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webrtc: Save the media kind in the transceiver
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
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7be09a5f22
3 changed files with 93 additions and 1 deletions
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@ -2858,6 +2858,47 @@ _media_add_rtx (GstSDPMedia * media, WebRTCTransceiver * trans,
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}
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}
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static GstWebRTCKind
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_kind_from_caps (const GstCaps * caps)
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{
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GstStructure *s;
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const gchar *media;
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if (gst_caps_get_size (caps) == 0)
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return GST_WEBRTC_KIND_UNKNOWN;
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s = gst_caps_get_structure (caps, 0);
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media = gst_structure_get_string (s, "media");
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if (media == NULL)
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return GST_WEBRTC_KIND_UNKNOWN;
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if (!g_strcmp0 (media, "audio"))
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return GST_WEBRTC_KIND_AUDIO;
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if (!g_strcmp0 (media, "video"))
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return GST_WEBRTC_KIND_VIDEO;
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return GST_WEBRTC_KIND_UNKNOWN;
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}
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static gboolean
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_update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans,
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const GstCaps * caps)
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{
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GstWebRTCKind kind = _kind_from_caps (caps);
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if (trans->kind == kind)
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return TRUE;
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if (trans->kind == GST_WEBRTC_KIND_UNKNOWN) {
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trans->kind = kind;
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return TRUE;
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} else {
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return FALSE;
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}
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}
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static void
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_get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt,
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guint * target_ssrc)
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@ -3168,6 +3209,10 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
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} else {
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trans = WEBRTC_TRANSCEIVER (rtp_trans);
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}
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if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps))
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GST_WARNING_OBJECT (webrtc,
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"Trying to change transceiver %d kind from %d to %d",
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rtp_trans->mline, rtp_trans->kind, _kind_from_caps (answer_caps));
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if (!trans->do_nack) {
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answer_caps = gst_caps_make_writable (answer_caps);
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@ -3787,6 +3832,20 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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rtp_trans->mline = media_idx;
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if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio")) {
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if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
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GST_FIXME_OBJECT (webrtc,
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"Updating video transceiver to audio, which isn't fully supported.");
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rtp_trans->kind = GST_WEBRTC_KIND_AUDIO;
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}
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if (!g_strcmp0 (gst_sdp_media_get_media (media), "video")) {
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if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
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GST_FIXME_OBJECT (webrtc,
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"Updating audio transceiver to video, which isn't fully supported.");
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rtp_trans->kind = GST_WEBRTC_KIND_VIDEO;
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}
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for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
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const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
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@ -5048,8 +5107,10 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
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"Created new unassociated transceiver %" GST_PTR_FORMAT, trans);
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rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
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if (caps)
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if (caps) {
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rtp_trans->codec_preferences = gst_caps_ref (caps);
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_update_transceiver_kind_from_caps (rtp_trans, caps);
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}
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return gst_object_ref (trans);
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}
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@ -5900,6 +5961,12 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
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}
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pad->trans = gst_object_ref (trans);
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if (caps && name && !_update_transceiver_kind_from_caps (trans, caps))
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GST_WARNING_OBJECT (webrtc,
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"Trying to create pad %s with caps %" GST_PTR_FORMAT
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" but transceiver %d already exists with a different"
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" media type", name, caps, serial);
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pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK |
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GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
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(GstPadProbeCallback) sink_pad_block, NULL, NULL);
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@ -61,6 +61,13 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
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*
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* Since: 1.16
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*/
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/**
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* GstWebRTCRTPTransceiver.kind:
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*
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* Type of media
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*
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* Since: 1.20
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*/
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struct _GstWebRTCRTPTransceiver
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{
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GstObject parent;
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@ -75,6 +82,7 @@ struct _GstWebRTCRTPTransceiver
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GstWebRTCRTPTransceiverDirection current_direction;
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GstCaps *codec_preferences;
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GstWebRTCKind kind;
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gpointer _padding[GST_PADDING];
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};
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@ -373,4 +373,21 @@ typedef enum /*<underscore_name=gst_webrtc_ice_transport_policy>*/
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GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY,
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} GstWebRTCICETransportPolicy;
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/**
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* GstWebRTCKind:
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* @GST_WEBRTC_KIND_UNKNOWN: Kind has not yet been set
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* @GST_WEBRTC_KIND_AUDIO: Kind is audio
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* @GST_WEBRTC_KIND_VIDEO: Kind is audio
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*
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* https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
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*
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* Since: 1.20
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*/
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typedef enum /*<underscore_name=gst_webrtc_kind>*/
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{
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GST_WEBRTC_KIND_UNKNOWN,
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GST_WEBRTC_KIND_AUDIO,
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GST_WEBRTC_KIND_VIDEO,
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} GstWebRTCKind;
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#endif /* __GST_WEBRTC_FWD_H__ */
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