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Revert "webrtc: Save the media kind in the transceiver"

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This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
This commit is contained in:
Sebastian Dröge 2020-10-08 18:50:12 +03:00
parent a40d6f4994
commit cc7e98816f
3 changed files with 1 additions and 87 deletions
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69
ext/webrtc/gstwebrtcbin.c
View file

@ -2870,47 +2870,6 @@ _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)
@ -3221,10 +3180,6 @@ _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);
@ -3844,20 +3799,6 @@ _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);
@ -5119,10 +5060,8 @@ 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);
}
@ -5973,12 +5912,6 @@ 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);

2
gst-libs/gst/webrtc/rtptransceiver.h
View file

@ -38,7 +38,6 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
/**
* GstWebRTCRTPTransceiver:
* @kind: Type of media (Since: 1.20)
*/
struct _GstWebRTCRTPTransceiver
{
@ -54,7 +53,6 @@ struct _GstWebRTCRTPTransceiver
GstWebRTCRTPTransceiverDirection current_direction;
GstCaps *codec_preferences;
GstWebRTCKind kind;
gpointer _padding[GST_PADDING];
};

17
gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -373,21 +373,4 @@ 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__ */