_renego_lose_media_fails)
t->on_offer_set = offer_set_produced_error;
t->on_answer_created = NULL;
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_create_offer (t);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
test_webrtc_free (t);
@ -3603,6 +3604,97 @@ GST_START_TEST (test_reject_set_description)
GST_END_TEST;
GST_START_TEST (test_force_second_media)
{
struct test_webrtc *t = test_webrtc_new ();
const gchar *media_types[] = { "audio" };
VAL_SDP_INIT (media_type, _verify_media_types, &media_types, NULL);
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &media_type);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&media_formats);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&media_formats);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer_direction, on_sdp_media_direction,
expected_offer_direction, &offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer_direction, on_sdp_media_direction,
expected_answer_direction, &answer_setup);
VAL_SDP_INIT (answer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&answer_direction);
VAL_SDP_INIT (offer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&offer_direction);
const gchar *second_media_types[] = { "audio", "video" };
VAL_SDP_INIT (second_media_type, _verify_media_types, &second_media_types,
NULL);
guint second_media_format_count[] = { 1, 1 };
VAL_SDP_INIT (second_media_formats, on_sdp_media_count_formats,
second_media_format_count, &second_media_type);
const gchar *second_expected_offer_setup[] = { "active", "actpass" };
VAL_SDP_INIT (second_offer_setup, on_sdp_media_setup,
second_expected_offer_setup, &second_media_formats);
const gchar *second_expected_answer_setup[] = { "passive", "active" };
VAL_SDP_INIT (second_answer_setup, on_sdp_media_setup,
second_expected_answer_setup, &second_media_formats);
const gchar *second_expected_answer_direction[] = { "sendonly", "recvonly" };
VAL_SDP_INIT (second_answer_direction, on_sdp_media_direction,
second_expected_answer_direction, &second_answer_setup);
const gchar *second_expected_offer_direction[] = { "recvonly", "sendrecv" };
VAL_SDP_INIT (second_offer_direction, on_sdp_media_direction,
second_expected_offer_direction, &second_offer_setup);
VAL_SDP_INIT (second_answer_count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&second_answer_direction);
VAL_SDP_INIT (second_offer_count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&second_offer_direction);
GstHarness *h;
guint negotiation_flag = 0;
GstPadTemplate *templ;
GstCaps *caps;
GstPad *pad;
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
t->on_negotiation_needed = on_negotiation_needed_hit;
t->negotiation_data = &negotiation_flag;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
templ = gst_element_get_pad_template (t->webrtc2, "sink_%u");
fail_unless (templ != NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
pad = gst_element_request_pad (t->webrtc2, templ, NULL, caps);
gst_caps_unref (caps);
fail_unless (pad != NULL);
h = gst_harness_new_with_element (t->webrtc2, GST_PAD_NAME (pad), NULL);
gst_object_unref (pad);
add_fake_video_src_harness (h, 97);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer_count, &answer_count);
fail_unless (negotiation_flag & 1 << 2);
test_webrtc_reset_negotiation (t);
t->offerror = 2;
test_validate_sdp (t, &second_offer_count, &second_answer_count);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -3649,6 +3741,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_reject_request_pad);
tcase_add_test (tc, test_reject_create_offer);
tcase_add_test (tc, test_reject_set_description);
tcase_add_test (tc, test_force_second_media);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);