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webrtc: Expose RTCError enum

Browse source 
The error codes not complying with the spec are now notified with the
GST_WEBRTC_ERROR_INTERNAL_FAILURE code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1485>
This commit is contained in:
Philippe Normand 2022-01-28 17:11:41 +00:00 committed by GStreamer Marge Bot
parent ea06439c39
commit 4254920b72
9 changed files with 184 additions and 123 deletions
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84
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -926,7 +926,7 @@ _execute_op (GstWebRTCBinTask * op)
if (op->promise) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"webrtcbin is closed. aborting execution.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
@ -1704,8 +1704,8 @@ _query_pad_caps (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * rtp_trans,
/* Only return an error if actual empty caps were returned from the query. */
if (gst_caps_is_empty (caps)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Caps negotiation on pad %s failed", GST_PAD_NAME (pad));
gst_clear_caps (&caps);
gst_caps_unref (filter);
@ -1840,9 +1840,9 @@ _find_codec_preferences (GstWebRTCBin * webrtc,
gst_clear_caps (&caps);
if (gst_caps_is_empty (intersection)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED,
"Caps negotiation on pad %s failed againt codec preferences",
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Caps negotiation on pad %s failed against codec preferences",
GST_PAD_NAME (pad));
gst_clear_caps (&intersection);
} else {
@ -2841,8 +2841,8 @@ _parse_extmap (GQuark field_id, const GValue * value, GError ** error)
if (!ret && error) {
gchar *val_str = gst_value_serialize (value);
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Invalid value for %s: %s", g_quark_to_string (field_id), val_str);
g_free (val_str);
}
@ -2886,8 +2886,8 @@ _dedup_extmap_field (GQuark field_id, const GValue * value, ExtmapData * data)
GST_ERROR
("extmap contains different values for id %s (%s != %s)",
g_quark_to_string (field_id), old_value, new_value);
g_set_error (data->error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED,
g_set_error (data->error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"extmap contains different values for id %s (%s != %s)",
g_quark_to_string (field_id), old_value, new_value);
data->ret = FALSE;
@ -3392,8 +3392,8 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
g_assert (!g_list_find (seen_transceivers, trans));
if (wtrans->mline_locked && trans->mline != media_idx) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Previous negotiatied transceiver %"
GST_PTR_FORMAT " with mid %s was in mline %d but transceiver"
" has locked mline %u", trans, trans->mid, media_idx,
@ -3414,8 +3414,8 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
if (g_hash_table_contains (all_mids, mid)) {
gst_sdp_media_free (media);
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_FAILED,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Duplicate mid %s when creating offer", mid);
goto cancel_offer;
}
@ -3459,7 +3459,7 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
if (trans->mid) {
if (g_hash_table_contains (all_mids, trans->mid)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_FAILED,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Duplicate mid %s when creating offer", trans->mid);
goto cancel_offer;
}
@ -3539,8 +3539,8 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
continue;
g_assert (wtrans->mline_locked);
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Tranceiver %" GST_PTR_FORMAT " with mid %s has locked mline %d"
" but the whole offer only has %u sections", trans, trans->mid,
trans->mline, media_idx);
@ -3779,8 +3779,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GstSDPMessage *last_answer = _get_latest_self_generated_sdp (webrtc);
if (!webrtc->pending_remote_description) {
g_set_error_literal (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_INVALID_STATE,
g_set_error_literal (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_STATE,
"Asked to create an answer without a remote description");
return NULL;
}
@ -3793,7 +3793,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
guint bundle_media_index;
if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no media found matching", bundled[0]);
goto out;
}
@ -4289,7 +4289,7 @@ gst_webrtc_bin_create_offer (GstWebRTCBin * webrtc,
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not create offer. webrtcbin is closed");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
@ -4313,7 +4313,7 @@ gst_webrtc_bin_create_answer (GstWebRTCBin * webrtc,
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not create answer. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
@ -4876,7 +4876,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect direction attributes for media %u", media_idx);
return;
}
@ -4885,7 +4885,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect dtls setup attributes for media %u", media_idx);
return;
}
@ -4893,8 +4893,8 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
&& new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
&& prev_dir != new_dir) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"transceiver direction changes are not implemented. Media %u",
media_idx);
return;
@ -5107,7 +5107,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
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) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect dtls setup for media %u", media_idx);
return;
}
@ -5120,7 +5120,7 @@ _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) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Could not find sctp port for media %u (local %i, remote %i)",
media_idx, local_port, remote_port);
return;
@ -5260,7 +5260,7 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
if (bundled) {
if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no media found matching", bundled[0]);
goto done;
}
@ -5312,7 +5312,7 @@ _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) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"State mismatch. Could not find local transceiver by mline %u", i);
goto done;
} else {
@ -5415,8 +5415,8 @@ check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
continue;
if (rtp_trans->mline != i) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line with mid %s is at position %d, but was locked to %d, "
"rejecting", rtp_trans->mid, i, rtp_trans->mline);
return FALSE;
@ -5425,8 +5425,8 @@ check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
if (rtp_trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio") &&
rtp_trans->kind != GST_WEBRTC_KIND_AUDIO) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line %d was locked to audio, but SDP has %s media", i,
gst_sdp_media_get_media (media));
return FALSE;
@ -5434,8 +5434,8 @@ check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video") &&
rtp_trans->kind != GST_WEBRTC_KIND_VIDEO) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line %d was locked to video, but SDP has %s media", i,
gst_sdp_media_get_media (media));
return FALSE;
@ -5511,7 +5511,7 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
if (bundled) {
if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) {
g_set_error (&error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no matching media found", bundled[0]);
goto out;
}
@ -5520,8 +5520,8 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
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,
g_set_error_literal (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"m=lines removed from the SDP. Processing a completely new connection "
"is not currently supported.");
goto out;
@ -5889,7 +5889,7 @@ gst_webrtc_bin_set_remote_description (GstWebRTCBin * webrtc,
(GstWebRTCBinFunc) _set_description_task, sd,
(GDestroyNotify) _free_set_description_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not set remote description. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
@ -5927,7 +5927,7 @@ gst_webrtc_bin_set_local_description (GstWebRTCBin * webrtc,
(GstWebRTCBinFunc) _set_description_task, sd,
(GDestroyNotify) _free_set_description_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not set local description. webrtcbin is closed");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
@ -6122,7 +6122,7 @@ gst_webrtc_bin_get_stats (GstWebRTCBin * webrtc, GstPad * pad,
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _get_stats_task,
stats, (GDestroyNotify) _free_get_stats, promise)) {
GError *error =
g_error_new (GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_CLOSED,
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not retrieve statistics. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);

6
subprojects/gst-plugins-bad/ext/webrtc/utils.c
View file

@ -26,12 +26,6 @@
#include "utils.h"
#include "gstwebrtcbin.h"
GQuark
gst_webrtc_bin_error_quark (void)
{
return g_quark_from_static_string ("gst-webrtc-bin-error-quark");
}
GstPadTemplate *
_find_pad_template (GstElement * element, GstPadDirection direction,
GstPadPresence presence, const gchar * name)

19
subprojects/gst-plugins-bad/ext/webrtc/utils.h
View file

@ -26,25 +26,6 @@
G_BEGIN_DECLS
#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
GQuark gst_webrtc_bin_error_quark (void);
typedef enum
{
GST_WEBRTC_BIN_ERROR_FAILED,
GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
GST_WEBRTC_BIN_ERROR_INVALID_STATE,
GST_WEBRTC_BIN_ERROR_BAD_SDP,
GST_WEBRTC_BIN_ERROR_FINGERPRINT,
GST_WEBRTC_BIN_ERROR_SCTP_FAILURE,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
GST_WEBRTC_BIN_ERROR_CLOSED,
GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED
} GstWebRTCError;
GstPadTemplate * _find_pad_template (GstElement * element,
GstPadDirection direction,
GstPadPresence presence,

61
subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c
View file

@ -413,8 +413,8 @@ _parse_control_packet (WebRTCDataChannel * channel, guint8 * data,
GST_INFO_OBJECT (channel, "Received channel open");
if (channel->parent.negotiated) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Data channel was signalled as negotiated already");
g_return_val_if_reached (GST_FLOW_ERROR);
}
@ -479,16 +479,15 @@ _parse_control_packet (WebRTCDataChannel * channel, guint8 * data,
ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer);
if (ret != GST_FLOW_OK) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Could not send ack packet");
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Could not send ack packet");
return ret;
}
return ret;
} else {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Unknown message type in control protocol");
return GST_FLOW_ERROR;
}
@ -497,8 +496,8 @@ parse_error:
{
g_free (label);
g_free (proto);
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to parse packet");
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Failed to parse packet");
g_return_val_if_reached (GST_FLOW_ERROR);
}
}
@ -550,14 +549,14 @@ _data_channel_have_sample (WebRTCDataChannel * channel, GstSample * sample,
buffer = gst_sample_get_buffer (sample);
if (!buffer) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "No buffer to handle");
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "No buffer to handle");
return GST_FLOW_ERROR;
}
receive = gst_sctp_buffer_get_receive_meta (buffer);
if (!receive) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"No SCTP Receive meta on the buffer");
return GST_FLOW_ERROR;
}
@ -566,8 +565,8 @@ _data_channel_have_sample (WebRTCDataChannel * channel, GstSample * sample,
case DATA_CHANNEL_PPID_WEBRTC_CONTROL:{
GstMapInfo info = GST_MAP_INFO_INIT;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
@ -580,8 +579,8 @@ _data_channel_have_sample (WebRTCDataChannel * channel, GstSample * sample,
case DATA_CHANNEL_PPID_WEBRTC_STRING_PARTIAL:{
GstMapInfo info = GST_MAP_INFO_INIT;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
@ -596,8 +595,8 @@ _data_channel_have_sample (WebRTCDataChannel * channel, GstSample * sample,
case DATA_CHANNEL_PPID_WEBRTC_BINARY_PARTIAL:{
struct map_info *info = g_new0 (struct map_info, 1);
if (!gst_buffer_map (buffer, &info->map_info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
@ -618,8 +617,8 @@ _data_channel_have_sample (WebRTCDataChannel * channel, GstSample * sample,
NULL);
break;
default:
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Unknown SCTP PPID %u received", receive->ppid);
ret = GST_FLOW_ERROR;
break;
@ -713,8 +712,8 @@ webrtc_data_channel_start_negotiation (WebRTCDataChannel * channel)
_channel_enqueue_task (channel, (ChannelTask) _emit_on_open, NULL, NULL);
} else {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Failed to send DCEP open packet");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
@ -765,8 +764,8 @@ webrtc_data_channel_send_data (GstWebRTCDataChannel * base_channel,
g_return_if_fail (data != NULL);
if (!_is_within_max_message_size (channel, size)) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Requested to send data that is too large");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL,
@ -795,8 +794,8 @@ webrtc_data_channel_send_data (GstWebRTCDataChannel * base_channel,
if (ret != GST_FLOW_OK) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to send data");
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Failed to send data");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
@ -826,8 +825,8 @@ webrtc_data_channel_send_string (GstWebRTCDataChannel * base_channel,
if (!_is_within_max_message_size (channel, size)) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Requested to send a string that is too large");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL,
@ -858,8 +857,8 @@ webrtc_data_channel_send_string (GstWebRTCDataChannel * base_channel,
if (ret != GST_FLOW_OK) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to send string");
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Failed to send string");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}

38
subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
View file

@ -84,8 +84,8 @@ _check_valid_state_for_sdp_change (GstWebRTCSignalingState state,
gchar *state_str = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
state);
gchar *type_str = _enum_value_to_string (GST_TYPE_WEBRTC_SDP_TYPE, type);
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_INVALID_STATE,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_STATE,
"Not in the correct state (%s) for setting %s %s description",
state_str, _sdp_source_to_string (source), type_str);
g_free (state_str);
@ -108,8 +108,8 @@ _check_sdp_crypto (SDPSource source, GstWebRTCSessionDescription * sdp,
key = gst_sdp_message_get_key (sdp->sdp);
if (!IS_EMPTY_SDP_ATTRIBUTE (key->data)) {
g_set_error_literal (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_BAD_SDP, "sdp contains a k line");
g_set_error_literal (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "sdp contains a k line");
return FALSE;
}
@ -122,8 +122,8 @@ _check_sdp_crypto (SDPSource source, GstWebRTCSessionDescription * sdp,
if (!IS_EMPTY_SDP_ATTRIBUTE (message_fingerprint)
&& !IS_EMPTY_SDP_ATTRIBUTE (media_fingerprint)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_FINGERPRINT,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_FINGERPRINT_FAILURE,
"No fingerprint lines in sdp for media %u", i);
return FALSE;
}
@ -132,8 +132,8 @@ _check_sdp_crypto (SDPSource source, GstWebRTCSessionDescription * sdp,
}
if (!IS_EMPTY_SDP_ATTRIBUTE (media_fingerprint)
&& g_strcmp0 (fingerprint, media_fingerprint) != 0) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_FINGERPRINT,
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_FINGERPRINT_FAILURE,
"Fingerprint in media %u differs from %s fingerprint. "
"\'%s\' != \'%s\'", i, message_fingerprint ? "global" : "previous",
fingerprint, media_fingerprint);
@ -178,8 +178,8 @@ static gboolean
_check_trickle_ice (GstSDPMessage * msg, GError ** error)
{
if (!_session_has_attribute_key_value (msg, "ice-options", "trickle")) {
g_set_error_literal (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error_literal (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"No required \'a=ice-options:trickle\' line in sdp");
}
return TRUE;
@ -204,7 +204,7 @@ _media_has_mid (const GstSDPMedia * media, guint media_idx, GError ** error)
{
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
if (IS_EMPTY_SDP_ATTRIBUTE (mid)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'mid\' attribute",
media_idx);
return FALSE;
@ -248,13 +248,13 @@ _media_has_setup (const GstSDPMedia * media, guint media_idx, GError ** error)
static const gchar *valid_setups[] = { "actpass", "active", "passive", NULL };
const gchar *setup = gst_sdp_media_get_attribute_val (media, "setup");
if (IS_EMPTY_SDP_ATTRIBUTE (setup)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'setup\' attribute",
media_idx);
return FALSE;
}
if (!g_strv_contains (valid_setups, setup)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u contains unknown \'setup\' attribute, \'%s\'", media_idx,
setup);
return FALSE;
@ -268,7 +268,7 @@ _media_has_dtls_id (const GstSDPMedia * media, guint media_idx, GError ** error)
{
const gchar *dtls_id = gst_sdp_media_get_attribute_val (media, "ice-pwd");
if (IS_EMPTY_SDP_ATTRIBUTE (dtls_id)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'dtls-id\' attribute",
media_idx);
return FALSE;
@ -307,13 +307,13 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
media_in_bundle = is_bundle
&& g_strv_contains ((const gchar **) group_members, mid);
if (!_media_get_ice_ufrag (sdp->sdp, i)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'ice-ufrag\' attribute",
i);
goto fail;
}
if (!_media_get_ice_pwd (sdp->sdp, i)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'ice-pwd\' attribute", i);
goto fail;
}
@ -327,7 +327,7 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
if (!bundle_ice_ufrag)
bundle_ice_ufrag = ice_ufrag;
else if (g_strcmp0 (bundle_ice_ufrag, ice_ufrag) != 0) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u has different ice-ufrag values in bundle. "
"%s != %s", i, bundle_ice_ufrag, ice_ufrag);
goto fail;
@ -335,7 +335,7 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
if (!bundle_ice_pwd) {
bundle_ice_pwd = ice_pwd;
} else if (g_strcmp0 (bundle_ice_pwd, ice_pwd) != 0) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u has different ice-pwd values in bundle. "
"%s != %s", i, bundle_ice_pwd, ice_pwd);
goto fail;
@ -884,7 +884,7 @@ _parse_bundle (GstSDPMessage * sdp, GStrv * bundled, GError ** error)
*bundled = g_strsplit (group + strlen ("BUNDLE "), " ", 0);
if (!(*bundled)[0]) {
g_set_error (error, GST_WEBRTC_BIN_ERROR, GST_WEBRTC_BIN_ERROR_BAD_SDP,
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Invalid format for BUNDLE group, expected at least one mid (%s)",
group);
g_strfreev (*bundled);

1
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/meson.build
View file

@ -7,6 +7,7 @@ webrtc_sources = files([
'rtptransceiver.c',
'datachannel.c',
'sctptransport.c',
'webrtc.c',
])
webrtc_headers = files([

35
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc.c Normal file
View file

@ -0,0 +1,35 @@
/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/webrtc/webrtc_fwd.h>
/**
* gst_webrtc_error_quark:
*
* Since: 1.20
*/
GQuark
gst_webrtc_error_quark (void)
{
return g_quark_from_static_string ("gst-webrtc-error-quark");
}

41
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -423,4 +423,45 @@ typedef enum /*<underscore_name=gst_webrtc_kind>*/
GST_WEBRTC_KIND_VIDEO,
} GstWebRTCKind;
GST_WEBRTC_API
GQuark gst_webrtc_error_quark (void);
/**
* GST_WEBRTC_ERROR:
*
* Since: 1.20
*/
#define GST_WEBRTC_ERROR gst_webrtc_error_quark ()
/**
* GstWebRTCError:
* @GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE: data-channel-failure
* @GST_WEBRTC_ERROR_DTLS_FAILURE: dtls-failure
* @GST_WEBRTC_ERROR_FINGERPRINT_FAILURE: fingerprint-failure
* @GST_WEBRTC_ERROR_SCTP_FAILURE: sctp-failure
* @GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR: sdp-syntax-error
* @GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE: hardware-encoder-not-available
* @GST_WEBRTC_ERROR_ENCODER_ERROR: encoder-error
* @GST_WEBRTC_ERROR_INVALID_STATE: invalid-state (part of WebIDL specification)
* @GST_WEBRTC_ERROR_INTERNAL_FAILURE: GStreamer-specific failure, not matching any other value from the specification
*
* See <https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype> for more information.
*
* Since: 1.20
*/
typedef enum /*<underscore_name=gst_webrtc_error>*/
{
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
GST_WEBRTC_ERROR_DTLS_FAILURE,
GST_WEBRTC_ERROR_FINGERPRINT_FAILURE,
GST_WEBRTC_ERROR_SCTP_FAILURE,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE,
GST_WEBRTC_ERROR_ENCODER_ERROR,
GST_WEBRTC_ERROR_INVALID_STATE,
GST_WEBRTC_ERROR_INTERNAL_FAILURE
} GstWebRTCError;
#endif /* __GST_WEBRTC_FWD_H__ */

22
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View file

@ -3862,8 +3862,11 @@ GST_START_TEST (test_reject_create_offer)
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION));
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless (g_str_match_string
("has locked mline 1 but the whole offer only has 0 sections",
error->message, FALSE));
g_clear_error (&error);
gst_promise_unref (promise);
@ -3933,8 +3936,12 @@ GST_START_TEST (test_reject_set_description)
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION));
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless (g_str_match_string
("m-line 0 was locked to audio, but SDP has audio media", error->message,
FALSE));
g_clear_error (&error);
fail_unless (s != NULL);
gst_promise_unref (promise);
@ -4129,8 +4136,11 @@ GST_START_TEST (test_codec_preferences_negotiation_sinkpad)
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED));
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless (g_str_match_string
("Caps negotiation on pad sink_0 failed against codec preferences",
error->message, FALSE));
g_clear_error (&error);
gst_promise_unref (promise);