#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_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)
+
+#define TEST_SDP_IS_LOCAL(t, e, d) ((TEST_IS_OFFER_ELEMENT (t, e) ^ ((d)->type == GST_WEBRTC_SDP_TYPE_OFFER)) == 0)
+
typedef enum
{
STATE_NEW,
STATE_NEGOTIATION_NEEDED,
STATE_OFFER_CREATED,
+ STATE_OFFER_SET,
STATE_ANSWER_CREATED,
+ STATE_ANSWER_SET,
STATE_EOS,
STATE_ERROR,
STATE_CUSTOM,
gpointer user_data);
gpointer ice_candidate_data;
GDestroyNotify ice_candidate_notify;
- GstWebRTCSessionDescription * (*on_offer_created) (struct test_webrtc * t,
- GstElement * element,
- GstPromise * promise,
- gpointer user_data);
+ void (*on_offer_created) (struct test_webrtc * t,
+ GstElement * element,
+ GstPromise * promise,
+ gpointer user_data);
+ GstWebRTCSessionDescription *offer_desc;
+ guint offer_set_count;
gpointer offer_data;
GDestroyNotify offer_notify;
- GstWebRTCSessionDescription * (*on_answer_created) (struct test_webrtc * t,
- GstElement * element,
- GstPromise * promise,
- gpointer user_data);
- gpointer data_channel_data;
- GDestroyNotify data_channel_notify;
- void (*on_data_channel) (struct test_webrtc * t,
- GstElement * element,
- GObject *data_channel,
- gpointer user_data);
+ void (*on_offer_set) (struct test_webrtc * t,
+ GstElement * element,
+ GstPromise * promise,
+ gpointer user_data);
+ gpointer offer_set_data;
+ GDestroyNotify offer_set_notify;
+ void (*on_answer_created) (struct test_webrtc * t,
+ GstElement * element,
+ GstPromise * promise,
+ gpointer user_data);
+ GstWebRTCSessionDescription *answer_desc;
+ guint answer_set_count;
gpointer answer_data;
GDestroyNotify answer_notify;
+ void (*on_answer_set) (struct test_webrtc * t,
+ GstElement * element,
+ GstPromise * promise,
+ gpointer user_data);
+ gpointer answer_set_data;
+ GDestroyNotify answer_set_notify;
+ void (*on_data_channel) (struct test_webrtc * t,
+ GstElement * element,
+ GObject *data_channel,
+ gpointer user_data);
+ gpointer data_channel_data;
+ GDestroyNotify data_channel_notify;
void (*on_pad_added) (struct test_webrtc * t,
GstElement * element,
GstPad * pad,
};
static void
+test_webrtc_signal_state_unlocked (struct test_webrtc *t, TestState state)
+{
+ t->state = state;
+ g_cond_broadcast (&t->cond);
+}
+
+static void
+test_webrtc_signal_state (struct test_webrtc *t, TestState state)
+{
+ g_mutex_lock (&t->lock);
+ test_webrtc_signal_state_unlocked (t, state);
+ g_mutex_unlock (&t->lock);
+}
+
+static void
+_on_answer_set (GstPromise * promise, gpointer user_data)
+{
+ struct test_webrtc *t = user_data;
+ GstElement *answerer = TEST_GET_ANSWERER (t);
+
+ g_mutex_lock (&t->lock);
+ if (++t->answer_set_count >= 2 && t->on_answer_set) {
+ t->on_answer_set (t, answerer, promise, t->answer_set_data);
+ }
+ if (t->state == STATE_ANSWER_CREATED)
+ t->state = STATE_ANSWER_SET;
+ g_cond_broadcast (&t->cond);
+ gst_promise_unref (promise);
+ g_mutex_unlock (&t->lock);
+}
+
+static void
_on_answer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
- GstElement *offeror = t->offerror == 1 ? t->webrtc1 : t->webrtc2;
- GstElement *answerer = t->offerror == 2 ? t->webrtc1 : t->webrtc2;
+ GstElement *offeror = TEST_GET_OFFEROR (t);
+ GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *answer = NULL;
- gchar *desc;
+ GError *error = NULL;
reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, "answer",
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
- desc = gst_sdp_message_as_text (answer->sdp);
- GST_INFO ("Created Answer: %s", desc);
- g_free (desc);
+ if (gst_structure_get (reply, "answer",
+ GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL)) {
+ gchar *desc = gst_sdp_message_as_text (answer->sdp);
+ GST_INFO ("Created Answer: %s", desc);
+ g_free (desc);
+ } else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
+ GST_INFO ("Creating answer resulted in error: %s", error->message);
+ } else {
+ g_assert_not_reached ();
+ }
g_mutex_lock (&t->lock);
+
+ g_assert (t->answer_desc == NULL);
+ t->answer_desc = answer;
+
if (t->on_answer_created) {
- gst_webrtc_session_description_free (answer);
- answer = t->on_answer_created (t, answerer, promise, t->answer_data);
+ t->on_answer_created (t, answerer, promise, t->answer_data);
}
gst_promise_unref (promise);
- g_signal_emit_by_name (answerer, "set-local-description", answer, NULL);
- g_signal_emit_by_name (offeror, "set-remote-description", answer, NULL);
+ if (error)
+ goto error;
- t->state = STATE_ANSWER_CREATED;
- g_cond_broadcast (&t->cond);
+ if (t->answer_desc) {
+ promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
+ g_signal_emit_by_name (answerer, "set-local-description", t->answer_desc,
+ promise);
+ promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
+ g_signal_emit_by_name (offeror, "set-remote-description", t->answer_desc,
+ promise);
+ }
+
+ test_webrtc_signal_state_unlocked (t, STATE_ANSWER_CREATED);
+ g_mutex_unlock (&t->lock);
+ return;
+
+error:
+ g_clear_error (&error);
+ if (t->state < STATE_ERROR)
+ test_webrtc_signal_state_unlocked (t, STATE_ERROR);
g_mutex_unlock (&t->lock);
+ return;
+}
+
+static void
+_on_offer_set (GstPromise * promise, gpointer user_data)
+{
+ struct test_webrtc *t = user_data;
+ GstElement *offeror = TEST_GET_OFFEROR (t);
- gst_webrtc_session_description_free (answer);
+ g_mutex_lock (&t->lock);
+ if (++t->offer_set_count >= 2 && t->on_offer_set) {
+ t->on_offer_set (t, offeror, promise, t->offer_set_data);
+ }
+ if (t->state == STATE_OFFER_CREATED)
+ t->state = STATE_OFFER_SET;
+ g_cond_broadcast (&t->cond);
+ gst_promise_unref (promise);
+ g_mutex_unlock (&t->lock);
}
static void
_on_offer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
- GstElement *offeror = t->offerror == 1 ? t->webrtc1 : t->webrtc2;
- GstElement *answerer = t->offerror == 2 ? t->webrtc1 : t->webrtc2;
+ GstElement *offeror = TEST_GET_OFFEROR (t);
+ GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *offer = NULL;
- gchar *desc;
+ GError *error = NULL;
reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, "offer",
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
- desc = gst_sdp_message_as_text (offer->sdp);
- GST_INFO ("Created offer: %s", desc);
- g_free (desc);
+ if (gst_structure_get (reply, "offer",
+ GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL)) {
+ gchar *desc = gst_sdp_message_as_text (offer->sdp);
+ GST_INFO ("Created offer: %s", desc);
+ g_free (desc);
+ } else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
+ GST_INFO ("Creating offer resulted in error: %s", error->message);
+ } else {
+ g_assert_not_reached ();
+ }
g_mutex_lock (&t->lock);
+
+ g_assert (t->offer_desc == NULL);
+ t->offer_desc = offer;
+
if (t->on_offer_created) {
- gst_webrtc_session_description_free (offer);
- offer = t->on_offer_created (t, offeror, promise, t->offer_data);
+ t->on_offer_created (t, offeror, promise, t->offer_data);
}
gst_promise_unref (promise);
- g_signal_emit_by_name (offeror, "set-local-description", offer, NULL);
- g_signal_emit_by_name (answerer, "set-remote-description", offer, NULL);
+ if (error)
+ goto error;
- promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
- g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
+ if (t->offer_desc) {
+ promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
+ g_signal_emit_by_name (offeror, "set-local-description", t->offer_desc,
+ promise);
+ promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
+ g_signal_emit_by_name (answerer, "set-remote-description", t->offer_desc,
+ promise);
- t->state = STATE_OFFER_CREATED;
- g_cond_broadcast (&t->cond);
+ promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
+ g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
+ }
+
+ test_webrtc_signal_state_unlocked (t, STATE_OFFER_CREATED);
g_mutex_unlock (&t->lock);
+ return;
- gst_webrtc_session_description_free (offer);
+error:
+ g_clear_error (&error);
+ if (t->state < STATE_ERROR)
+ test_webrtc_signal_state_unlocked (t, STATE_ERROR);
+ g_mutex_unlock (&t->lock);
+ return;
}
static gboolean
}
gst_message_parse_error (msg, &err, &dbg_info);
- GST_WARNING ("ERROR from element %s: %s\n",
+ GST_WARNING ("ERROR from element %s: %s",
GST_OBJECT_NAME (msg->src), err->message);
- GST_WARNING ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
+ GST_WARNING ("Debugging info: %s", (dbg_info) ? dbg_info : "none");
g_error_free (err);
g_free (dbg_info);
- t->state = STATE_ERROR;
- g_cond_broadcast (&t->cond);
+ test_webrtc_signal_state_unlocked (t, STATE_ERROR);
break;
}
case GST_MESSAGE_EOS:{
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
- GST_INFO ("EOS received\n");
- t->state = STATE_EOS;
- g_cond_broadcast (&t->cond);
+ GST_INFO ("EOS received");
+ test_webrtc_signal_state_unlocked (t, STATE_EOS);
break;
}
default:
}
}
-static GstWebRTCSessionDescription *
+static void
_offer_answer_not_reached (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
g_mutex_init (&ret->lock);
g_cond_init (&ret->cond);
+ ret->thread = g_thread_new ("test-webrtc", (GThreadFunc) _bus_thread, ret);
+
+ g_mutex_lock (&ret->lock);
+ while (!ret->loop)
+ g_cond_wait (&ret->cond, &ret->lock);
+ g_mutex_unlock (&ret->lock);
+
ret->bus1 = gst_bus_new ();
ret->bus2 = gst_bus_new ();
gst_bus_add_watch (ret->bus1, (GstBusFunc) _bus_watch, ret);
g_signal_connect_swapped (ret->webrtc2, "notify::ice-connection-state",
G_CALLBACK (_broadcast), ret);
- ret->thread = g_thread_new ("test-webrtc", (GThreadFunc) _bus_thread, ret);
+ return ret;
+}
- g_mutex_lock (&ret->lock);
- while (!ret->loop)
- g_cond_wait (&ret->cond, &ret->lock);
- g_mutex_unlock (&ret->lock);
+static void
+test_webrtc_reset_negotiation (struct test_webrtc *t)
+{
+ if (t->offer_desc)
+ gst_webrtc_session_description_free (t->offer_desc);
+ t->offer_desc = NULL;
+ t->offer_set_count = 0;
+ if (t->answer_desc)
+ gst_webrtc_session_description_free (t->answer_desc);
+ t->answer_desc = NULL;
+ t->answer_set_count = 0;
- return ret;
+ test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
}
static void
t->ice_candidate_notify (t->ice_candidate_data);
if (t->offer_notify)
t->offer_notify (t->offer_data);
+ if (t->offer_set_notify)
+ t->offer_set_notify (t->offer_set_data);
if (t->answer_notify)
t->answer_notify (t->answer_data);
+ if (t->answer_set_notify)
+ t->answer_set_notify (t->answer_set_data);
if (t->pad_added_notify)
t->pad_added_notify (t->pad_added_data);
if (t->data_channel_notify)
fail_unless_equals_int (GST_STATE_CHANGE_SUCCESS,
gst_element_set_state (t->webrtc2, GST_STATE_NULL));
+ test_webrtc_reset_negotiation (t);
+
gst_object_unref (t->webrtc1);
gst_object_unref (t->webrtc2);
test_webrtc_wait_for_answer_error_eos (struct test_webrtc *t)
{
TestState states = 0;
- states |= (1 << STATE_ANSWER_CREATED);
+ states |= (1 << STATE_ANSWER_SET);
states |= (1 << STATE_EOS);
states |= (1 << STATE_ERROR);
test_webrtc_wait_for_state_mask (t, states);
}
static void
-test_webrtc_signal_state_unlocked (struct test_webrtc *t, TestState state)
-{
- t->state = state;
- g_cond_broadcast (&t->cond);
-}
-
-static void
-test_webrtc_signal_state (struct test_webrtc *t, TestState state)
-{
- g_mutex_lock (&t->lock);
- test_webrtc_signal_state_unlocked (t, state);
- g_mutex_unlock (&t->lock);
-}
-
-#if 0
-static void
test_webrtc_wait_for_ice_gathering_complete (struct test_webrtc *t)
{
GstWebRTCICEGatheringState ice_state1, ice_state2;
g_mutex_unlock (&t->lock);
}
+#if 0
static void
test_webrtc_wait_for_ice_connection (struct test_webrtc *t,
GstWebRTCICEConnectionState states)
t->harnesses = g_list_prepend (t->harnesses, h);
}
-static GstWebRTCSessionDescription *
-_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
+static void
+on_negotiation_needed_hit (struct test_webrtc *t, GstElement * element,
+ gpointer user_data)
+{
+ guint *flag = (guint *) user_data;
+
+ *flag = 1;
+}
+
+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;
+};
+
+#define VAL_SDP_INIT(name,func,data,next) \
+ struct validate_sdp name = { func, data, next }
+
+static void
+_check_validate_sdp (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
- GstWebRTCSessionDescription *offer = NULL;
- guint expected = GPOINTER_TO_UINT (user_data);
- const GstStructure *reply;
- const gchar *field;
+ struct validate_sdp *validate = user_data;
+ GstWebRTCSessionDescription *desc = NULL;
- field = t->offerror == 1 && t->webrtc1 == element ? "offer" : "answer";
+ if (t->offerror == 1 && t->webrtc1 == element)
+ desc = t->offer_desc;
+ else
+ desc = t->answer_desc;
- reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, field,
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
+ while (validate) {
+ validate->validate (t, element, desc, validate->user_data);
+ validate = validate->next;
+ }
+}
+
+static void
+test_validate_sdp_full (struct test_webrtc *t, struct validate_sdp *offer,
+ struct validate_sdp *answer, TestState wait_mask,
+ gboolean perform_state_change)
+{
+ if (offer) {
+ t->offer_data = offer;
+ t->on_offer_created = _check_validate_sdp;
+ } else {
+ t->offer_data = NULL;
+ t->on_offer_created = NULL;
+ }
+ if (answer) {
+ t->answer_data = answer;
+ t->on_answer_created = _check_validate_sdp;
+ } else {
+ t->answer_data = NULL;
+ t->on_answer_created = NULL;
+ }
+
+ if (perform_state_change) {
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
+ }
+
+ test_webrtc_create_offer (t, t->webrtc1);
+
+ if (wait_mask == 0) {
+ test_webrtc_wait_for_answer_error_eos (t);
+ fail_unless (t->state == STATE_ANSWER_SET);
+ } else {
+ test_webrtc_wait_for_state_mask (t, wait_mask);
+ }
+}
+
+static void
+test_validate_sdp (struct test_webrtc *t, struct validate_sdp *offer,
+ struct validate_sdp *answer)
+{
+ test_validate_sdp_full (t, offer, answer, 0, TRUE);
+}
- fail_unless_equals_int (gst_sdp_message_medias_len (offer->sdp), expected);
+static void
+_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ guint expected = GPOINTER_TO_UINT (user_data);
- return offer;
+ fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), expected);
}
GST_START_TEST (test_sdp_no_media)
{
struct test_webrtc *t = test_webrtc_new ();
+ VAL_SDP_INIT (count, _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;
-
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
+ test_validate_sdp (t, &count, &count);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless (t->state == STATE_ANSWER_CREATED);
test_webrtc_free (t);
}
GST_END_TEST;
static void
+on_sdp_media_direction (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ gchar **expected_directions = user_data;
+ int i;
+
+ for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+
+ if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0
+ || g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
+ gboolean have_direction = FALSE;
+ int j;
+
+ for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
+ const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
+
+ if (g_strcmp0 (attr->key, "inactive") == 0) {
+ fail_unless (have_direction == FALSE,
+ "duplicate/multiple directions for media %u", j);
+ have_direction = TRUE;
+ fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
+ } else if (g_strcmp0 (attr->key, "sendonly") == 0) {
+ fail_unless (have_direction == FALSE,
+ "duplicate/multiple directions for media %u", j);
+ have_direction = TRUE;
+ fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
+ } else if (g_strcmp0 (attr->key, "recvonly") == 0) {
+ fail_unless (have_direction == FALSE,
+ "duplicate/multiple directions for media %u", j);
+ have_direction = TRUE;
+ fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
+ } else if (g_strcmp0 (attr->key, "sendrecv") == 0) {
+ fail_unless (have_direction == FALSE,
+ "duplicate/multiple directions for media %u", j);
+ have_direction = TRUE;
+ fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
+ }
+ }
+ fail_unless (have_direction, "no direction attribute in media %u", i);
+ }
+ }
+}
+
+static void
+on_sdp_media_no_duplicate_payloads (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ int i, j, k;
+
+ for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+
+ GArray *media_formats = g_array_new (FALSE, FALSE, sizeof (int));
+ for (j = 0; j < gst_sdp_media_formats_len (media); j++) {
+ int pt = atoi (gst_sdp_media_get_format (media, j));
+ for (k = 0; k < media_formats->len; k++) {
+ int val = g_array_index (media_formats, int, k);
+ if (pt == val)
+ fail ("found an unexpected duplicate payload type %u within media %u",
+ pt, i);
+ }
+ g_array_append_val (media_formats, pt);
+ }
+ g_array_free (media_formats, TRUE);
+ }
+}
+
+static void
+on_sdp_media_count_formats (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ guint *expected_n_media_formats = user_data;
+ int i;
+
+ for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+ fail_unless_equals_int (gst_sdp_media_formats_len (media),
+ expected_n_media_formats[i]);
+ }
+}
+
+static void
+on_sdp_media_setup (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ gchar **expected_setup = user_data;
+ int i;
+
+ for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+ gboolean have_setup = FALSE;
+ int j;
+
+ for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
+ const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
+
+ if (g_strcmp0 (attr->key, "setup") == 0) {
+ fail_unless (have_setup == FALSE,
+ "duplicate/multiple setup for media %u", j);
+ have_setup = TRUE;
+ fail_unless (g_strcmp0 (attr->value, expected_setup[i]) == 0);
+ }
+ }
+ fail_unless (have_setup, "no setup attribute in media %u", i);
+ }
+}
+
+static void
add_fake_audio_src_harness (GstHarness * h, gint pt)
{
GstCaps *caps = gst_caps_from_string (OPUS_RTP_CAPS (pt));
GstHarness *h;
t->on_negotiation_needed = NULL;
+ t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
GST_START_TEST (test_audio)
{
struct test_webrtc *t = create_audio_test ();
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
+ &media_formats);
+ const gchar *expected_offer_setup[] = { "actpass", };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_answer_setup[] = { "active", };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &count);
+ const gchar *expected_offer_direction[] = { "sendrecv", };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "recvonly", };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
/* 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->on_ice_candidate = NULL;
-
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
-static struct test_webrtc *
-create_audio_video_test (void)
+static void
+_check_ice_port_restriction (struct test_webrtc *t, GstElement * element,
+ guint mlineindex, gchar * candidate, GstElement * other, gpointer user_data)
{
- struct test_webrtc *t = test_webrtc_new ();
- GstHarness *h;
+ GRegex *regex;
+ GMatchInfo *match_info;
- t->on_negotiation_needed = NULL;
- t->on_pad_added = _pad_added_fakesink;
+ gchar *candidate_port;
+ gchar *candidate_protocol;
+ gchar *candidate_typ;
+ guint port_as_int;
+ guint peer_number;
- 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);
+ regex =
+ g_regex_new ("candidate:(\\d+) (1) (UDP|TCP) (\\d+) ([0-9.]+|[0-9a-f:]+)"
+ " (\\d+) typ ([a-z]+)", 0, 0, NULL);
- h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
- add_fake_video_src_harness (h, 97);
- t->harnesses = g_list_prepend (t->harnesses, h);
+ g_regex_match (regex, candidate, 0, &match_info);
+ fail_unless (g_match_info_get_match_count (match_info) == 8, candidate);
- return t;
+ candidate_protocol = g_match_info_fetch (match_info, 2);
+ candidate_port = g_match_info_fetch (match_info, 6);
+ candidate_typ = g_match_info_fetch (match_info, 7);
+
+ peer_number = t->webrtc1 == element ? 1 : 2;
+
+ port_as_int = atoi (candidate_port);
+
+ if (!g_strcmp0 (candidate_typ, "host") && port_as_int != 9) {
+ guint expected_min = peer_number * 10000 + 1000;
+ guint expected_max = expected_min + 999;
+
+ fail_unless (port_as_int >= expected_min);
+ fail_unless (port_as_int <= expected_max);
+ }
+
+ g_free (candidate_port);
+ g_free (candidate_protocol);
+ g_free (candidate_typ);
+ g_match_info_free (match_info);
+ g_regex_unref (regex);
}
-GST_START_TEST (test_audio_video)
+GST_START_TEST (test_ice_port_restriction)
{
- struct test_webrtc *t = create_audio_video_test ();
+ struct test_webrtc *t = create_audio_test ();
+ GObject *webrtcice;
- /* check that a dual stream connection creates the associated number
- * of media sections */
+ VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
- 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->on_ice_candidate = NULL;
+ /*
+ * Ports are defined as follows "{peer}{protocol}000"
+ * - peer number: "1" for t->webrtc1, "2" for t->webrtc2
+ */
+ g_object_get (t->webrtc1, "ice-agent", &webrtcice, NULL);
+ g_object_set (webrtcice, "min-rtp-port", 11000, "max-rtp-port", 11999, NULL);
+ g_object_unref (webrtcice);
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
+ g_object_get (t->webrtc2, "ice-agent", &webrtcice, NULL);
+ g_object_set (webrtcice, "min-rtp-port", 21000, "max-rtp-port", 21999, NULL);
+ g_object_unref (webrtcice);
- test_webrtc_create_offer (t, t->webrtc1);
+ t->on_ice_candidate = _check_ice_port_restriction;
+ test_validate_sdp (t, &offer, &answer);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_webrtc_wait_for_ice_gathering_complete (t);
test_webrtc_free (t);
}
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)
+static struct test_webrtc *
+create_audio_video_test (void)
{
- 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);
+ struct test_webrtc *t = create_audio_test ();
+ GstHarness *h;
- while (validate) {
- validate->validate (t, element, offer, validate->user_data);
- validate = validate->next;
- }
+ h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
+ add_fake_video_src_harness (h, 97);
+ t->harnesses = g_list_prepend (t->harnesses, h);
- return offer;
+ return t;
}
-static void
-on_sdp_media_direction (struct test_webrtc *t, GstElement * element,
- GstWebRTCSessionDescription * desc, gpointer user_data)
+GST_START_TEST (test_audio_video)
{
- gchar **expected_directions = user_data;
- int i;
-
- for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
- const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
- gboolean have_direction = FALSE;
- int j;
+ struct test_webrtc *t = create_audio_video_test ();
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &count);
+ 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);
- for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
- const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
+ /* check that a dual stream connection creates the associated number
+ * of media sections */
- if (g_strcmp0 (attr->key, "inactive") == 0) {
- fail_unless (have_direction == FALSE,
- "duplicate/multiple directions for media %u", j);
- have_direction = TRUE;
- fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
- } else if (g_strcmp0 (attr->key, "sendonly") == 0) {
- fail_unless (have_direction == FALSE,
- "duplicate/multiple directions for media %u", j);
- have_direction = TRUE;
- fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
- } else if (g_strcmp0 (attr->key, "recvonly") == 0) {
- fail_unless (have_direction == FALSE,
- "duplicate/multiple directions for media %u", j);
- have_direction = TRUE;
- fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
- } else if (g_strcmp0 (attr->key, "sendrecv") == 0) {
- fail_unless (have_direction == FALSE,
- "duplicate/multiple directions for media %u", j);
- have_direction = TRUE;
- fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
- }
- }
- fail_unless (have_direction, "no direction attribute in media %u", j);
- }
+ test_validate_sdp (t, &offer, &answer);
+ test_webrtc_free (t);
}
+GST_END_TEST;
+
GST_START_TEST (test_media_direction)
{
struct test_webrtc *t = create_audio_video_test ();
- const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
- const gchar *expected_answer[] = { "sendrecv", "recvonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
- struct validate_sdp answer =
- { on_sdp_media_direction, expected_answer, NULL };
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &count);
+
+ 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[] = { "sendrecv", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
GstHarness *h;
/* check the default media directions for transceivers */
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
- t->on_negotiation_needed = NULL;
- 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,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
const GstSDPMedia *vmedia;
guint j;
- fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), 2);
-
vmedia = gst_sdp_message_get_media (desc->sdp, 1);
for (j = 0; j < gst_sdp_media_attributes_len (vmedia); j++) {
GST_START_TEST (test_payload_types)
{
struct test_webrtc *t = create_audio_video_test ();
- struct validate_sdp offer = { on_sdp_media_payload_types, NULL, NULL };
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1, 5, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (payloads, on_sdp_media_payload_types, NULL, &media_formats);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
- t->on_negotiation_needed = NULL;
- t->offer_data = &offer;
- t->on_offer_created = _check_validate_sdp;
- t->on_ice_candidate = NULL;
- /* We don't really care about the answer here */
- t->on_answer_created = NULL;
-
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless_equals_int (transceivers->len, 2);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
NULL);
g_array_unref (transceivers);
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
- test_webrtc_free (t);
-}
-
-GST_END_TEST;
-
-static void
-on_sdp_media_setup (struct test_webrtc *t, GstElement * element,
- GstWebRTCSessionDescription * desc, gpointer user_data)
-{
- gchar **expected_setup = user_data;
- int i;
-
- for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
- const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
- gboolean have_setup = FALSE;
- int j;
-
- for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
- const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
-
- if (g_strcmp0 (attr->key, "setup") == 0) {
- fail_unless (have_setup == FALSE,
- "duplicate/multiple setup for media %u", j);
- have_setup = TRUE;
- fail_unless (g_strcmp0 (attr->value, expected_setup[i]) == 0);
- }
- }
- fail_unless (have_setup, "no setup attribute in media %u", j);
- }
-}
-
-GST_START_TEST (test_media_setup)
-{
- struct test_webrtc *t = create_audio_test ();
- const gchar *expected_offer[] = { "actpass" };
- const gchar *expected_answer[] = { "active" };
- struct validate_sdp offer = { on_sdp_media_setup, expected_offer, NULL };
- struct validate_sdp answer = { on_sdp_media_setup, expected_answer, NULL };
-
- /* check the default dtls setup negotiation values */
-
- t->on_negotiation_needed = NULL;
- 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,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ /* We don't really care about the answer here */
+ test_validate_sdp (t, &offer, NULL);
test_webrtc_free (t);
}
GstPromise *p;
/* test that the stats generated without any streams are sane */
-
t->on_negotiation_needed = NULL;
- t->on_offer_created = NULL;
- t->on_answer_created = NULL;
-
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, NULL, NULL);
p = gst_promise_new_with_change_func (_on_stats, t, NULL);
g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
- const gchar *expected_offer[] = { "recvonly" };
- const gchar *expected_answer[] = { "sendonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
- struct validate_sdp answer =
- { on_sdp_media_direction, expected_answer, NULL };
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
+ &media_formats);
+ const gchar *expected_offer_setup[] = { "actpass", };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_answer_setup[] = { "active", };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &count);
+ const gchar *expected_offer_direction[] = { "recvonly", };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "sendonly", };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
GstCaps *caps;
GstHarness *h;
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
-
- t->on_negotiation_needed = NULL;
- t->on_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
- 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;
+ t->on_pad_added = _pad_added_fakesink;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
+ test_validate_sdp (t, &offer, &answer);
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
- const gchar *expected_offer[] = { "recvonly", "sendonly" };
- const gchar *expected_answer[] = { "sendonly", "recvonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
- struct validate_sdp answer =
- { on_sdp_media_direction, expected_answer, NULL };
+ VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+ NULL, NULL);
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, &no_duplicate_payloads);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &count);
+ const gchar *expected_offer_direction[] = { "recvonly", "sendonly" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "sendonly", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
GstCaps *caps;
GstHarness *h;
GArray *transceivers;
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
-
- t->on_negotiation_needed = NULL;
- t->on_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
- 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;
+ t->on_pad_added = _pad_added_fakesink;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
t->harnesses = g_list_prepend (t->harnesses, h);
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
+ fail_unless_equals_int (transceivers->len, 2);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
- fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
+ test_validate_sdp (t, &offer, &answer);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
gchar *label;
t->on_negotiation_needed = NULL;
- 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,
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- test_webrtc_create_offer (t, t->webrtc1);
+ test_validate_sdp (t, &offer, &offer);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
g_object_unref (channel);
g_free (label);
test_webrtc_free (t);
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
static void
on_message_string (GObject * channel, const gchar * str, struct test_webrtc *t)
{
- gchar *expected = g_object_steal_data (channel, "expected");
+ GstWebRTCDataChannelState state;
+ gchar *expected;
+
+ g_object_get (channel, "ready-state", &state, NULL);
+ fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
+
+ expected = g_object_steal_data (channel, "expected");
g_assert_cmpstr (expected, ==, str);
g_free (expected);
g_object_get (our, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
- g_object_get (other, "ready-state", &state, NULL);
- fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_set_data_full (our, "expected", g_strdup (test_string), g_free);
g_signal_connect (our, "on-message-string", G_CALLBACK (on_message_string),
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel_transfer_string;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
static void
on_message_data (GObject * channel, GBytes * data, struct test_webrtc *t)
{
- GBytes *expected = g_object_steal_data (channel, "expected");
+ GstWebRTCDataChannelState state;
+ GBytes *expected;
+
+ g_object_get (channel, "ready-state", &state, NULL);
+ fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
+
+ expected = g_object_steal_data (channel, "expected");
g_assert_cmpbytes (data, expected);
g_bytes_unref (expected);
g_object_get (our, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
- g_object_get (other, "ready-state", &state, NULL);
- fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_set_data_full (our, "expected", g_bytes_ref (data),
(GDestroyNotify) g_bytes_unref);
g_signal_connect (other, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
g_signal_emit_by_name (other, "send-data", data);
+ g_bytes_unref (data);
}
GST_START_TEST (test_data_channel_transfer_data)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel_transfer_data;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
&another);
g_assert_nonnull (another);
t->data_channel_data = another;
+ t->data_channel_notify = (GDestroyNotify) g_object_unref;
g_signal_connect (another, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
}
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel_create_data_channel;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
g_signal_connect (our, "on-error", G_CALLBACK (on_channel_error_not_reached),
NULL);
- g_signal_emit_by_name (our, "send-string", "DATA");
+ g_signal_emit_by_name (our, "send-string", "A");
}
GST_START_TEST (test_data_channel_low_threshold)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel_check_low_threshold_emitted;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
for (i = 0; i < size; i++)
random_data[i] = (guint8) (i & 0xff);
- data = g_bytes_new_static (random_data, size);
+ data = g_bytes_new_with_free_func (random_data, size,
+ (GDestroyNotify) g_free, random_data);
g_object_set_data_full (our, "expected", g_bytes_ref (data),
(GDestroyNotify) g_bytes_unref);
g_signal_connect (other, "on-error", G_CALLBACK (on_channel_error), t);
g_signal_emit_by_name (other, "send-data", data);
+ g_bytes_unref (data);
}
GST_START_TEST (test_data_channel_max_message_size)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
- 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;
t->on_data_channel = have_data_channel_transfer_large_data;
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+ test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
g_object_get (channel, "ready-state", &ready_state, NULL);
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
- if (++(*n_ready) >= 2)
+ if (g_atomic_int_add (n_ready, 1) >= 1) {
test_webrtc_signal_state (t, STATE_CUSTOM);
+ }
}
}
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel1 = NULL, *channel2 = NULL;
- struct validate_sdp offer = { on_sdp_has_datachannel, NULL, NULL };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL, NULL };
+ VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
GstStructure *s;
gint n_ready = 0;
t->on_negotiation_needed = NULL;
- 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,
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless (t->state == STATE_ANSWER_CREATED);
+ test_validate_sdp_full (t, &offer, &offer, 0, FALSE);
t->data_channel_data = &n_ready;
GST_END_TEST;
-typedef struct
+static void
+_count_non_rejected_media (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * sd, gpointer user_data)
{
- guint num_media;
- guint num_active_media;
- const gchar **bundled;
- const gchar **bundled_only;
-} BundleCheckData;
+ guint expected = GPOINTER_TO_UINT (user_data);
+ guint non_rejected_media;
+ guint i;
+
+ non_rejected_media = 0;
+
+ for (i = 0; i < gst_sdp_message_medias_len (sd->sdp); i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp, i);
+
+ if (gst_sdp_media_get_port (media) != 0)
+ non_rejected_media += 1;
+ }
+
+ fail_unless_equals_int (non_rejected_media, expected);
+}
static void
-_check_bundled_sdp_media (struct test_webrtc *t, GstElement * element,
+_check_bundle_tag (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
{
gchar **bundled = NULL;
- BundleCheckData *data = (BundleCheckData *) user_data;
+ GStrv expected = user_data;
guint i;
- guint active_media;
-
- fail_unless_equals_int (gst_sdp_message_medias_len (sd->sdp),
- data->num_media);
- fail_unless (_parse_bundle (sd->sdp, &bundled));
+ fail_unless (_parse_bundle (sd->sdp, &bundled, NULL));
if (!bundled) {
- fail_unless_equals_int (g_strv_length ((GStrv) data->bundled), 0);
+ fail_unless_equals_int (g_strv_length (expected), 0);
} else {
- fail_unless_equals_int (g_strv_length (bundled),
- g_strv_length ((GStrv) data->bundled));
+ fail_unless_equals_int (g_strv_length (bundled), g_strv_length (expected));
}
- for (i = 0; data->bundled[i]; i++) {
- fail_unless (g_strv_contains ((const gchar **) bundled, data->bundled[i]));
+ for (i = 0; i < g_strv_length (expected); i++) {
+ fail_unless (g_strv_contains ((const gchar **) bundled, expected[i]));
}
- active_media = 0;
+ g_strfreev (bundled);
+}
+
+static void
+_check_bundle_only_media (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * sd, gpointer user_data)
+{
+ gchar **expected_bundle_only = user_data;
+ guint i;
for (i = 0; i < gst_sdp_message_medias_len (sd->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp, i);
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
- if (g_strv_contains ((const gchar **) data->bundled_only, mid))
+ if (g_strv_contains ((const gchar **) expected_bundle_only, mid))
fail_unless (_media_has_attribute_key (media, "bundle-only"));
-
- if (gst_sdp_media_get_port (media) != 0)
- active_media += 1;
}
-
- fail_unless_equals_int (active_media, data->num_active_media);
-
- if (bundled)
- g_strfreev (bundled);
}
GST_START_TEST (test_bundle_audio_video_max_bundle_max_bundle)
const gchar *bundle[] = { "audio0", "video1", NULL };
const gchar *offer_bundle_only[] = { "video1", NULL };
const gchar *answer_bundle_only[] = { NULL };
+
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &payloads);
+ VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &bundle_tag);
+ VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (2), &bundle_tag);
+ VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
+ &offer_non_reject);
+ VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
+ &answer_non_reject);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &offer_bundle);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &answer_bundle);
+ 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);
+
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
- BundleCheckData offer_data = {
- 2,
- 1,
- bundle,
- offer_bundle_only,
- };
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-bundle");
/* We also set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
- BundleCheckData answer_data = {
- 2,
- 2,
- bundle,
- answer_bundle_only,
- };
- struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data, NULL };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data, NULL };
-
- 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");
- t->on_negotiation_needed = NULL;
- 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,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", NULL };
const gchar *bundle_only[] = { NULL };
+
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
+ VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (2), &bundle_tag);
+ VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
+ &count_non_reject);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &bundle_sdp);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &bundle_sdp);
+ 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);
+
/* We set a max-compat policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should *not* be marked
* as bundle-only
*/
- BundleCheckData offer_data = {
- 2,
- 2,
- bundle,
- bundle_only,
- };
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-compat");
/* We set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
- BundleCheckData answer_data = {
- 2,
- 2,
- bundle,
- bundle_only,
- };
- struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data, NULL };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data, NULL };
-
- gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
- "max-compat");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
- t->on_negotiation_needed = NULL;
- 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,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_START_TEST (test_bundle_audio_video_max_bundle_none)
{
struct test_webrtc *t = create_audio_video_test ();
- const gchar *offer_bundle[] = { "audio0", "video1", NULL };
+ const gchar *offer_mid[] = { "audio0", "video1", NULL };
const gchar *offer_bundle_only[] = { "video1", NULL };
- const gchar *answer_bundle[] = { NULL };
+ const gchar *answer_mid[] = { NULL };
const gchar *answer_bundle_only[] = { NULL };
+
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &payloads);
+ VAL_SDP_INIT (offer_bundle_tag, _check_bundle_tag, offer_mid,
+ &count_non_reject);
+ VAL_SDP_INIT (answer_bundle_tag, _check_bundle_tag, answer_mid,
+ &count_non_reject);
+ VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
+ &offer_bundle_tag);
+ VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
+ &answer_bundle_tag);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &offer_bundle);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &answer_bundle);
+ 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);
+
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
- BundleCheckData offer_data = {
- 2,
- 1,
- offer_bundle,
- offer_bundle_only,
- };
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-bundle");
/* We set a none policy on the answering webrtcbin,
* this means that the answer should contain no bundled
* medias, and as the bundle-policy of the offering webrtcbin
* is set to max-bundle, only one media should be active.
*/
- BundleCheckData answer_data = {
- 2,
- 1,
- answer_bundle,
- answer_bundle_only,
- };
- struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data, NULL };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data, NULL };
-
- gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
- "max-bundle");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy", "none");
- t->on_negotiation_needed = NULL;
- 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,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
- fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
-
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_START_TEST (test_bundle_audio_video_data)
{
struct test_webrtc *t = create_audio_video_test ();
- const gchar *bundle[] = { "audio0", "video1", "application2", NULL };
+ const gchar *mids[] = { "audio0", "video1", "application2", NULL };
const gchar *offer_bundle_only[] = { "video1", "application2", NULL };
const gchar *answer_bundle_only[] = { NULL };
GObject *channel = NULL;
+
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (3),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, mids, &payloads);
+ VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &bundle_tag);
+ VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (3), &bundle_tag);
+ VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
+ &offer_non_reject);
+ VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
+ &answer_non_reject);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &offer_bundle);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &answer_bundle);
+ const gchar *expected_offer_direction[] =
+ { "sendrecv", "sendrecv", "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] =
+ { "recvonly", "recvonly", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
- BundleCheckData offer_data = {
- 3,
- 1,
- bundle,
- offer_bundle_only,
- };
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-bundle");
/* We also set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
- BundleCheckData answer_data = {
- 3,
- 3,
- bundle,
- answer_bundle_only,
- };
- struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data, NULL };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data, NULL };
-
- 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");
- t->on_negotiation_needed = NULL;
- 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,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
- test_webrtc_create_offer (t, t->webrtc1);
-
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_validate_sdp (t, &offer, &answer);
g_object_unref (channel);
test_webrtc_free (t);
GST_START_TEST (test_duplicate_nego)
{
struct test_webrtc *t = create_audio_video_test ();
- const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
- const gchar *expected_answer[] = { "sendrecv", "recvonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
- struct validate_sdp answer =
- { on_sdp_media_direction, expected_answer, NULL };
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ 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[] = { "sendrecv", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
GstHarness *h;
+ guint negotiation_flag = 0;
/* check that negotiating twice succeeds */
+ t->on_negotiation_needed = on_negotiation_needed_hit;
+ t->negotiation_data = &negotiation_flag;
+
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ test_validate_sdp (t, &offer, &answer);
+ fail_unless_equals_int (negotiation_flag, 1);
+
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_dual_audio)
+{
+ struct test_webrtc *t = create_audio_test ();
+ guint media_format_count[] = { 1, 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ 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[] = { "sendrecv", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ GstHarness *h;
+ GstWebRTCRTPTransceiver *trans;
+ GArray *transceivers;
+
+ /* test that each mline gets a unique transceiver even with the same caps */
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ t->on_negotiation_needed = NULL;
+ test_validate_sdp (t, &offer, &answer);
+
+ g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
+ fail_unless (transceivers != NULL);
+ fail_unless_equals_int (2, transceivers->len);
+
+ trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
+ fail_unless (trans != NULL);
+ fail_unless_equals_int (trans->mline, 0);
+
+ trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
+ fail_unless (trans != NULL);
+ fail_unless_equals_int (trans->mline, 1);
+
+ g_array_unref (transceivers);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+static void
+sdp_increasing_session_version (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ GstWebRTCSessionDescription *previous;
+ const GstSDPOrigin *our_origin, *previous_origin;
+ const gchar *prop;
+ guint64 our_v, previous_v;
+
+ prop =
+ TEST_SDP_IS_LOCAL (t, element,
+ desc) ? "current-local-description" : "current-remote-description";
+ g_object_get (element, prop, &previous, NULL);
+
+ our_origin = gst_sdp_message_get_origin (desc->sdp);
+ previous_origin = gst_sdp_message_get_origin (previous->sdp);
+
+ our_v = g_ascii_strtoull (our_origin->sess_version, NULL, 10);
+ previous_v = g_ascii_strtoull (previous_origin->sess_version, NULL, 10);
+
+ ck_assert_int_lt (previous_v, our_v);
+
+ gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_equal_session_id (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ GstWebRTCSessionDescription *previous;
+ const GstSDPOrigin *our_origin, *previous_origin;
+ const gchar *prop;
+
+ prop =
+ TEST_SDP_IS_LOCAL (t, element,
+ desc) ? "current-local-description" : "current-remote-description";
+ g_object_get (element, prop, &previous, NULL);
+
+ our_origin = gst_sdp_message_get_origin (desc->sdp);
+ previous_origin = gst_sdp_message_get_origin (previous->sdp);
+
+ fail_unless_equals_string (previous_origin->sess_id, our_origin->sess_id);
+ gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_attribute (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, GstWebRTCSessionDescription * previous,
+ const gchar * attr)
+{
+ guint i, n;
+
+ n = MIN (gst_sdp_message_medias_len (previous->sdp),
+ gst_sdp_message_medias_len (desc->sdp));
+
+ for (i = 0; i < n; i++) {
+ const GstSDPMedia *our_media, *other_media;
+ const gchar *our_mid, *other_mid;
+
+ our_media = gst_sdp_message_get_media (desc->sdp, i);
+ other_media = gst_sdp_message_get_media (previous->sdp, i);
+
+ our_mid = gst_sdp_media_get_attribute_val (our_media, attr);
+ other_mid = gst_sdp_media_get_attribute_val (other_media, attr);
+
+ fail_unless_equals_string (our_mid, other_mid);
+ }
+}
+
+static void
+sdp_media_equal_mid (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ GstWebRTCSessionDescription *previous;
+ const gchar *prop;
+
+ prop =
+ TEST_SDP_IS_LOCAL (t, element,
+ desc) ? "current-local-description" : "current-remote-description";
+ g_object_get (element, prop, &previous, NULL);
+
+ sdp_media_equal_attribute (t, element, desc, previous, "mid");
+
+ gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_ice_params (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ GstWebRTCSessionDescription *previous;
+ const gchar *prop;
+
+ prop =
+ TEST_SDP_IS_LOCAL (t, element,
+ desc) ? "current-local-description" : "current-remote-description";
+ g_object_get (element, prop, &previous, NULL);
+
+ sdp_media_equal_attribute (t, element, desc, previous, "ice-ufrag");
+ sdp_media_equal_attribute (t, element, desc, previous, "ice-pwd");
+
+ gst_webrtc_session_description_free (previous);
+}
+
+static void
+sdp_media_equal_fingerprint (struct test_webrtc *t, GstElement * element,
+ GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+ GstWebRTCSessionDescription *previous;
+ const gchar *prop;
+
+ prop =
+ TEST_SDP_IS_LOCAL (t, element,
+ desc) ? "current-local-description" : "current-remote-description";
+ g_object_get (element, prop, &previous, NULL);
+
+ sdp_media_equal_attribute (t, element, desc, previous, "fingerprint");
+
+ gst_webrtc_session_description_free (previous);
+}
+
+GST_START_TEST (test_renego_add_stream)
+{
+ struct test_webrtc *t = create_audio_video_test ();
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] =
+ { "sendrecv", "sendrecv", "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] =
+ { "sendrecv", "recvonly", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ GstHarness *h;
+
+ /* negotiate an AV stream and then renegotiate an extra stream */
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ test_validate_sdp (t, &offer, &answer);
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+ add_fake_audio_src_harness (h, 98);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ media_formats.next = &renego_fingerprint;
+ count.user_data = GUINT_TO_POINTER (3);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_stream_add_data_channel)
+{
+ struct test_webrtc *t = create_audio_video_test ();
+
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", NULL };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "sendrecv", "recvonly", NULL };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ GObject *channel;
+ GstHarness *h;
+
+ /* negotiate an AV stream and then renegotiate a data channel */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
+ test_validate_sdp (t, &offer, &answer);
+
+ g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+ &channel);
+
+ media_formats.next = &renego_fingerprint;
+ count.user_data = GUINT_TO_POINTER (3);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ g_object_unref (channel);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_data_channel_add_stream)
+{
+ struct test_webrtc *t = test_webrtc_new ();
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] = { NULL, "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { NULL, "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ GObject *channel;
+ GstHarness *h;
+
+ /* negotiate an data channel and then renegotiate to add a av stream */
t->on_negotiation_needed = NULL;
- 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;
+ t->on_pad_added = _pad_added_fakesink;
fail_if (gst_element_set_state (t->webrtc1,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
- GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
- test_webrtc_create_offer (t, t->webrtc1);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+ &channel);
- test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
+ test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
+ add_fake_audio_src_harness (h, 97);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ media_formats.next = &renego_fingerprint;
+ count.user_data = GUINT_TO_POINTER (2);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
+
+ g_object_unref (channel);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_renego_stream_data_channel_add_stream)
+{
+ struct test_webrtc *t = test_webrtc_new ();
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] = { "sendrecv", NULL, "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "recvonly", NULL, "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ GObject *channel;
+ GstHarness *h;
+
+ /* Negotiate a stream and a data channel, then renogotiate with a new stream */
+ t->on_negotiation_needed = NULL;
+ t->on_ice_candidate = NULL;
+ t->on_pad_added = _pad_added_fakesink;
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 97);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ 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);
+
+ g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+ &channel);
+
+ test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+ add_fake_audio_src_harness (h, 97);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ media_formats.next = &renego_fingerprint;
+ count.user_data = GUINT_TO_POINTER (3);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
+
+ g_object_unref (channel);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bundle_renego_add_stream)
+{
+ struct test_webrtc *t = create_audio_video_test ();
+ const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+ const gchar *offer_bundle_only[] = { "video1", "audio2", NULL };
+ const gchar *answer_bundle_only[] = { NULL };
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] =
+ { "sendrecv", "sendrecv", "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] =
+ { "sendrecv", "recvonly", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, &payloads);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
+ VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &bundle_tag);
+ VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (3), &bundle_tag);
+ VAL_SDP_INIT (offer_bundle_only_sdp, _check_bundle_only_media,
+ &offer_bundle_only, &offer_non_reject);
+ VAL_SDP_INIT (answer_bundle_only_sdp, _check_bundle_only_media,
+ &answer_bundle_only, &answer_non_reject);
+ GstHarness *h;
+
+ /* We set a max-bundle policy on the offering webrtcbin,
+ * this means that all the offered medias should be part
+ * of the group:BUNDLE attribute, and they should be marked
+ * as bundle-only
+ */
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-bundle");
+ /* We also set a max-bundle policy on the answering webrtcbin,
+ * this means that all the offered medias should be part
+ * of the group:BUNDLE attribute, but need not be marked
+ * as bundle-only.
+ */
+ gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
+ "max-bundle");
+
+ /* negotiate an AV stream and then renegotiate an extra stream */
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ test_validate_sdp (t, &offer, &answer);
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+ add_fake_audio_src_harness (h, 98);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ offer_setup.next = &offer_bundle_only_sdp;
+ answer_setup.next = &answer_bundle_only_sdp;
+ count.user_data = GUINT_TO_POINTER (3);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bundle_max_compat_max_bundle_renego_add_stream)
+{
+ struct test_webrtc *t = create_audio_video_test ();
+ const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+ const gchar *bundle_only[] = { NULL };
+ guint media_format_count[] = { 1, 1, 1 };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", "active", "active" };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] =
+ { "sendrecv", "sendrecv", "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] =
+ { "sendrecv", "recvonly", "recvonly" };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+
+ VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
+ VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
+ &renego_mid);
+ VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
+ VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
+ &renego_sess_id);
+ VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
+ &renego_sess_ver);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
+ VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (3), &bundle_tag);
+ VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
+ &count_non_reject);
+ GstHarness *h;
+
+ /* We set a max-compat policy on the offering webrtcbin,
+ * this means that all the offered medias should be part
+ * of the group:BUNDLE attribute, and they should *not* be marked
+ * as bundle-only
+ */
+ gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
+ "max-compat");
+ /* We set a max-bundle policy on the answering webrtcbin,
+ * this means that all the offered medias should be part
+ * of the group:BUNDLE attribute, but need not be marked
+ * as bundle-only.
+ */
+ gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
+ "max-bundle");
+
+ /* negotiate an AV stream and then renegotiate an extra stream */
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ test_validate_sdp (t, &offer, &answer);
+
+ h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
+ add_fake_audio_src_harness (h, 98);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ media_formats.next = &bundle_sdp;
+ count.user_data = GUINT_TO_POINTER (3);
+
+ /* renegotiate! */
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_renego_transceiver_set_direction)
+{
+ struct test_webrtc *t = create_audio_test ();
+ guint media_format_count[] = { 1, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
+ &media_formats);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
+ const gchar *expected_offer_setup[] = { "actpass", };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &payloads);
+ const gchar *expected_answer_setup[] = { "active", };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &payloads);
+ const gchar *expected_offer_direction[] = { "sendrecv", };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "sendrecv", };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ GstWebRTCRTPTransceiver *transceiver;
+ GstHarness *h;
+ GstPad *pad;
+
+ /* negotiate an AV stream and then change the transceiver direction */
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_audio_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+
+ test_validate_sdp (t, &offer, &answer);
+
+ /* renegotiate an inactive transceiver! */
+ pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
+ g_object_get (pad, "transceiver", &transceiver, NULL);
+ fail_unless (transceiver != NULL);
+ g_object_set (transceiver, "direction",
+ GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE, NULL);
+ expected_offer_direction[0] = "inactive";
+ expected_answer_direction[0] = "inactive";
+
+ /* TODO: also validate EOS events from the inactive change */
+
+ test_webrtc_reset_negotiation (t);
+ test_validate_sdp (t, &offer, &answer);
+
+ gst_object_unref (pad);
+ gst_object_unref (transceiver);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+static void
+offer_remove_last_media (struct test_webrtc *t, GstElement * element,
+ GstPromise * promise, gpointer user_data)
+{
+ guint i, n;
+ GstSDPMessage *new, *old;
+ const GstSDPOrigin *origin;
+ const GstSDPConnection *conn;
+
+ old = t->offer_desc->sdp;
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_new (&new));
+
+ origin = gst_sdp_message_get_origin (old);
+ conn = gst_sdp_message_get_connection (old);
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_version (new,
+ gst_sdp_message_get_version (old)));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_origin (new,
+ origin->username, origin->sess_id, origin->sess_version,
+ origin->nettype, origin->addrtype, origin->addr));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_session_name (new,
+ gst_sdp_message_get_session_name (old)));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_information (new,
+ gst_sdp_message_get_information (old)));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_uri (new,
+ gst_sdp_message_get_uri (old)));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_connection (new,
+ conn->nettype, conn->addrtype, conn->address, conn->ttl,
+ conn->addr_number));
+
+ n = gst_sdp_message_attributes_len (old);
+ for (i = 0; i < n; i++) {
+ const GstSDPAttribute *a = gst_sdp_message_get_attribute (old, i);
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_attribute (new,
+ a->key, a->value));
+ }
+
+ n = gst_sdp_message_medias_len (old);
+ fail_unless (n > 0);
+ for (i = 0; i < n - 1; i++) {
+ const GstSDPMedia *m = gst_sdp_message_get_media (old, i);
+ GstSDPMedia *new_m;
+
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_media_copy (m, &new_m));
+ fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_media (new, new_m));
+ gst_sdp_media_init (new_m);
+ gst_sdp_media_free (new_m);
+ }
+
+ gst_webrtc_session_description_free (t->offer_desc);
+ t->offer_desc = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER,
+ new);
+}
+
+static void
+offer_set_produced_error (struct test_webrtc *t, GstElement * element,
+ GstPromise * promise, gpointer user_data)
+{
+ const GstStructure *reply;
+ GError *error = NULL;
+
+ reply = gst_promise_get_reply (promise);
+ fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
+ GST_INFO ("error produced: %s", error->message);
+ g_clear_error (&error);
+
+ test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
+}
+
+GST_START_TEST (test_renego_lose_media_fails)
+{
+ struct test_webrtc *t = create_audio_video_test ();
+ VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
+ VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
+
+ /* check that removing an m=line will produce an error */
+
+ test_validate_sdp (t, &offer, &answer);
+
+ test_webrtc_reset_negotiation (t);
+
+ t->on_offer_created = offer_remove_last_media;
+ t->on_offer_set = offer_set_produced_error;
+ t->on_answer_created = NULL;
test_webrtc_create_offer (t, t->webrtc1);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bundle_codec_preferences_rtx_no_duplicate_payloads)
+{
+ struct test_webrtc *t = test_webrtc_new ();
+ GstWebRTCRTPTransceiverDirection direction;
+ GstWebRTCRTPTransceiver *trans;
+ guint offer_media_format_count[] = { 2, };
+ guint answer_media_format_count[] = { 1, };
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, NULL);
+ VAL_SDP_INIT (offer_media_formats, on_sdp_media_count_formats,
+ offer_media_format_count, &payloads);
+ VAL_SDP_INIT (answer_media_formats, on_sdp_media_count_formats,
+ answer_media_format_count, &payloads);
+ const gchar *expected_offer_setup[] = { "actpass", };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &offer_media_formats);
+ const gchar *expected_answer_setup[] = { "active", };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &answer_media_formats);
+ const gchar *expected_offer_direction[] = { "recvonly", };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "sendonly", };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ GstCaps *caps;
+ GstHarness *h;
+
+ /* add a transceiver that will only receive an opus stream and check that
+ * the created offer is marked as recvonly */
+ t->on_negotiation_needed = NULL;
+ t->on_ice_candidate = NULL;
+ t->on_pad_added = _pad_added_fakesink;
+
+ 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");
+
+ /* setup recvonly transceiver */
+ caps = gst_caps_from_string (VP8_RTP_CAPS (96));
+ direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
+ g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
+ &trans);
+ g_object_set (GST_OBJECT (trans), "do-nack", TRUE, NULL);
+ gst_caps_unref (caps);
+ fail_unless (trans != NULL);
+ gst_object_unref (trans);
+
+ /* setup sendonly peer */
+ h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+ add_fake_video_src_harness (h, 96);
+ t->harnesses = g_list_prepend (t->harnesses, h);
+ test_validate_sdp (t, &offer, &answer);
+
test_webrtc_free (t);
}
tcase_add_test (tc, test_sdp_no_media);
tcase_add_test (tc, test_session_stats);
tcase_add_test (tc, test_audio);
+ tcase_add_test (tc, test_ice_port_restriction);
tcase_add_test (tc, test_audio_video);
tcase_add_test (tc, test_media_direction);
- tcase_add_test (tc, test_media_setup);
tcase_add_test (tc, test_add_transceiver);
tcase_add_test (tc, test_get_transceivers);
tcase_add_test (tc, test_add_recvonly_transceiver);
tcase_add_test (tc, test_bundle_audio_video_max_bundle_max_bundle);
tcase_add_test (tc, test_bundle_audio_video_max_bundle_none);
tcase_add_test (tc, test_bundle_audio_video_max_compat_max_bundle);
+ tcase_add_test (tc, test_dual_audio);
tcase_add_test (tc, test_duplicate_nego);
+ tcase_add_test (tc, test_renego_add_stream);
+ tcase_add_test (tc, test_bundle_renego_add_stream);
+ tcase_add_test (tc, test_bundle_max_compat_max_bundle_renego_add_stream);
+ tcase_add_test (tc, test_renego_transceiver_set_direction);
+ tcase_add_test (tc, test_renego_lose_media_fails);
+ tcase_add_test (tc,
+ test_bundle_codec_preferences_rtx_no_duplicate_payloads);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);
tcase_add_test (tc, test_data_channel_max_message_size);
tcase_add_test (tc, test_data_channel_pre_negotiated);
tcase_add_test (tc, test_bundle_audio_video_data);
+ tcase_add_test (tc, test_renego_stream_add_data_channel);
+ tcase_add_test (tc, test_renego_data_channel_add_stream);
+ tcase_add_test (tc, test_renego_stream_data_channel_add_stream);
} else {
GST_WARNING ("Some required elements were not found. "
- "All datachannel are disabled. sctpenc %p, sctpdec %p", sctpenc,
+ "All datachannel tests are disabled. sctpenc %p, sctpdec %p", sctpenc,
sctpdec);
}
} else {
projects / platform / upstream / gstreamer.git / blobdiff