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webrtc: move webrtc_kind_from_caps() to utils

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2220>
This commit is contained in:
Matthew Waters 2021-05-06 17:06:44 +10:00 committed by GStreamer Marge Bot
parent 90edef103c
commit 1470660976
3 changed files with 33 additions and 29 deletions
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35
ext/webrtc/gstwebrtcbin.c
View file

@ -3235,35 +3235,11 @@ _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 static gboolean
_update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans, _update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans,
const GstCaps * caps) const GstCaps * caps)
{ {
GstWebRTCKind kind = _kind_from_caps (caps); GstWebRTCKind kind = webrtc_kind_from_caps (caps);
if (trans->kind == kind) if (trans->kind == kind)
return TRUE; return TRUE;
@ -3583,7 +3559,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps)) if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps))
GST_WARNING_OBJECT (webrtc, GST_WARNING_OBJECT (webrtc,
"Trying to change transceiver %d kind from %d to %d", "Trying to change transceiver %d kind from %d to %d",
rtp_trans->mline, rtp_trans->kind, _kind_from_caps (answer_caps)); rtp_trans->mline, rtp_trans->kind,
webrtc_kind_from_caps (answer_caps));
if (!trans->do_nack) { if (!trans->do_nack) {
answer_caps = gst_caps_make_writable (answer_caps); answer_caps = gst_caps_make_writable (answer_caps);
@ -5656,7 +5633,7 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
if (webrtc->priv->sctp_transport) { if (webrtc->priv->sctp_transport) {
/* Let transport be the connection's [[SctpTransport]] slot. /* Let transport be the connection's [[SctpTransport]] slot.
* *
* If the [[DataChannelId]] slot is not null, transport is in * If the [[DataChannelId]] slot is not null, transport is in
* connected state and [[DataChannelId]] is greater or equal to the * connected state and [[DataChannelId]] is greater or equal to the
* transport's [[MaxChannels]] slot, throw an OperationError. * transport's [[MaxChannels]] slot, throw an OperationError.
*/ */
@ -6420,7 +6397,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
} }
if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) { if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
GstWebRTCKind kind = _kind_from_caps (caps); GstWebRTCKind kind = webrtc_kind_from_caps (caps);
if (trans->kind != kind) { if (trans->kind != kind) {
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for" GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
@ -6440,7 +6417,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
guint i; guint i;
if (caps) if (caps)
kind = _kind_from_caps (caps); kind = webrtc_kind_from_caps (caps);
for (i = 0; i < webrtc->priv->transceivers->len; i++) { for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *tmptrans = GstWebRTCRTPTransceiver *tmptrans =

25
ext/webrtc/utils.c
View file

@ -205,3 +205,28 @@ _rtp_caps_from_media (const GstSDPMedia * media)
return ret; return ret;
} }
GstWebRTCKind
webrtc_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;
}

2
ext/webrtc/utils.h
View file

@ -82,6 +82,8 @@ G_GNUC_INTERNAL
const gchar * _g_checksum_to_webrtc_string (GChecksumType type); const gchar * _g_checksum_to_webrtc_string (GChecksumType type);
G_GNUC_INTERNAL G_GNUC_INTERNAL
GstCaps * _rtp_caps_from_media (const GstSDPMedia * media); GstCaps * _rtp_caps_from_media (const GstSDPMedia * media);
G_GNUC_INTERNAL
GstWebRTCKind webrtc_kind_from_caps (const GstCaps * caps);
G_END_DECLS G_END_DECLS