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webrtcbin: Store pending mid to make create-offer idempotent
If the mid is not stored in the transceiver, but it is stored in last_offer, then a further create-offer call will just ignore that transceiver. Also include unit test for ensure it doesn't regress. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
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3c9e4f4886
commit
3503599e0a
4 changed files with 101 additions and 19 deletions
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@ -3213,7 +3213,7 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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* multiple dtls fingerprints https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-05
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*/
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GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
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gchar *direction, *ufrag, *pwd, *mid;
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gchar *direction, *ufrag, *pwd, *mid = NULL;
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gboolean bundle_only;
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guint rtp_session_idx;
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GstCaps *caps;
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@ -3422,11 +3422,13 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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return FALSE;
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}
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mid = g_strdup (trans->mid);
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} else {
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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}
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if (mid == NULL) {
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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mid = g_strdup (gst_structure_get_string (s, "a-mid"));
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if (mid) {
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_set_error (error, GST_WEBRTC_ERROR,
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@ -3436,19 +3438,39 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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media_idx);
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return FALSE;
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}
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g_free (WEBRTC_TRANSCEIVER (trans)->pending_mid);
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WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid);
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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} else {
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/* Make sure to avoid mid collisions */
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while (TRUE) {
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mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
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webrtc->priv->media_counter++);
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_free (mid);
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} else {
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gst_sdp_media_add_attribute (media, "mid", mid);
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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break;
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}
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}
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}
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if (mid == NULL) {
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mid = g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid);
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if (mid) {
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/* If it's already used, just ignore the pending one and generate
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* a new one */
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_clear_pointer (&mid, free);
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g_clear_pointer (&WEBRTC_TRANSCEIVER (trans)->pending_mid, free);
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} else {
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gst_sdp_media_add_attribute (media, "mid", mid);
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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}
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}
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}
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if (mid == NULL) {
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/* Make sure to avoid mid collisions */
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while (TRUE) {
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mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
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webrtc->priv->media_counter++);
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_free (mid);
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} else {
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gst_sdp_media_add_attribute (media, "mid", mid);
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid);
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break;
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}
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}
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}
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@ -3706,14 +3728,22 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
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|| g_strcmp0 (gst_sdp_media_get_media (last_media), "video") == 0) {
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const gchar *last_mid;
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int j;
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last_mid = gst_sdp_media_get_attribute_val (last_media, "mid");
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for (j = 0; j < webrtc->priv->transceivers->len; j++) {
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trans = g_ptr_array_index (webrtc->priv->transceivers, j);
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WebRTCTransceiver *wtrans;
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const gchar *mid;
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if (trans->mid && g_strcmp0 (trans->mid, last_mid) == 0) {
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WebRTCTransceiver *wtrans = WEBRTC_TRANSCEIVER (trans);
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const char *mid;
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trans = g_ptr_array_index (webrtc->priv->transceivers, j);
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wtrans = WEBRTC_TRANSCEIVER (trans);
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if (trans->mid)
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mid = trans->mid;
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else
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mid = wtrans->pending_mid;
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if (mid && g_strcmp0 (mid, last_mid) == 0) {
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GstSDPMedia media;
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memset (&media, 0, sizeof (media));
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@ -3800,6 +3830,11 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
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}
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g_hash_table_insert (all_mids, g_strdup (trans->mid), NULL);
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} else if (WEBRTC_TRANSCEIVER (trans)->pending_mid &&
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!g_hash_table_contains (all_mids,
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WEBRTC_TRANSCEIVER (trans)->pending_mid)) {
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g_hash_table_insert (all_mids,
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g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid), NULL);
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}
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}
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@ -160,6 +160,8 @@ webrtc_transceiver_finalize (GObject * object)
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gst_caps_replace (&trans->last_retrieved_caps, NULL);
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gst_caps_replace (&trans->last_send_configured_caps, NULL);
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g_free (trans->pending_mid);
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gst_event_replace (&trans->tos_event, NULL);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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@ -54,6 +54,8 @@ struct _WebRTCTransceiver
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*/
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GstCaps *last_send_configured_caps;
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gchar *pending_mid;
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gboolean mline_locked;
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GstElement *ulpfecdec;
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@ -5426,6 +5426,48 @@ GST_START_TEST (test_invalid_add_media_in_answer)
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GST_END_TEST;
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GST_START_TEST (test_data_channel_recreate_offer)
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{
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GstHarness *h;
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GstWebRTCDataChannel *channel;
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GstPromise *promise;
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const GstStructure *s;
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GstPromiseResult res;
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GstPad *pad;
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h = gst_harness_new_with_padnames ("webrtcbin", "sink_0", NULL);
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add_audio_test_src_harness (h, 0xDEADBEEF);
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g_signal_emit_by_name (h->element, "create-data-channel", "label", NULL,
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&channel);
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fail_unless (GST_IS_WEBRTC_DATA_CHANNEL (channel));
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pad = gst_element_get_static_pad (h->element, "sink_0");
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fail_unless (pad != NULL);
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promise = gst_promise_new ();
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g_signal_emit_by_name (h->element, "create-offer", NULL, promise);
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res = gst_promise_wait (promise);
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fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
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s = gst_promise_get_reply (promise);
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fail_unless (s != NULL);
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gst_promise_unref (promise);
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promise = gst_promise_new ();
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g_signal_emit_by_name (h->element, "create-offer", NULL, promise);
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res = gst_promise_wait (promise);
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fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
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s = gst_promise_get_reply (promise);
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fail_unless (s != NULL);
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gst_promise_unref (promise);
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gst_object_unref (pad);
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gst_object_unref (channel);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static Suite *
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webrtcbin_suite (void)
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{
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@ -5502,6 +5544,7 @@ webrtcbin_suite (void)
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tcase_add_test (tc, test_renego_stream_add_data_channel);
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tcase_add_test (tc, test_renego_data_channel_add_stream);
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tcase_add_test (tc, test_renego_stream_data_channel_add_stream);
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tcase_add_test (tc, test_data_channel_recreate_offer);
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} else {
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GST_WARNING ("Some required elements were not found. "
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"All datachannel tests are disabled. sctpenc %p, sctpdec %p", sctpenc,
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