diff options
| -rw-r--r-- | ext/webrtc/gstwebrtcbin.c | 69 | ||||
| -rw-r--r-- | gst-libs/gst/webrtc/rtptransceiver.h | 8 | ||||
| -rw-r--r-- | gst-libs/gst/webrtc/webrtc_fwd.h | 17 |
3 files changed, 93 insertions, 1 deletions
diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index a62c444d3..dbe919e53 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -2858,6 +2858,47 @@ _media_add_rtx (GstSDPMedia * media, WebRTCTransceiver * trans, } } +static GstWebRTCKind +_kind_from_caps (const GstCaps * caps) +{ + GstStructure *s; + const gchar *media; + + if (gst_caps_get_size (caps) == 0) + return GST_WEBRTC_KIND_UNKNOWN; + + s = gst_caps_get_structure (caps, 0); + + media = gst_structure_get_string (s, "media"); + if (media == NULL) + return GST_WEBRTC_KIND_UNKNOWN; + + if (!g_strcmp0 (media, "audio")) + return GST_WEBRTC_KIND_AUDIO; + + if (!g_strcmp0 (media, "video")) + return GST_WEBRTC_KIND_VIDEO; + + return GST_WEBRTC_KIND_UNKNOWN; +} + +static gboolean +_update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans, + const GstCaps * caps) +{ + GstWebRTCKind kind = _kind_from_caps (caps); + + if (trans->kind == kind) + return TRUE; + + if (trans->kind == GST_WEBRTC_KIND_UNKNOWN) { + trans->kind = kind; + return TRUE; + } else { + return FALSE; + } +} + static void _get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt, guint * target_ssrc) @@ -3168,6 +3209,10 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options, } else { trans = WEBRTC_TRANSCEIVER (rtp_trans); } + if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps)) + GST_WARNING_OBJECT (webrtc, + "Trying to change transceiver %d kind from %d to %d", + rtp_trans->mline, rtp_trans->kind, _kind_from_caps (answer_caps)); if (!trans->do_nack) { answer_caps = gst_caps_make_writable (answer_caps); @@ -3787,6 +3832,20 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc, rtp_trans->mline = media_idx; + if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio")) { + if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO) + GST_FIXME_OBJECT (webrtc, + "Updating video transceiver to audio, which isn't fully supported."); + rtp_trans->kind = GST_WEBRTC_KIND_AUDIO; + } + + if (!g_strcmp0 (gst_sdp_media_get_media (media), "video")) { + if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO) + GST_FIXME_OBJECT (webrtc, + "Updating audio transceiver to video, which isn't fully supported."); + rtp_trans->kind = GST_WEBRTC_KIND_VIDEO; + } + for (i = 0; i < gst_sdp_media_attributes_len (media); i++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i); @@ -5048,8 +5107,10 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc, "Created new unassociated transceiver %" GST_PTR_FORMAT, trans); rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans); - if (caps) + if (caps) { rtp_trans->codec_preferences = gst_caps_ref (caps); + _update_transceiver_kind_from_caps (rtp_trans, caps); + } return gst_object_ref (trans); } @@ -5900,6 +5961,12 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ, } pad->trans = gst_object_ref (trans); + if (caps && name && !_update_transceiver_kind_from_caps (trans, caps)) + GST_WARNING_OBJECT (webrtc, + "Trying to create pad %s with caps %" GST_PTR_FORMAT + " but transceiver %d already exists with a different" + " media type", name, caps, serial); + pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) sink_pad_block, NULL, NULL); diff --git a/gst-libs/gst/webrtc/rtptransceiver.h b/gst-libs/gst/webrtc/rtptransceiver.h index 73391f06b..a14dc4a7b 100644 --- a/gst-libs/gst/webrtc/rtptransceiver.h +++ b/gst-libs/gst/webrtc/rtptransceiver.h @@ -61,6 +61,13 @@ GType gst_webrtc_rtp_transceiver_get_type(void); * * Since: 1.16 */ +/** + * GstWebRTCRTPTransceiver.kind: + * + * Type of media + * + * Since: 1.20 + */ struct _GstWebRTCRTPTransceiver { GstObject parent; @@ -75,6 +82,7 @@ struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiverDirection current_direction; GstCaps *codec_preferences; + GstWebRTCKind kind; gpointer _padding[GST_PADDING]; }; diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h index 5c727d234..4de341231 100644 --- a/gst-libs/gst/webrtc/webrtc_fwd.h +++ b/gst-libs/gst/webrtc/webrtc_fwd.h @@ -373,4 +373,21 @@ typedef enum /*<underscore_name=gst_webrtc_ice_transport_policy>*/ GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY, } GstWebRTCICETransportPolicy; +/** + * GstWebRTCKind: + * @GST_WEBRTC_KIND_UNKNOWN: Kind has not yet been set + * @GST_WEBRTC_KIND_AUDIO: Kind is audio + * @GST_WEBRTC_KIND_VIDEO: Kind is audio + * + * https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind + * + * Since: 1.20 + */ +typedef enum /*<underscore_name=gst_webrtc_kind>*/ +{ + GST_WEBRTC_KIND_UNKNOWN, + GST_WEBRTC_KIND_AUDIO, + GST_WEBRTC_KIND_VIDEO, +} GstWebRTCKind; + #endif /* __GST_WEBRTC_FWD_H__ */ |