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webrtc: propagate more errors through the promise

Browse source 
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
This commit is contained in:
Matthew Waters 2020-08-26 15:45:35 +10:00 committed by GStreamer Merge Bot
parent d4fa35efb9
commit e2d88f0569
5 changed files with 281 additions and 71 deletions
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158
ext/webrtc/gstwebrtcbin.c
View file

@ -2561,7 +2561,8 @@ _add_data_channel_offer (GstWebRTCBin * webrtc, GstSDPMessage * msg,
/* TODO: use the options argument */
static GstSDPMessage *
_create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
_create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GError ** error)
{
GstSDPMessage *ret;
GString *bundled_mids = NULL;
@ -2604,8 +2605,8 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
guint bundle_media_index;
reserved_pts = gather_reserved_pts (webrtc);
if (last_offer && _parse_bundle (last_offer, &last_bundle) && last_bundle
&& last_bundle && last_bundle[0]
if (last_offer && _parse_bundle (last_offer, &last_bundle, NULL)
&& last_bundle && last_bundle && last_bundle[0]
&& _get_bundle_index (last_offer, last_bundle, &bundle_media_index)) {
bundle_ufrag =
g_strdup (_media_get_ice_ufrag (last_offer, bundle_media_index));
@ -2881,7 +2882,8 @@ _get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt,
/* TODO: use the options argument */
static GstSDPMessage *
_create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
_create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GError ** error)
{
GstSDPMessage *ret = NULL;
const GstWebRTCSessionDescription *pending_remote =
@ -2896,12 +2898,13 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
GstSDPMessage *last_answer = _get_latest_self_generated_sdp (webrtc);
if (!webrtc->pending_remote_description) {
GST_ERROR_OBJECT (webrtc,
g_set_error_literal (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_INVALID_STATE,
"Asked to create an answer without a remote description");
return NULL;
}
if (!_parse_bundle (pending_remote->sdp, &bundled))
if (!_parse_bundle (pending_remote->sdp, &bundled, error))
goto out;
if (bundled) {
@ -2909,8 +2912,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
guint bundle_media_index;
if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) {
GST_ERROR_OBJECT (webrtc, "Bundle tag is %s but no media found matching",
bundled[0]);
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Bundle tag is %s but no media found matching", bundled[0]);
goto out;
}
@ -2918,7 +2921,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options)
bundled_mids = g_string_new ("BUNDLE");
}
if (last_answer && _parse_bundle (last_answer, &last_bundle)
if (last_answer && _parse_bundle (last_answer, &last_bundle, NULL)
&& last_bundle && last_bundle[0]
&& _get_bundle_index (last_answer, last_bundle, &bundle_media_index)) {
bundle_ufrag =
@ -3305,14 +3308,15 @@ _create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data)
GstWebRTCSessionDescription *desc = NULL;
GstSDPMessage *sdp = NULL;
GstStructure *s = NULL;
GError *error = NULL;
GST_INFO_OBJECT (webrtc, "creating %s sdp with options %" GST_PTR_FORMAT,
gst_webrtc_sdp_type_to_string (data->type), data->options);
if (data->type == GST_WEBRTC_SDP_TYPE_OFFER)
sdp = _create_offer_task (webrtc, data->options);
sdp = _create_offer_task (webrtc, data->options, &error);
else if (data->type == GST_WEBRTC_SDP_TYPE_ANSWER)
sdp = _create_answer_task (webrtc, data->options);
sdp = _create_answer_task (webrtc, data->options, &error);
else {
g_assert_not_reached ();
goto out;
@ -3323,6 +3327,13 @@ _create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data)
s = gst_structure_new ("application/x-gst-promise",
gst_webrtc_sdp_type_to_string (data->type),
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, desc, NULL);
} else {
g_warn_if_fail (error != NULL);
GST_WARNING_OBJECT (webrtc, "returning error: %s",
error ? error->message : "Unknown");
s = gst_structure_new ("application/x-gstwebrtcbin-error",
"error", G_TYPE_ERROR, error, NULL);
g_clear_error (&error);
}
out:
@ -3775,7 +3786,7 @@ static void
_update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMessage * sdp, guint media_idx,
TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans,
GStrv bundled, guint bundle_idx)
GStrv bundled, guint bundle_idx, GError ** error)
{
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
GstWebRTCRTPTransceiverDirection prev_dir = rtp_trans->current_direction;
@ -3812,20 +3823,28 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
local_setup = _get_dtls_setup_from_media (local_media);
remote_setup = _get_dtls_setup_from_media (remote_media);
new_setup = _get_final_setup (local_setup, remote_setup);
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE)
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Cannot intersect direction attributes for media %u", media_idx);
return;
}
local_dir = _get_direction_from_media (local_media);
remote_dir = _get_direction_from_media (remote_media);
new_dir = _get_final_direction (local_dir, remote_dir);
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE)
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Cannot intersect dtls setup attributes for media %u", media_idx);
return;
}
if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
&& new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
&& prev_dir != new_dir) {
GST_FIXME_OBJECT (webrtc, "implement transceiver direction changes");
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
"transceiver direction changes are not implemented. Media %u",
media_idx);
return;
}
@ -4023,7 +4042,8 @@ _generate_data_channel_id (GstWebRTCBin * webrtc)
static void
_update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMessage * sdp, guint media_idx, TransportStream * stream)
const GstSDPMessage * sdp, guint media_idx, TransportStream * stream,
GError ** error)
{
const GstSDPMedia *local_media, *remote_media;
GstWebRTCDTLSSetup local_setup, remote_setup, new_setup;
@ -4042,8 +4062,11 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
local_setup = _get_dtls_setup_from_media (local_media);
remote_setup = _get_dtls_setup_from_media (remote_media);
new_setup = _get_final_setup (local_setup, remote_setup);
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE)
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Cannot intersect dtls setup for media %u", media_idx);
return;
}
/* data channel is always rtcp-muxed to avoid generating ICE candidates
* for RTCP */
@ -4052,8 +4075,12 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
local_port = _get_sctp_port_from_media (local_media);
remote_port = _get_sctp_port_from_media (local_media);
if (local_port == -1 || remote_port == -1)
if (local_port == -1 || remote_port == -1) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Could not find sctp port for media %u (local %i, remote %i)",
media_idx, local_port, remote_port);
return;
}
if (0 == (local_max_size =
_get_sctp_max_message_size_from_media (local_media)))
@ -4166,7 +4193,7 @@ done:
static gboolean
_update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
GstWebRTCSessionDescription * sdp)
GstWebRTCSessionDescription * sdp, GError ** error)
{
int i;
gboolean ret = FALSE;
@ -4177,14 +4204,14 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
/* FIXME: With some peers, it's possible we could have
* multiple bundles to deal with, although I've never seen one yet */
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
if (!_parse_bundle (sdp->sdp, &bundled))
if (!_parse_bundle (sdp->sdp, &bundled, error))
goto done;
if (bundled) {
if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) {
GST_ERROR_OBJECT (webrtc, "Bundle tag is %s but no media found matching",
bundled[0]);
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Bundle tag is %s but no media found matching", bundled[0]);
goto done;
}
@ -4235,7 +4262,8 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
if (source == SDP_LOCAL && sdp->type == GST_WEBRTC_SDP_TYPE_OFFER && !trans) {
GST_ERROR ("State mismatch. Could not find local transceiver by mline.");
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"State mismatch. Could not find local transceiver by mline %u", i);
goto done;
} else {
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
@ -4258,9 +4286,10 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
}
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
trans, bundled, bundle_idx);
trans, bundled, bundle_idx, error);
} else if (_message_media_is_datachannel (sdp->sdp, i)) {
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream);
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream,
error);
} else {
GST_ERROR_OBJECT (webrtc, "Unknown media type in SDP at index %u", i);
}
@ -4284,6 +4313,20 @@ done:
return ret;
}
static gboolean
check_transceivers_not_removed (GstWebRTCBin * webrtc,
GstWebRTCSessionDescription * previous, GstWebRTCSessionDescription * new)
{
if (!previous)
return TRUE;
if (gst_sdp_message_medias_len (previous->sdp) >
gst_sdp_message_medias_len (new->sdp))
return FALSE;
return TRUE;
}
struct set_description
{
GstPromise *promise;
@ -4291,6 +4334,30 @@ struct set_description
GstWebRTCSessionDescription *sdp;
};
static GstWebRTCSessionDescription *
get_previous_description (GstWebRTCBin * webrtc, SDPSource source,
GstWebRTCSDPType type)
{
switch (type) {
case GST_WEBRTC_SDP_TYPE_OFFER:
case GST_WEBRTC_SDP_TYPE_PRANSWER:
case GST_WEBRTC_SDP_TYPE_ANSWER:
if (source == SDP_LOCAL) {
return webrtc->current_local_description;
} else {
return webrtc->current_remote_description;
}
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
return NULL;
default:
/* other values mean memory corruption/uninitialized! */
g_assert_not_reached ();
break;
}
return NULL;
}
/* http://w3c.github.io/webrtc-pc/#set-description */
static void
_set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
@ -4316,29 +4383,31 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
g_free (type_str);
}
if (!validate_sdp (webrtc->signaling_state, sd->source, sd->sdp, &error)) {
GST_ERROR_OBJECT (webrtc, "%s", error->message);
g_clear_error (&error);
if (!validate_sdp (webrtc->signaling_state, sd->source, sd->sdp, &error))
goto out;
}
if (webrtc->priv->is_closed) {
GST_WARNING_OBJECT (webrtc, "we are closed");
goto out;
}
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
if (!_parse_bundle (sd->sdp->sdp, &bundled))
if (!_parse_bundle (sd->sdp->sdp, &bundled, &error))
goto out;
if (bundled) {
if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) {
GST_ERROR_OBJECT (webrtc, "Bundle tag is %s but no media found matching",
bundled[0]);
g_set_error (&error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Bundle tag is %s but no matching media found", bundled[0]);
goto out;
}
}
if (!check_transceivers_not_removed (webrtc,
get_previous_description (webrtc, sd->source, sd->sdp->type),
sd->sdp)) {
g_set_error_literal (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_BAD_SDP,
"m=lines removed from the SDP. Processing a completely new connection "
"is not currently supported.");
goto out;
}
switch (sd->sdp->type) {
case GST_WEBRTC_SDP_TYPE_OFFER:{
if (sd->source == SDP_LOCAL) {
@ -4484,7 +4553,8 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
GList *tmp;
/* media modifications */
_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp);
if (!_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp, &error))
goto out;
for (tmp = webrtc->priv->pending_sink_transceivers; tmp;) {
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (tmp->data);
@ -4664,7 +4734,15 @@ out:
g_strfreev (bundled);
PC_UNLOCK (webrtc);
gst_promise_reply (sd->promise, NULL);
if (error) {
GST_WARNING_OBJECT (webrtc, "returning error: %s", error->message);
gst_promise_reply (sd->promise,
gst_structure_new ("application/x-getwebrtcbin-error", "error",
G_TYPE_ERROR, error, NULL));
g_clear_error (&error);
} else {
gst_promise_reply (sd->promise, NULL);
}
PC_LOCK (webrtc);
}

1
ext/webrtc/utils.h
View file

@ -40,6 +40,7 @@ typedef enum
GST_WEBRTC_BIN_ERROR_SCTP_FAILURE,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
GST_WEBRTC_BIN_ERROR_CLOSED,
GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
} GstWebRTCError;
GstPadTemplate * _find_pad_template (GstElement * element,

7
ext/webrtc/webrtcsdp.c
View file

@ -872,7 +872,7 @@ _get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp, guint media_idx,
}
gboolean
_parse_bundle (GstSDPMessage * sdp, GStrv * bundled)
_parse_bundle (GstSDPMessage * sdp, GStrv * bundled, GError ** error)
{
const gchar *group;
gboolean ret = FALSE;
@ -883,8 +883,9 @@ _parse_bundle (GstSDPMessage * sdp, GStrv * bundled)
*bundled = g_strsplit (group + strlen ("BUNDLE "), " ", 0);
if (!(*bundled)[0]) {
GST_ERROR ("Invalid format for BUNDLE group, expected at least "
"one mid (%s)", group);
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
"Invalid format for BUNDLE group, expected at least one mid (%s)",
group);
g_strfreev (*bundled);
*bundled = NULL;
goto done;

3
ext/webrtc/webrtcsdp.h
View file

@ -101,7 +101,8 @@ gboolean _get_bundle_index (Gst
guint * idx);
G_GNUC_INTERNAL
gboolean _parse_bundle (GstSDPMessage * sdp,
GStrv * bundled);
GStrv * bundled,
GError ** error);
G_GNUC_INTERNAL
const gchar * _media_get_ice_pwd (const GstSDPMessage * msg,

183
tests/check/elements/webrtcbin.c
View file

@ -177,14 +177,19 @@ _on_answer_received (GstPromise * promise, gpointer user_data)
GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *answer = NULL;
gchar *desc;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
desc = gst_sdp_message_as_text (answer->sdp);
GST_INFO ("Created Answer: %s", desc);
g_free (desc);
if (gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL)) {
gchar *desc = gst_sdp_message_as_text (answer->sdp);
GST_INFO ("Created Answer: %s", desc);
g_free (desc);
} else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
GST_INFO ("Creating answer resulted in error: %s", error->message);
} else {
g_assert_not_reached ();
}
g_mutex_lock (&t->lock);
@ -196,15 +201,28 @@ _on_answer_received (GstPromise * promise, gpointer user_data)
}
gst_promise_unref (promise);
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);
if (error)
goto error;
if (t->answer_desc) {
promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
g_signal_emit_by_name (answerer, "set-local-description", t->answer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
g_signal_emit_by_name (offeror, "set-remote-description", t->answer_desc,
promise);
}
test_webrtc_signal_state_unlocked (t, STATE_ANSWER_CREATED);
g_mutex_unlock (&t->lock);
return;
error:
g_clear_error (&error);
if (t->state < STATE_ERROR)
test_webrtc_signal_state_unlocked (t, STATE_ERROR);
g_mutex_unlock (&t->lock);
return;
}
static void
@ -232,14 +250,19 @@ _on_offer_received (GstPromise * promise, gpointer user_data)
GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *offer = NULL;
gchar *desc;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
desc = gst_sdp_message_as_text (offer->sdp);
GST_INFO ("Created offer: %s", desc);
g_free (desc);
if (gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL)) {
gchar *desc = gst_sdp_message_as_text (offer->sdp);
GST_INFO ("Created offer: %s", desc);
g_free (desc);
} else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
GST_INFO ("Creating offer resulted in error: %s", error->message);
} else {
g_assert_not_reached ();
}
g_mutex_lock (&t->lock);
@ -251,18 +274,31 @@ _on_offer_received (GstPromise * promise, gpointer user_data)
}
gst_promise_unref (promise);
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);
if (error)
goto error;
promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
if (t->offer_desc) {
promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
g_signal_emit_by_name (offeror, "set-local-description", t->offer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
g_signal_emit_by_name (answerer, "set-remote-description", t->offer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
}
test_webrtc_signal_state_unlocked (t, STATE_OFFER_CREATED);
g_mutex_unlock (&t->lock);
return;
error:
g_clear_error (&error);
if (t->state < STATE_ERROR)
test_webrtc_signal_state_unlocked (t, STATE_ERROR);
g_mutex_unlock (&t->lock);
return;
}
static gboolean
@ -2053,7 +2089,7 @@ _check_bundle_tag (struct test_webrtc *t, GstElement * element,
GStrv expected = user_data;
guint i;
fail_unless (_parse_bundle (sd->sdp, &bundled));
fail_unless (_parse_bundle (sd->sdp, &bundled, NULL));
if (!bundled) {
fail_unless_equals_int (g_strv_length (expected), 0);
@ -2745,6 +2781,98 @@ GST_START_TEST (test_renego_transceiver_set_direction)
GST_END_TEST;
static void
offer_remove_last_media (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
guint i, n;
GstSDPMessage *new, *old;
const GstSDPOrigin *origin;
const GstSDPConnection *conn;
old = t->offer_desc->sdp;
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_new (&new));
origin = gst_sdp_message_get_origin (old);
conn = gst_sdp_message_get_connection (old);
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_version (new,
gst_sdp_message_get_version (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_origin (new,
origin->username, origin->sess_id, origin->sess_version,
origin->nettype, origin->addrtype, origin->addr));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_session_name (new,
gst_sdp_message_get_session_name (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_information (new,
gst_sdp_message_get_information (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_uri (new,
gst_sdp_message_get_uri (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_connection (new,
conn->nettype, conn->addrtype, conn->address, conn->ttl,
conn->addr_number));
n = gst_sdp_message_attributes_len (old);
for (i = 0; i < n; i++) {
const GstSDPAttribute *a = gst_sdp_message_get_attribute (old, i);
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_attribute (new,
a->key, a->value));
}
n = gst_sdp_message_medias_len (old);
fail_unless (n > 0);
for (i = 0; i < n - 1; i++) {
const GstSDPMedia *m = gst_sdp_message_get_media (old, i);
GstSDPMedia *new_m;
fail_unless_equals_int (GST_SDP_OK, gst_sdp_media_copy (m, &new_m));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_media (new, new_m));
gst_sdp_media_init (new_m);
gst_sdp_media_free (new_m);
}
gst_webrtc_session_description_free (t->offer_desc);
t->offer_desc = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER,
new);
}
static void
offer_set_produced_error (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
const GstStructure *reply;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
GST_INFO ("error produced: %s", error->message);
g_clear_error (&error);
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
GST_START_TEST (test_renego_lose_media_fails)
{
struct test_webrtc *t = create_audio_video_test ();
VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
/* check that removing an m=line will produce an error */
test_validate_sdp (t, &offer, &answer);
test_webrtc_reset_negotiation (t);
t->on_offer_created = offer_remove_last_media;
t->on_offer_set = offer_set_produced_error;
t->on_answer_created = NULL;
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -2785,6 +2913,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_bundle_renego_add_stream);
tcase_add_test (tc, test_bundle_max_compat_max_bundle_renego_add_stream);
tcase_add_test (tc, test_renego_transceiver_set_direction);
tcase_add_test (tc, test_renego_lose_media_fails);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);