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webrtc: explicitly error out in a couple of renegotiation cases

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters 2021-11-16 17:28:04 +11:00 committed by GStreamer Marge Bot
parent 318a639e43
commit e28c45fd05
2 changed files with 29 additions and 8 deletions
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24
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View File

@ -3047,10 +3047,15 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
if (last_media) { if (last_media) {
const char *setup = gst_sdp_media_get_attribute_val (last_media, "setup"); const char *setup = gst_sdp_media_get_attribute_val (last_media, "setup");
if (setup) if (setup) {
gst_sdp_media_add_attribute (media, "setup", setup); gst_sdp_media_add_attribute (media, "setup", setup);
else } else {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
"media %u cannot renegotiate without an existing a=setup line",
media_idx);
return FALSE; return FALSE;
}
} else { } else {
/* mandated by JSEP */ /* mandated by JSEP */
gst_sdp_media_add_attribute (media, "setup", "actpass"); gst_sdp_media_add_attribute (media, "setup", "actpass");
@ -3179,7 +3184,16 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
/* Some identifier; we also add the media name to it so it's identifiable */ /* Some identifier; we also add the media name to it so it's identifiable */
if (trans->mid) { if (trans->mid) {
const char *media_mid = gst_sdp_media_get_attribute_val (media, "mid");
if (!media_mid) {
gst_sdp_media_add_attribute (media, "mid", trans->mid); gst_sdp_media_add_attribute (media, "mid", trans->mid);
} else if (g_strcmp0 (media_mid, trans->mid) != 0) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
"Cannot change media %u mid value from \'%s\' to \'%s\'",
media_idx, media_mid, trans->mid);
return FALSE;
}
mid = g_strdup (trans->mid); mid = g_strdup (trans->mid);
} else { } else {
const GstStructure *s = gst_caps_get_structure (caps, 0); const GstStructure *s = gst_caps_get_structure (caps, 0);
@ -4168,7 +4182,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
* that we cannot actually support */ * that we cannot actually support */
if (trans_caps) { if (trans_caps) {
answer_caps = gst_caps_intersect (offer_caps, trans_caps); answer_caps = gst_caps_intersect (offer_caps, trans_caps);
gst_caps_unref (trans_caps); gst_clear_caps (&trans_caps);
} else { } else {
answer_caps = gst_caps_ref (offer_caps); answer_caps = gst_caps_ref (offer_caps);
} }
@ -4180,8 +4194,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
if (gst_caps_is_empty (answer_caps)) { if (gst_caps_is_empty (answer_caps)) {
GST_WARNING_OBJECT (webrtc, "Could not create caps for media"); GST_WARNING_OBJECT (webrtc, "Could not create caps for media");
gst_caps_unref (answer_caps); gst_clear_caps (&answer_caps);
gst_caps_unref (offer_caps); gst_clear_caps (&offer_caps);
goto rejected; goto rejected;
} }

11
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h
View File

@ -450,6 +450,13 @@ GQuark gst_webrtc_error_quark (void);
* *
* Since: 1.20 * Since: 1.20
*/ */
/**
* GST_WEBRTC_ERROR_INVALID_MODIFICATION:
*
* invalid-modification (part of WebIDL specification)
*
* Since: 1.22
*/
typedef enum /*<underscore_name=gst_webrtc_error>*/ typedef enum /*<underscore_name=gst_webrtc_error>*/
{ {
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
@ -460,8 +467,8 @@ typedef enum /*<underscore_name=gst_webrtc_error>*/
GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE, GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE,
GST_WEBRTC_ERROR_ENCODER_ERROR, GST_WEBRTC_ERROR_ENCODER_ERROR,
GST_WEBRTC_ERROR_INVALID_STATE, GST_WEBRTC_ERROR_INVALID_STATE,
GST_WEBRTC_ERROR_INTERNAL_FAILURE GST_WEBRTC_ERROR_INTERNAL_FAILURE,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
} GstWebRTCError; } GstWebRTCError;
#endif /* __GST_WEBRTC_FWD_H__ */ #endif /* __GST_WEBRTC_FWD_H__ */