#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_NEGOTATION_NEEDED,
+ 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;
g_free (desc);
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);
+ 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);
- t->state = STATE_ANSWER_CREATED;
- g_cond_broadcast (&t->cond);
+ test_webrtc_signal_state_unlocked (t, STATE_ANSWER_CREATED);
g_mutex_unlock (&t->lock);
+}
+
+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;
g_free (desc);
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);
+ 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);
promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
- t->state = STATE_OFFER_CREATED;
- g_cond_broadcast (&t->cond);
+ test_webrtc_signal_state_unlocked (t, STATE_OFFER_CREATED);
g_mutex_unlock (&t->lock);
-
- gst_webrtc_session_description_free (offer);
}
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:
if (t->on_negotiation_needed)
t->on_negotiation_needed (t, webrtc, t->negotiation_data);
if (t->state == STATE_NEW)
- t->state = STATE_NEGOTATION_NEEDED;
+ t->state = STATE_NEGOTIATION_NEEDED;
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
}
}
}
-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)
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 (offer, _count_num_sdp_media, GUINT_TO_POINTER (0), NULL);
+ VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (0), NULL);
/* check that a no stream connection creates 0 media sections */
t->on_negotiation_needed = NULL;
- t->offer_data = GUINT_TO_POINTER (0);
- t->on_offer_created = _count_num_sdp_media;
- t->answer_data = GUINT_TO_POINTER (0);
- t->on_answer_created = _count_num_sdp_media;
+ 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 (t->state == STATE_ANSWER_CREATED);
test_webrtc_free (t);
}
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 (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
+ VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
/* check that a single stream connection creates the associated number
* of media sections */
- t->on_negotiation_needed = NULL;
- t->offer_data = GUINT_TO_POINTER (1);
- t->on_offer_created = _count_num_sdp_media;
- t->answer_data = GUINT_TO_POINTER (1);
- t->on_answer_created = _count_num_sdp_media;
- t->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);
}
static struct test_webrtc *
create_audio_video_test (void)
{
- struct test_webrtc *t = test_webrtc_new ();
+ struct test_webrtc *t = create_audio_test ();
GstHarness *h;
- t->on_negotiation_needed = 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, 96);
- t->harnesses = g_list_prepend (t->harnesses, h);
-
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);
GST_START_TEST (test_audio_video)
{
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 a dual stream connection creates the associated number
* of media sections */
- t->on_negotiation_needed = NULL;
- t->offer_data = GUINT_TO_POINTER (2);
- t->on_offer_created = _count_num_sdp_media;
- t->answer_data = GUINT_TO_POINTER (2);
- t->on_answer_created = _count_num_sdp_media;
- t->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;
-typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
- GstWebRTCSessionDescription * desc, gpointer user_data);
-
-struct validate_sdp
-{
- ValidateSDPFunc validate;
- gpointer user_data;
-};
-
-static GstWebRTCSessionDescription *
-_check_validate_sdp (struct test_webrtc *t, GstElement * element,
- GstPromise * promise, gpointer user_data)
-{
- struct validate_sdp *validate = user_data;
- GstWebRTCSessionDescription *offer = NULL;
- const GstStructure *reply;
- const gchar *field;
-
- field = t->offerror == 1 && t->webrtc1 == element ? "offer" : "answer";
-
- reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, field,
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
-
- validate->validate (t, element, offer, validate->user_data);
-
- return offer;
-}
-
static void
on_sdp_media_direction (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
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;
- 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);
+ 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);
}
- fail_unless (have_direction, "no direction attribute in media %u", j);
}
}
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 };
- struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
GstHarness *h;
+ VAL_SDP_INIT (offer_direction, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &offer_direction);
+ VAL_SDP_INIT (answer_direction, on_sdp_media_direction, expected_answer,
+ NULL);
+ VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2),
+ &answer_direction);
/* 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 };
+ VAL_SDP_INIT (payloads, on_sdp_media_payload_types, NULL, NULL);
+ VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
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);
+ /* We don't really care about the answer here */
+ test_validate_sdp (t, &offer, NULL);
test_webrtc_free (t);
}
fail_unless (g_strcmp0 (attr->value, expected_setup[i]) == 0);
}
}
- fail_unless (have_setup, "no setup attribute in media %u", j);
+ fail_unless (have_setup, "no setup attribute in media %u", i);
}
}
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 };
- struct validate_sdp answer = { on_sdp_media_setup, expected_answer };
+ VAL_SDP_INIT (offer, on_sdp_media_setup, expected_offer, NULL);
+ VAL_SDP_INIT (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);
+ test_validate_sdp (t, &offer, &answer);
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);
GstWebRTCRTPTransceiver *trans;
const gchar *expected_offer[] = { "recvonly" };
const gchar *expected_answer[] = { "sendonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer };
- struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
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);
}
GstWebRTCRTPTransceiver *trans;
const gchar *expected_offer[] = { "recvonly", "sendonly" };
const gchar *expected_answer[] = { "sendonly", "recvonly" };
- struct validate_sdp offer = { on_sdp_media_direction, expected_offer };
- struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
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));
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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer);
- 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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 };
- struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
+ VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, NULL);
+ VAL_SDP_INIT (answer, on_sdp_has_datachannel, NULL, NULL);
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, &answer, 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));
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 };
+
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
+ 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, _check_bundle_only_media, &offer_bundle_only,
+ &offer_non_reject);
+ VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
+ &answer_non_reject);
+
/* 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 };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
-
- 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 };
+
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
+ 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);
+
/* 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 };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
-
- 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, &bundle_sdp, &bundle_sdp);
test_webrtc_free (t);
}
const gchar *offer_bundle_only[] = { "video1", NULL };
const gchar *answer_bundle[] = { NULL };
const gchar *answer_bundle_only[] = { NULL };
+
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
+ VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &count);
+ VAL_SDP_INIT (offer_bundle_tag, _check_bundle_tag, offer_bundle,
+ &count_non_reject);
+ VAL_SDP_INIT (answer_bundle_tag, _check_bundle_tag, answer_bundle,
+ &count_non_reject);
+ VAL_SDP_INIT (offer, _check_bundle_only_media, &offer_bundle_only,
+ &offer_bundle_tag);
+ VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
+ &answer_bundle_tag);
+
/* 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 };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
-
- 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);
}
const gchar *offer_bundle_only[] = { "video1", "application2", NULL };
const gchar *answer_bundle_only[] = { NULL };
GObject *channel = NULL;
+
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (3), NULL);
+ VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
+ 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, _check_bundle_only_media, &offer_bundle_only,
+ &offer_non_reject);
+ VAL_SDP_INIT (answer, _check_bundle_only_media, &answer_bundle_only,
+ &answer_non_reject);
+
/* 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 };
- struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
-
- 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");
+ 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_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
+ &channel);
+
+ test_validate_sdp (t, &offer, &answer);
+
+ g_object_unref (channel);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+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" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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 ();
+ const gchar *expected_offer[] = { "sendrecv", "sendrecv", };
+ const gchar *expected_answer[] = { "sendrecv", "recvonly" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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 ();
+ const gchar *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+ const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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);
+
+ offer.next = &renego_fingerprint;
+ answer.next = &renego_fingerprint;
+
+ /* 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 ();
+ const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
+ const gchar *expected_answer[] = { "sendrecv", "recvonly" };
+
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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);
+
+ offer.next = &renego_fingerprint;
+ answer.next = &renego_fingerprint;
+
+ /* 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 ();
+ const gchar *expected_offer[] = { NULL, "sendrecv" };
+ const gchar *expected_answer[] = { NULL, "recvonly" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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 */
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);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
- test_webrtc_create_offer (t, t->webrtc1);
+ test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
- test_webrtc_wait_for_answer_error_eos (t);
- fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+ 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);
+
+ offer.next = &renego_fingerprint;
+ answer.next = &renego_fingerprint;
+
+ /* 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 *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+ const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+ const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+ const gchar *offer_bundle_only[] = { "video1", "audio2", NULL };
+ const gchar *answer_bundle_only[] = { NULL };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+
+ 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 (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.next = &offer_bundle_only_sdp;
+ answer.next = &answer_bundle_only_sdp;
+
+ /* 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 *expected_offer[] = { "sendrecv", "sendrecv", "sendrecv" };
+ const gchar *expected_answer[] = { "sendrecv", "recvonly", "recvonly" };
+ const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
+ const gchar *bundle_only[] = { NULL };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+
+ 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);
+
+ offer.next = &bundle_sdp;
+ answer.next = &bundle_sdp;
+
+ /* 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 ();
+ const gchar *expected_offer[] = { "sendrecv" };
+ const gchar *expected_answer[] = { "sendrecv" };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer, NULL);
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer, NULL);
+ 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[0] = "inactive";
+ expected_answer[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 Suite *
webrtcbin_suite (void)
{
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);
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);
} 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 {
+++ /dev/null
-/*
- * Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
- * with a browser JS app.
- *
- * gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
- *
- * Author: Nirbheek Chauhan <nirbheek@centricular.com>
- */
-#include <gst/gst.h>
-#include <gst/sdp/sdp.h>
-
-#ifndef GST_USE_UNSTABLE_API
-#define GST_USE_UNSTABLE_API
-#endif
-#include <gst/webrtc/webrtc.h>
-
-/* For signalling */
-#include <libsoup/soup.h>
-#include <json-glib/json-glib.h>
-
-#include <string.h>
-#define HTTP_PROXY "http://10.112.1.184:8080"
-#define ENTER g_print ("%s:%d>%s\n",__FILE__, __LINE__, __FUNCTION__);
-enum AppState {
- APP_STATE_UNKNOWN = 0,
- APP_STATE_ERROR = 1, /* generic error */
- SERVER_CONNECTING = 1000,
- SERVER_CONNECTION_ERROR,
- SERVER_CONNECTED, /* Ready to register */
- SERVER_REGISTERING = 2000,
- SERVER_REGISTRATION_ERROR,
- SERVER_REGISTERED, /* Ready to call a peer */
- SERVER_CLOSED, /* server connection closed by us or the server */
- PEER_CONNECTING = 3000,
- PEER_CONNECTION_ERROR,
- PEER_CONNECTED,
- PEER_CALL_NEGOTIATING = 4000,
- PEER_CALL_WAITING,
- PEER_CALL_STARTED,
- PEER_CALL_STOPPING,
- PEER_CALL_STOPPED,
- PEER_CALL_ERROR,
-};
-
-static GMainLoop *loop;
-static GstElement *pipe1, *webrtc1;
-static GObject *send_channel, *receive_channel;
-
-static SoupWebsocketConnection *ws_conn = NULL;
-static enum AppState app_state = 0;
-static const gchar *peer_id = NULL;
-static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
-static gboolean disable_ssl = FALSE;
-static gboolean remote_is_offerer = FALSE;
-static gboolean use_camera_mic = FALSE;
-static gboolean use_proxy = FALSE;
-static gint32 our_id = 0;
-
-static GOptionEntry entries[] =
-{
- { "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" },
- { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
- { "disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL },
- { "remote-offerer", 0, 0, G_OPTION_ARG_NONE, &remote_is_offerer, "Request that the peer generate the offer and we'll answer", NULL },
- { "use-camera-mic", 0, 0, G_OPTION_ARG_NONE, &use_camera_mic, "Use camera and mic", NULL },
- { "use-proxy", 0, 0, G_OPTION_ARG_NONE, &use_proxy, "Use proxy", NULL },
- { NULL },
-};
-
-static gboolean
-cleanup_and_quit_loop (const gchar * msg, enum AppState state)
-{
- ENTER;
-
- if (msg)
- g_printerr ("%s\n", msg);
- if (state > 0)
- app_state = state;
-
- if (ws_conn) {
- if (soup_websocket_connection_get_state (ws_conn) ==
- SOUP_WEBSOCKET_STATE_OPEN)
- /* This will call us again */
- soup_websocket_connection_close (ws_conn, 1000, "");
- else
- g_object_unref (ws_conn);
- }
-
- if (loop) {
- g_main_loop_quit (loop);
- loop = NULL;
- }
-
- /* To allow usage as a GSourceFunc */
- return G_SOURCE_REMOVE;
-}
-
-static gchar*
-get_string_from_json_object (JsonObject * object)
-{
- JsonNode *root;
- JsonGenerator *generator;
- gchar *text;
- ENTER;
-
- /* Make it the root node */
- root = json_node_init_object (json_node_alloc (), object);
- generator = json_generator_new ();
- json_generator_set_root (generator, root);
- text = json_generator_to_data (generator, NULL);
-
- /* Release everything */
- g_object_unref (generator);
- json_node_free (root);
- return text;
-}
-
-static void
-handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
- const char * sink_name)
-{
- GstPad *qpad;
- GstElement *q, *conv, *resample, *sink;
- GstPadLinkReturn ret;
- ENTER;
-
- g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
-
- q = gst_element_factory_make ("queue", NULL);
- g_assert_nonnull (q);
- conv = gst_element_factory_make (convert_name, NULL);
- g_assert_nonnull (conv);
- sink = gst_element_factory_make (sink_name, NULL);
- g_assert_nonnull (sink);
-
- if (g_strcmp0 (convert_name, "audioconvert") == 0) {
- /* Might also need to resample, so add it just in case.
- * Will be a no-op if it's not required. */
- resample = gst_element_factory_make ("audioresample", NULL);
- g_assert_nonnull (resample);
- gst_bin_add_many (GST_BIN (pipe), q, conv, resample, sink, NULL);
- gst_element_sync_state_with_parent (q);
- gst_element_sync_state_with_parent (conv);
- gst_element_sync_state_with_parent (resample);
- gst_element_sync_state_with_parent (sink);
- gst_element_link_many (q, conv, resample, sink, NULL);
- } else {
- gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL);
- gst_element_sync_state_with_parent (q);
- gst_element_sync_state_with_parent (conv);
- gst_element_sync_state_with_parent (sink);
- gst_element_link_many (q, conv, sink, NULL);
- }
-
- qpad = gst_element_get_static_pad (q, "sink");
-
- ret = gst_pad_link (pad, qpad);
- g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);
-}
-
-static void
-on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
- GstElement * pipe)
-{
- GstCaps *caps;
- const gchar *name;
- ENTER;
-
- if (!gst_pad_has_current_caps (pad)) {
- g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
- GST_PAD_NAME (pad));
- return;
- }
-
- caps = gst_pad_get_current_caps (pad);
- name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
-
- if (g_str_has_prefix (name, "video")) {
- handle_media_stream (pad, pipe, "videoconvert", "autovideosink");
- } else if (g_str_has_prefix (name, "audio")) {
- handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
- } else {
- g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
- }
-}
-
-static void
-on_incoming_stream (GstElement * webrtc, GstPad * pad, GstElement * pipe)
-{
- GstElement *decodebin;
- GstPad *sinkpad;
- ENTER;
-
- if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
- return;
-
- decodebin = gst_element_factory_make ("decodebin", NULL);
- g_signal_connect (decodebin, "pad-added",
- G_CALLBACK (on_incoming_decodebin_stream), pipe);
- gst_bin_add (GST_BIN (pipe), decodebin);
- gst_element_sync_state_with_parent (decodebin);
-
- sinkpad = gst_element_get_static_pad (decodebin, "sink");
- gst_pad_link (pad, sinkpad);
- gst_object_unref (sinkpad);
-}
-
-static void
-send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex,
- gchar * candidate, gpointer user_data G_GNUC_UNUSED)
-{
- gchar *text;
- JsonObject *ice, *msg;
- ENTER;
-
- if (app_state < PEER_CALL_NEGOTIATING) {
- cleanup_and_quit_loop ("Can't send ICE, not in call", APP_STATE_ERROR);
- return;
- }
-
- ice = json_object_new ();
- json_object_set_string_member (ice, "candidate", candidate);
- json_object_set_int_member (ice, "sdpMLineIndex", mlineindex);
- msg = json_object_new ();
- json_object_set_object_member (msg, "ice", ice);
- text = get_string_from_json_object (msg);
- json_object_unref (msg);
-
- soup_websocket_connection_send_text (ws_conn, text);
- g_free (text);
-}
-
-static void
-send_sdp_to_peer (GstWebRTCSessionDescription *desc)
-{
- gchar *text;
- JsonObject *msg, *sdp;
- ENTER;
-
- if (app_state < PEER_CALL_NEGOTIATING) {
- cleanup_and_quit_loop ("Can't send SDP to peer, not in call", APP_STATE_ERROR);
- return;
- }
-
- text = gst_sdp_message_as_text (desc->sdp);
- sdp = json_object_new ();
-
- if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER) {
- g_print ("Sending offer:\n%s\n", text);
- json_object_set_string_member (sdp, "type", "offer");
- }
- else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
- g_print ("Sending answer:\n%s\n", text);
- json_object_set_string_member (sdp, "type", "answer");
- }
- else {
- g_assert_not_reached ();
- }
-
- json_object_set_string_member (sdp, "sdp", text);
- g_free (text);
-
- msg = json_object_new ();
- json_object_set_object_member (msg, "sdp", sdp);
- text = get_string_from_json_object (msg);
- json_object_unref (msg);
-
- soup_websocket_connection_send_text (ws_conn, text);
- g_free (text);
-}
-
-/* Offer created by our pipeline, to be sent to the peer */
-static void
-on_offer_created (GstPromise * promise, gpointer user_data)
-{
- GstWebRTCSessionDescription *offer = NULL;
- const GstStructure *reply;
- ENTER;
-
- g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
-
- g_assert_cmphex (gst_promise_wait(promise), ==, GST_PROMISE_RESULT_REPLIED);
- reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, "offer",
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
- gst_promise_unref (promise);
-
- promise = gst_promise_new ();
- g_signal_emit_by_name (webrtc1, "set-local-description", offer, promise);
- gst_promise_interrupt (promise);
- gst_promise_unref (promise);
-
- /* Send offer to peer */
- send_sdp_to_peer (offer);
- gst_webrtc_session_description_free (offer);
-}
-
-static void
-on_negotiation_needed (GstElement * element, gpointer user_data)
-{
- app_state = PEER_CALL_NEGOTIATING;
- ENTER;
-
- if (remote_is_offerer) {
- gchar *msg = g_strdup_printf ("OFFER_REQUEST");
- soup_websocket_connection_send_text (ws_conn, msg);
- g_free (msg);
- } else {
- GstPromise *promise;
- promise = gst_promise_new_with_change_func (on_offer_created, user_data, NULL);;
- g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
- }
-}
-
-#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
-#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
-#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="
-
-static void
-data_channel_on_error (GObject * dc, gpointer user_data)
-{
- ENTER;
-
- cleanup_and_quit_loop ("Data channel error", 0);
-}
-
-static void
-data_channel_on_open (GObject * dc, gpointer user_data)
-{
- GBytes *bytes = g_bytes_new ("data", strlen("data"));
- ENTER;
-
- g_print ("data channel opened\n");
- g_signal_emit_by_name (dc, "send-string", "Hi! from GStreamer");
- g_signal_emit_by_name (dc, "send-data", bytes);
- g_bytes_unref (bytes);
-}
-
-static void
-data_channel_on_close (GObject * dc, gpointer user_data)
-{
- ENTER;
-
- cleanup_and_quit_loop ("Data channel closed", 0);
-}
-
-static void
-data_channel_on_message_string (GObject * dc, gchar *str, gpointer user_data)
-{
- ENTER;
-
- g_print ("Received data channel message: %s\n", str);
-}
-
-static void
-connect_data_channel_signals (GObject * data_channel)
-{
- ENTER;
-
- g_signal_connect (data_channel, "on-error", G_CALLBACK (data_channel_on_error),
- NULL);
- g_signal_connect (data_channel, "on-open", G_CALLBACK (data_channel_on_open),
- NULL);
- g_signal_connect (data_channel, "on-close", G_CALLBACK (data_channel_on_close),
- NULL);
- g_signal_connect (data_channel, "on-message-string", G_CALLBACK (data_channel_on_message_string),
- NULL);
-}
-
-static void
-on_data_channel (GstElement * webrtc, GObject * data_channel, gpointer user_data)
-{
- ENTER;
-
- connect_data_channel_signals (data_channel);
- receive_channel = data_channel;
-}
-
-static void
-on_ice_gathering_state_notify (GstElement * webrtcbin, GParamSpec * pspec,
- gpointer user_data)
-{
- GstWebRTCICEGatheringState ice_gather_state;
- const gchar *new_state = "unknown";
- ENTER;
-
- g_object_get (webrtcbin, "ice-gathering-state", &ice_gather_state,
- NULL);
- switch (ice_gather_state) {
- case GST_WEBRTC_ICE_GATHERING_STATE_NEW:
- new_state = "new";
- break;
- case GST_WEBRTC_ICE_GATHERING_STATE_GATHERING:
- new_state = "gathering";
- break;
- case GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE:
- new_state = "complete";
- break;
- }
- g_print ("ICE gathering state changed to %s\n", new_state);
-}
-
-static gboolean
-start_pipeline (void)
-{
- GstStateChangeReturn ret;
- GError *error = NULL;
- ENTER;
-
- if (!use_camera_mic)
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
- "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
- "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
- &error);
- else
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
- "camerasrc camera-id=1 ! ""video/x-raw,format=I420,width=352,height=288"" ! queue ! vp8enc deadline=1 ! rtpvp8pay ! " //avenc_h263 ! rtph263pay ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
- "pulsesrc ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
- &error);
-
- if (error) {
- g_printerr ("Failed to parse launch: %s\n", error->message);
- g_error_free (error);
- goto err;
- }
-
- webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
- g_assert_nonnull (webrtc1);
-
- /* This is the gstwebrtc entry point where we create the offer and so on. It
- * will be called when the pipeline goes to PLAYING. */
- g_signal_connect (webrtc1, "on-negotiation-needed",
- G_CALLBACK (on_negotiation_needed), NULL);
- /* We need to transmit this ICE candidate to the browser via the websockets
- * signalling server. Incoming ice candidates from the browser need to be
- * added by us too, see on_server_message() */
- g_signal_connect (webrtc1, "on-ice-candidate",
- G_CALLBACK (send_ice_candidate_message), NULL);
- g_signal_connect (webrtc1, "notify::ice-gathering-state",
- G_CALLBACK (on_ice_gathering_state_notify), NULL);
-
- gst_element_set_state (pipe1, GST_STATE_READY);
-
- g_signal_emit_by_name (webrtc1, "create-data-channel", "channel", NULL,
- &send_channel);
- if (send_channel) {
- g_print ("Created data channel\n");
- connect_data_channel_signals (send_channel);
- } else {
- g_print ("Could not create data channel, is usrsctp available?\n");
- }
-
- g_signal_connect (webrtc1, "on-data-channel", G_CALLBACK (on_data_channel),
- NULL);
- /* Incoming streams will be exposed via this signal */
- g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
- pipe1);
- /* Lifetime is the same as the pipeline itself */
- gst_object_unref (webrtc1);
-
- g_print ("Starting pipeline\n");
- ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
- if (ret == GST_STATE_CHANGE_FAILURE)
- goto err;
-
- return TRUE;
-
-err:
- if (pipe1)
- g_clear_object (&pipe1);
- if (webrtc1)
- webrtc1 = NULL;
- return FALSE;
-}
-
-static gboolean
-start_pipeline_answer (void)
-{
- GstStateChangeReturn ret;
- GError *error = NULL;
- ENTER;
-
- if (!use_camera_mic)
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
- "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
- "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
- &error);
- else
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
- "camerasrc camera-id=1 ! ""video/x-raw,format=I420,width=352,height=288"" ! queue ! vp8enc deadline=1 ! rtpvp8pay ! " //avenc_h263 ! rtph263pay ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
- "pulsesrc ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
- &error);
-
- if (error) {
- g_printerr ("Failed to parse launch: %s\n", error->message);
- g_error_free (error);
- goto err;
- }
-
- webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
- g_assert_nonnull (webrtc1);
-
- /* We need to transmit this ICE candidate to the browser via the websockets
- * signalling server. Incoming ice candidates from the browser need to be
- * added by us too, see on_server_message() */
- g_signal_connect (webrtc1, "on-ice-candidate",
- G_CALLBACK (send_ice_candidate_message), NULL);
- g_signal_connect (webrtc1, "notify::ice-gathering-state",
- G_CALLBACK (on_ice_gathering_state_notify), NULL);
-
- gst_element_set_state (pipe1, GST_STATE_READY);
-
- g_signal_emit_by_name (webrtc1, "create-data-channel", "channel", NULL,
- &send_channel);
- if (send_channel) {
- g_print ("Created data channel\n");
- connect_data_channel_signals (send_channel);
- } else {
- g_print ("Could not create data channel, is usrsctp available?\n");
- }
-
- g_signal_connect (webrtc1, "on-data-channel", G_CALLBACK (on_data_channel),
- NULL);
- /* Incoming streams will be exposed via this signal */
- g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
- pipe1);
- /* Lifetime is the same as the pipeline itself */
- gst_object_unref (webrtc1);
-
- g_print ("Starting pipeline, our id(%d)\n", our_id);
- ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
- if (ret == GST_STATE_CHANGE_FAILURE)
- goto err;
-
- return TRUE;
-
-err:
- if (pipe1)
- g_clear_object (&pipe1);
- if (webrtc1)
- webrtc1 = NULL;
- return FALSE;
-}
-
-
-static gboolean
-setup_call (void)
-{
- gchar *msg;
- ENTER;
-
- if (soup_websocket_connection_get_state (ws_conn) !=
- SOUP_WEBSOCKET_STATE_OPEN)
- return FALSE;
-
- if (!peer_id)
- return FALSE;
-
- g_print ("Setting up signalling server call with %s\n", peer_id);
- app_state = PEER_CONNECTING;
- msg = g_strdup_printf ("SESSION %s", peer_id);
- soup_websocket_connection_send_text (ws_conn, msg);
- g_free (msg);
- return TRUE;
-}
-
-static gint32
-register_with_server (void)
-{
- gchar *hello;
- gint32 our_id;
- ENTER;
-
- if (soup_websocket_connection_get_state (ws_conn) !=
- SOUP_WEBSOCKET_STATE_OPEN)
- return -1;
-
- our_id = g_random_int_range (10, 10000);
- g_print ("Registering id %i with server\n", our_id);
- app_state = SERVER_REGISTERING;
-
- /* Register with the server with a random integer id. Reply will be received
- * by on_server_message() */
- hello = g_strdup_printf ("HELLO %i", our_id);
- soup_websocket_connection_send_text (ws_conn, hello);
- g_free (hello);
-
- return our_id;
-}
-
-static void
-on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
- gpointer user_data G_GNUC_UNUSED)
-{
- app_state = SERVER_CLOSED;
- ENTER;
-
- cleanup_and_quit_loop ("Server connection closed", 0);
-}
-
-/* Answer created by our pipeline, to be sent to the peer */
-static void
-on_answer_created (GstPromise * promise, gpointer user_data)
-{
- GstWebRTCSessionDescription *answer = NULL;
- const GstStructure *reply;
- ENTER;
-
- if (peer_id)
- g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
- else
- g_assert_cmphex (app_state, ==, PEER_CALL_WAITING);
-
- g_assert_cmphex (gst_promise_wait(promise), ==, GST_PROMISE_RESULT_REPLIED);
- reply = gst_promise_get_reply (promise);
- gst_structure_get (reply, "answer",
- GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
- gst_promise_unref (promise);
-
- promise = gst_promise_new ();
- g_signal_emit_by_name (webrtc1, "set-local-description", answer, promise);
- gst_promise_interrupt (promise);
- gst_promise_unref (promise);
-
- /* Send answer to peer */
- send_sdp_to_peer (answer);
- gst_webrtc_session_description_free (answer);
-}
-
-static void
-on_offer_received (GstSDPMessage *sdp)
-{
- GstWebRTCSessionDescription *offer = NULL;
- GstPromise *promise;
- ENTER;
-
- offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, sdp);
- g_assert_nonnull (offer);
-
- /* Set remote description on our pipeline */
- {
- promise = gst_promise_new ();
- g_signal_emit_by_name (webrtc1, "set-remote-description", offer,
- promise);
- gst_promise_interrupt (promise);
- gst_promise_unref (promise);
- }
- gst_webrtc_session_description_free (offer);
-
- promise = gst_promise_new_with_change_func (on_answer_created, NULL,
- NULL);
- g_signal_emit_by_name (webrtc1, "create-answer", NULL, promise);
-}
-
-/* One mega message handler for our asynchronous calling mechanism */
-static void
-on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
- GBytes * message, gpointer user_data)
-{
- gchar *text;
- ENTER;
-
- switch (type) {
- case SOUP_WEBSOCKET_DATA_BINARY:
- g_printerr ("Received unknown binary message, ignoring\n");
- return;
- case SOUP_WEBSOCKET_DATA_TEXT: {
- gsize size;
- const gchar *data = g_bytes_get_data (message, &size);
- /* Convert to NULL-terminated string */
- text = g_strndup (data, size);
- g_print ("Received text message, [%s]\n", text);
- break;
- }
- default:
- g_assert_not_reached ();
- }
-
- /* Server has accepted our registration, we are ready to send commands */
- if (g_strcmp0 (text, "HELLO") == 0) {
- if (app_state != SERVER_REGISTERING) {
- cleanup_and_quit_loop ("ERROR: Received HELLO when not registering",
- APP_STATE_ERROR);
- goto out;
- }
- app_state = SERVER_REGISTERED;
- g_print ("Registered with server\n");
- /* Ask signalling server to connect us with a specific peer */
- if (peer_id) {
- if (!setup_call ()) {
- cleanup_and_quit_loop ("ERROR: Failed to setup call", PEER_CALL_ERROR);
- goto out;
- }
- } else {
- /* should WAIT for another peer */
- g_print ("need to wait for another peer...(our id:%d)\n", our_id);
- app_state = PEER_CALL_WAITING;
- /* Start negotiation (exchange SDP and ICE candidates) */
- if (!start_pipeline_answer ())
- cleanup_and_quit_loop ("ERROR: failed to start pipeline",
- PEER_CALL_ERROR);
- }
- /* Call has been setup by the server, now we can start negotiation */
- } else if (g_strcmp0 (text, "SESSION_OK") == 0) {
- if (app_state != PEER_CONNECTING) {
- cleanup_and_quit_loop ("ERROR: Received SESSION_OK when not calling",
- PEER_CONNECTION_ERROR);
- goto out;
- }
-
- app_state = PEER_CONNECTED;
- /* Start negotiation (exchange SDP and ICE candidates) */
- if (!start_pipeline ())
- cleanup_and_quit_loop ("ERROR: failed to start pipeline",
- PEER_CALL_ERROR);
- /* Handle errors */
- } else if (g_str_has_prefix (text, "ERROR")) {
- switch (app_state) {
- case SERVER_CONNECTING:
- app_state = SERVER_CONNECTION_ERROR;
- break;
- case SERVER_REGISTERING:
- app_state = SERVER_REGISTRATION_ERROR;
- break;
- case PEER_CONNECTING:
- app_state = PEER_CONNECTION_ERROR;
- break;
- case PEER_CALL_WAITING:
- case PEER_CONNECTED:
- case PEER_CALL_NEGOTIATING:
- app_state = PEER_CALL_ERROR;
- break;
- default:
- app_state = APP_STATE_ERROR;
- }
- cleanup_and_quit_loop (text, 0);
- /* Look for JSON messages containing SDP and ICE candidates */
- } else {
- JsonNode *root;
- JsonObject *object, *child;
- JsonParser *parser = json_parser_new ();
- if (!json_parser_load_from_data (parser, text, -1, NULL)) {
- g_printerr ("Unknown message '%s', ignoring", text);
- g_object_unref (parser);
- goto out;
- }
-
- root = json_parser_get_root (parser);
- if (!JSON_NODE_HOLDS_OBJECT (root)) {
- g_printerr ("Unknown json message '%s', ignoring", text);
- g_object_unref (parser);
- goto out;
- }
-
- object = json_node_get_object (root);
- /* Check type of JSON message */
- if (json_object_has_member (object, "sdp")) {
- int ret;
- GstSDPMessage *sdp;
- const gchar *text, *sdptype;
- GstWebRTCSessionDescription *answer;
-
- if (peer_id)
- g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
- else
- g_assert_cmphex (app_state, ==, PEER_CALL_WAITING);
-
- child = json_object_get_object_member (object, "sdp");
-
- if (!json_object_has_member (child, "type")) {
- cleanup_and_quit_loop ("ERROR: received SDP without 'type'",
- PEER_CALL_ERROR);
- goto out;
- }
-
- sdptype = json_object_get_string_member (child, "type");
- /* In this example, we create the offer and receive one answer by default,
- * but it's possible to comment out the offer creation and wait for an offer
- * instead, so we handle either here.
- *
- * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for another
- * example how to handle offers from peers and reply with answers using webrtcbin. */
- text = json_object_get_string_member (child, "sdp");
- ret = gst_sdp_message_new (&sdp);
- g_assert_cmphex (ret, ==, GST_SDP_OK);
- ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp);
- g_assert_cmphex (ret, ==, GST_SDP_OK);
-
- if (g_str_equal (sdptype, "answer")) {
- g_print ("Received answer:\n%s\n", text);
- answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
- sdp);
- g_assert_nonnull (answer);
-
- /* Set remote description on our pipeline */
- {
- GstPromise *promise = gst_promise_new ();
- g_signal_emit_by_name (webrtc1, "set-remote-description", answer,
- promise);
- gst_promise_interrupt (promise);
- gst_promise_unref (promise);
- }
- app_state = PEER_CALL_STARTED;
- }
- else {
- g_print ("Received offer:\n%s\n", text);
- on_offer_received (sdp);
- }
-
- } else if (json_object_has_member (object, "ice")) {
- const gchar *candidate;
- gint sdpmlineindex;
-
- child = json_object_get_object_member (object, "ice");
- candidate = json_object_get_string_member (child, "candidate");
- sdpmlineindex = json_object_get_int_member (child, "sdpMLineIndex");
-
- /* Add ice candidate sent by remote peer */
- g_signal_emit_by_name (webrtc1, "add-ice-candidate", sdpmlineindex,
- candidate);
- } else {
- g_printerr ("Ignoring unknown JSON message:\n%s\n", text);
- }
- g_object_unref (parser);
- }
-
-out:
- g_free (text);
-}
-
-static void
-on_server_connected (SoupSession * session, GAsyncResult * res,
- SoupMessage *msg)
-{
- GError *error = NULL;
- ENTER;
-
- g_print("on_server_connected\n");
- ws_conn = soup_session_websocket_connect_finish (session, res, &error);
- if (error) {
- cleanup_and_quit_loop (error->message, SERVER_CONNECTION_ERROR);
- g_error_free (error);
- return;
- }
-
- g_assert_nonnull (ws_conn);
-
- app_state = SERVER_CONNECTED;
- g_print ("Connected to signalling server\n");
-
- g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL);
- g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL);
-
- /* Register with the server so it knows about us and can accept commands */
- our_id = register_with_server ();
-}
-
-/*
- * Connect to the signalling server. This is the entrypoint for everything else.
- */
-
-/* TIZEN: add for log */
-static inline gchar
-gst_soup_util_log_make_level_tag (SoupLoggerLogLevel level)
-{
- gchar c;
-
- if (G_UNLIKELY ((gint) level > 9))
- return '?';
-
- switch (level) {
- case SOUP_LOGGER_LOG_MINIMAL:
- c = 'M';
- break;
- case SOUP_LOGGER_LOG_HEADERS:
- c = 'H';
- break;
- case SOUP_LOGGER_LOG_BODY:
- c = 'B';
- break;
- default:
- /* Unknown level. If this is hit libsoup likely added a new
- * log level to SoupLoggerLogLevel and it should be added
- * as a case */
- c = level + '0';
- break;
- }
- return c;
-}
-
-static void
-_log_printer_cb (SoupLogger G_GNUC_UNUSED * logger,
- SoupLoggerLogLevel level, char direction, const char *data,
- gpointer user_data)
-{
- gchar c;
-
- c = gst_soup_util_log_make_level_tag (level);
- g_print("HTTP_SESSION(%c): %c %s\n", c, direction, data);
-}
-
-static void
-connect_to_websocket_server_async (void)
-{
- SoupLogger *logger;
- SoupMessage *message;
- SoupSession *session;
- SoupURI *proxy_uri;
- const char *https_aliases[] = {"wss", NULL};
- ENTER;
-
- if (!use_proxy){
- session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, !disable_ssl,
- SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
- } else {
- proxy_uri = soup_uri_new (HTTP_PROXY);
- session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, !disable_ssl,
- SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
- SOUP_SESSION_PROXY_URI, proxy_uri,
- SOUP_SESSION_SSL_CA_FILE, "/opt/var/lib/ca-certificates/ca-bundle.pem",
- SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
- soup_uri_free (proxy_uri);
- }
-
- logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1);
-
- /* TIZEN: add for log */
- soup_logger_set_printer (logger, _log_printer_cb, NULL, NULL);
-
- soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger));
- g_object_unref (logger);
-
- message = soup_message_new (SOUP_METHOD_GET, server_url);
-
- g_print ("Connecting to server[%s]...\n", server_url);
-
- /* Once connected, we will register */
- soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
- (GAsyncReadyCallback) on_server_connected, message);
- app_state = SERVER_CONNECTING;
-}
-
-static gboolean
-check_plugins (void)
-{
- int i;
- gboolean ret;
- GstPlugin *plugin;
- GstRegistry *registry;
- const gchar *needed[] = { "opus", "vpx", "nice", "webrtc", "dtls", "srtp",
- "rtpmanager", "videotestsrc", "audiotestsrc", NULL};
- ENTER;
-
- registry = gst_registry_get ();
- ret = TRUE;
- for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
- plugin = gst_registry_find_plugin (registry, needed[i]);
- if (!plugin) {
- g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
- ret = FALSE;
- continue;
- }
- gst_object_unref (plugin);
- }
- return ret;
-}
-
-int
-main (int argc, char *argv[])
-{
- GOptionContext *context;
- GError *error = NULL;
- ENTER;
-
- context = g_option_context_new ("- gstreamer webrtc sendrecv demo");
- g_option_context_add_main_entries (context, entries, NULL);
- g_option_context_add_group (context, gst_init_get_option_group ());
- if (!g_option_context_parse (context, &argc, &argv, &error)) {
- g_printerr ("Error initializing: %s\n", error->message);
- return -1;
- }
-
- if (!check_plugins ())
- return -1;
-#if 0
- if (!peer_id) {
- g_printerr ("--peer-id is a required argument\n");
- return -1;
- }
-#endif
-
- /* Disable ssl when running a localhost server, because
- * it's probably a test server with a self-signed certificate */
- {
- GstUri *uri = gst_uri_from_string (server_url);
- if (g_strcmp0 ("localhost", gst_uri_get_host (uri)) == 0 ||
- g_strcmp0 ("127.0.0.1", gst_uri_get_host (uri)) == 0)
- disable_ssl = TRUE;
- gst_uri_unref (uri);
- }
-
- loop = g_main_loop_new (NULL, FALSE);
-
- connect_to_websocket_server_async ();
-
- g_main_loop_run (loop);
- g_main_loop_unref (loop);
-
- if (pipe1) {
- gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
- g_print ("Pipeline stopped\n");
- gst_object_unref (pipe1);
- }
-
- return 0;
-}
projects / platform / upstream / gst-plugins-bad.git / commitdiff