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webrtcbin: Hold transceiver lock when accessing codec_preferences
This is required to allow the applications to modify the preferences. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
This commit is contained in:
parent
e9f14ed117
commit
dc6655542d
1 changed files with 72 additions and 59 deletions
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@ -1505,66 +1505,69 @@ _find_codec_preferences (GstWebRTCBin * webrtc,
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{
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WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans;
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GstCaps *ret = NULL;
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GstWebRTCBinPad *pad = NULL;
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GST_LOG_OBJECT (webrtc, "retrieving codec preferences from %" GST_PTR_FORMAT,
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trans);
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if (rtp_trans && rtp_trans->codec_preferences) {
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GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
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rtp_trans->codec_preferences);
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ret = gst_caps_ref (rtp_trans->codec_preferences);
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} else {
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GstWebRTCBinPad *pad = NULL;
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/* try to find a pad */
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if (!trans
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|| !(pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans)))
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pad = _find_pad_for_mline (webrtc, direction, media_idx);
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if (!pad) {
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if (trans && trans->last_configured_caps)
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ret = gst_caps_ref (trans->last_configured_caps);
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} else {
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GstCaps *caps = NULL;
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if (pad->received_caps) {
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caps = gst_caps_ref (pad->received_caps);
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} else if ((caps = gst_pad_get_current_caps (GST_PAD (pad)))) {
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GST_LOG_OBJECT (webrtc, "Using current pad caps: %" GST_PTR_FORMAT,
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caps);
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} else {
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static GstStaticCaps static_filter =
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
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GstCaps *filter = gst_static_caps_get (&static_filter);
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filter = gst_caps_make_writable (filter);
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if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
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gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
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else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
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gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
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caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
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GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT,
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caps);
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if (!gst_caps_is_fixed (caps) || gst_caps_is_equal (caps, filter)
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|| gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
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gst_caps_unref (caps);
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caps = NULL;
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}
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gst_caps_unref (filter);
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}
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if (caps) {
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if (trans)
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gst_caps_replace (&trans->last_configured_caps, caps);
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ret = caps;
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}
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gst_object_unref (pad);
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if (rtp_trans) {
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GST_OBJECT_LOCK (rtp_trans);
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if (rtp_trans->codec_preferences) {
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GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
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rtp_trans->codec_preferences);
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ret = gst_caps_ref (rtp_trans->codec_preferences);
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}
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GST_OBJECT_UNLOCK (rtp_trans);
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if (ret)
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return ret;
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}
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/* try to find a pad */
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if (!trans
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|| !(pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans)))
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pad = _find_pad_for_mline (webrtc, direction, media_idx);
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if (!pad) {
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if (trans && trans->last_configured_caps)
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ret = gst_caps_ref (trans->last_configured_caps);
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} else {
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GstCaps *caps = NULL;
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if (pad->received_caps) {
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caps = gst_caps_ref (pad->received_caps);
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} else if ((caps = gst_pad_get_current_caps (GST_PAD (pad)))) {
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GST_LOG_OBJECT (webrtc, "Using current pad caps: %" GST_PTR_FORMAT, caps);
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} else {
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static GstStaticCaps static_filter =
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
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GstCaps *filter = gst_static_caps_get (&static_filter);
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filter = gst_caps_make_writable (filter);
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if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
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gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
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else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
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gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
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caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
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GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT, caps);
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if (!gst_caps_is_fixed (caps) || gst_caps_is_equal (caps, filter)
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|| gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
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gst_caps_unref (caps);
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caps = NULL;
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}
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gst_caps_unref (filter);
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}
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if (caps) {
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if (trans)
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gst_caps_replace (&trans->last_configured_caps, caps);
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ret = caps;
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}
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gst_object_unref (pad);
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}
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if (!ret)
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@ -2697,6 +2700,7 @@ gather_reserved_pts (GstWebRTCBin * webrtc)
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GstWebRTCRTPTransceiver *trans;
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trans = g_ptr_array_index (webrtc->priv->transceivers, i);
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GST_OBJECT_LOCK (trans);
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if (trans->codec_preferences) {
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guint j, n;
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gint pt;
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@ -2711,6 +2715,7 @@ gather_reserved_pts (GstWebRTCBin * webrtc)
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}
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}
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}
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GST_OBJECT_UNLOCK (trans);
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}
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GST_OBJECT_UNLOCK (webrtc);
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@ -5497,7 +5502,9 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
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rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
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if (caps) {
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GST_OBJECT_LOCK (trans);
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rtp_trans->codec_preferences = gst_caps_ref (caps);
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GST_OBJECT_UNLOCK (trans);
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_update_transceiver_kind_from_caps (rtp_trans, caps);
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}
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@ -6403,14 +6410,17 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
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}
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if (caps) {
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GST_OBJECT_LOCK (trans);
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if (trans->codec_preferences &&
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!gst_caps_can_intersect (caps, trans->codec_preferences)) {
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GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
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" existing m-line %d, but requested caps %" GST_PTR_FORMAT
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" don't match existing codec preferences %" GST_PTR_FORMAT,
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name, serial, caps, trans->codec_preferences);
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GST_OBJECT_UNLOCK (trans);
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goto error_out;
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}
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GST_OBJECT_UNLOCK (trans);
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if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
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GstWebRTCKind kind = webrtc_kind_from_caps (caps);
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@ -6439,6 +6449,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
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GstWebRTCRTPTransceiver *tmptrans =
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g_ptr_array_index (webrtc->priv->transceivers, i);
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GstWebRTCBinPad *pad2;
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gboolean has_matching_caps;
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/* Ignore transceivers with a non-matching kind */
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if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN &&
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@ -6462,11 +6473,13 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
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continue;
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}
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GST_OBJECT_LOCK (tmptrans);
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has_matching_caps = (caps && tmptrans->codec_preferences &&
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!gst_caps_can_intersect (caps, tmptrans->codec_preferences));
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GST_OBJECT_UNLOCK (tmptrans);
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/* Ignore transceivers with non-matching caps */
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if (caps && tmptrans->codec_preferences &&
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!gst_caps_can_intersect (caps, tmptrans->codec_preferences)) {
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if (!has_matching_caps)
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continue;
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}
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trans = tmptrans;
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break;
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