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webrtcbin: Remove dead code

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The function is only called to create an offer, so no
need to pass the offer parameter and then check it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
This commit is contained in:
Olivier Crête 2021-05-05 19:21:18 -04:00
parent 2aa7efedd3
commit 4bb94c6970
1 changed files with 8 additions and 13 deletions
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21
ext/webrtc/gstwebrtcbin.c
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@ -2636,7 +2636,7 @@ _add_fingerprint_to_media (GstWebRTCDTLSTransport * transport,
/* based off https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-18#section-5.2.1 */
static gboolean
sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
GstWebRTCRTPTransceiver * trans, GstWebRTCSDPType type, guint media_idx,
GstWebRTCRTPTransceiver * trans, guint media_idx,
GString * bundled_mids, guint bundle_idx, gchar * bundle_ufrag,
gchar * bundle_pwd, GArray * reserved_pts, GHashTable * all_mids,
GError ** error)
@ -2714,14 +2714,9 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
gst_sdp_media_add_attribute (media, direction, "");
g_free (direction);
if (type == GST_WEBRTC_SDP_TYPE_OFFER) {
caps = _find_codec_preferences (webrtc, trans, media_idx, error);
caps =
_add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
caps);
} else {
g_assert_not_reached ();
}
caps = _find_codec_preferences (webrtc, trans, media_idx, error);
caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
caps);
if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
GST_WARNING_OBJECT (webrtc, "no caps available for transceiver, skipping");
@ -2746,7 +2741,7 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
gst_caps_unref (format);
}
if (type == GST_WEBRTC_SDP_TYPE_OFFER) {
{
const GstStructure *s = gst_caps_get_structure (caps, 0);
gint clockrate = -1;
gint rtx_target_pt;
@ -3229,9 +3224,9 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GST_LOG_OBJECT (webrtc, "adding transceiver %" GST_PTR_FORMAT " at media "
"index %u", trans, media_idx);
if (sdp_media_from_transceiver (webrtc, &media, trans,
GST_WEBRTC_SDP_TYPE_OFFER, media_idx, bundled_mids, 0, bundle_ufrag,
bundle_pwd, reserved_pts, all_mids, error)) {
if (sdp_media_from_transceiver (webrtc, &media, trans, media_idx,
bundled_mids, 0, bundle_ufrag, bundle_pwd, reserved_pts, all_mids,
error)) {
gst_sdp_message_add_media (ret, &media);
media_idx++;
} else {