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authorOlivier CrĂȘte <olivier.crete@collabora.com>2021-05-05 19:21:18 -0400
committerOlivier CrĂȘte <olivier.crete@collabora.com>2021-05-13 15:05:00 -0400
commit4bb94c69701ea18371d572510219c9bbd2f187c4 (patch)
treebf332af1b02ba4f1737a9df7dde9ee4f2b430e3a /ext/webrtc/gstwebrtcbin.c
parent2aa7efedd3ea3ab73eb7884ae569098418065e55 (diff)
downloadgstreamer-plugins-bad-4bb94c69701ea18371d572510219c9bbd2f187c4.tar.gz
webrtcbin: Remove dead code
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>
Diffstat (limited to 'ext/webrtc/gstwebrtcbin.c')
-rw-r--r--ext/webrtc/gstwebrtcbin.c21
1 files changed, 8 insertions, 13 deletions
diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index 52a4f4960..c08d83748 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -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 {