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tests/webrtc: validate number of sdp media using validate_sdp

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This commit is contained in:
Matthew Waters 2018-11-26 19:54:18 +11:00
parent 7e1cdbfd4d
commit eb79f95bf8
1 changed files with 50 additions and 49 deletions
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99
tests/check/elements/webrtcbin.c
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@ -621,12 +621,32 @@ _pad_added_fakesink (struct test_webrtc *t, GstElement * element,
t->harnesses = g_list_prepend (t->harnesses, h);
}
static GstWebRTCSessionDescription *
static void
_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
guint expected = GPOINTER_TO_UINT (user_data);
fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), expected);
}
typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data);
struct validate_sdp;
struct validate_sdp
{
ValidateSDPFunc validate;
gpointer user_data;
struct validate_sdp *next;
};
static GstWebRTCSessionDescription *
_check_validate_sdp (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
struct validate_sdp *validate = user_data;
GstWebRTCSessionDescription *offer = NULL;
guint expected = GPOINTER_TO_UINT (user_data);
const GstStructure *reply;
const gchar *field;
@ -636,7 +656,10 @@ _count_num_sdp_media (struct test_webrtc *t, GstElement * element,
gst_structure_get (reply, field,
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
fail_unless_equals_int (gst_sdp_message_medias_len (offer->sdp), expected);
while (validate) {
validate->validate (t, element, offer, validate->user_data);
validate = validate->next;
}
return offer;
}
@ -644,14 +667,18 @@ _count_num_sdp_media (struct test_webrtc *t, GstElement * element,
GST_START_TEST (test_sdp_no_media)
{
struct test_webrtc *t = test_webrtc_new ();
struct validate_sdp offer =
{ _count_num_sdp_media, GUINT_TO_POINTER (0), NULL };
struct validate_sdp answer =
{ _count_num_sdp_media, GUINT_TO_POINTER (0), NULL };
/* check that a no stream connection creates 0 media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (0);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (0);
t->on_answer_created = _count_num_sdp_media;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
@ -706,15 +733,19 @@ create_audio_test (void)
GST_START_TEST (test_audio)
{
struct test_webrtc *t = create_audio_test ();
struct validate_sdp offer =
{ _count_num_sdp_media, GUINT_TO_POINTER (1), NULL };
struct validate_sdp answer =
{ _count_num_sdp_media, GUINT_TO_POINTER (1), NULL };
/* check that a single stream connection creates the associated number
* of media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (1);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (1);
t->on_answer_created = _count_num_sdp_media;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
@ -754,15 +785,19 @@ create_audio_video_test (void)
GST_START_TEST (test_audio_video)
{
struct test_webrtc *t = create_audio_video_test ();
struct validate_sdp offer =
{ _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
struct validate_sdp answer =
{ _count_num_sdp_media, GUINT_TO_POINTER (2), NULL };
/* check that a dual stream connection creates the associated number
* of media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (2);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (2);
t->on_answer_created = _count_num_sdp_media;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
@ -779,40 +814,6 @@ GST_START_TEST (test_audio_video)
GST_END_TEST;
typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data);
struct validate_sdp;
struct validate_sdp
{
ValidateSDPFunc validate;
gpointer user_data;
struct validate_sdp *next;
};
static GstWebRTCSessionDescription *
_check_validate_sdp (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
struct validate_sdp *validate = user_data;
GstWebRTCSessionDescription *offer = NULL;
const GstStructure *reply;
const gchar *field;
field = t->offerror == 1 && t->webrtc1 == element ? "offer" : "answer";
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, field,
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
while (validate) {
validate->validate (t, element, offer, validate->user_data);
validate = validate->next;
}
return offer;
}
static void
on_sdp_media_direction (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)