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webrtcbin: explicitly use a variable for the rtp session idx

Browse source 
Slightly clearer in meaning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters 2021-07-20 20:17:13 +10:00 committed by GStreamer Marge Bot
parent 9a758d78a9
commit 79d58200c9
1 changed files with 11 additions and 9 deletions
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20
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
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@ -2976,6 +2976,7 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
gchar *direction, *ufrag, *pwd, *mid;
gboolean bundle_only;
guint rtp_session_idx;
GstCaps *caps;
GstStructure *extmap;
int i;
@ -2985,6 +2986,8 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
g_assert (trans->mline == -1 || trans->mline == media_idx);
rtp_session_idx = bundled_mids ? bundle_idx : media_idx;
bundle_only = bundled_mids && bundle_idx != media_idx
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE;
@ -3196,9 +3199,7 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
if (!trans->sender->transport) {
TransportStream *item;
item =
_get_or_create_transport_stream (webrtc,
bundled_mids ? bundle_idx : media_idx, FALSE);
item = _get_or_create_transport_stream (webrtc, rtp_session_idx, FALSE);
webrtc_transceiver_set_transport (WEBRTC_TRANSCEIVER (trans), item);
}
@ -5126,6 +5127,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
if (new_dir != prev_dir) {
gchar *prev_dir_s, *new_dir_s;
guint rtp_session_id = bundled ? bundle_idx : media_idx;
prev_dir_s =
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
@ -5166,6 +5168,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *pad =
_find_pad_for_transceiver (webrtc, GST_PAD_SINK, rtp_trans);
if (pad) {
GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT, pad, trans);
@ -5197,13 +5200,11 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
TransportStream *item;
item =
_get_or_create_transport_stream (webrtc,
bundled ? bundle_idx : media_idx, FALSE);
_get_or_create_transport_stream (webrtc, rtp_session_id, FALSE);
webrtc_transceiver_set_transport (trans, item);
}
_connect_output_stream (webrtc, trans->stream,
bundled ? bundle_idx : media_idx);
_connect_output_stream (webrtc, trans->stream, rtp_session_id);
/* delay adding the pad until rtpbin creates the recv output pad
* to ghost to so queries/events travel through the pipeline correctly
* as soon as the pad is added */
@ -5931,10 +5932,11 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp->sdp, i);
gchar *ufrag, *pwd;
TransportStream *item;
guint rtp_session_id = bundled ? bundle_idx : i;
item =
_get_or_create_transport_stream (webrtc, bundled ? bundle_idx : i,
_message_media_is_datachannel (sd->sdp->sdp, bundled ? bundle_idx : i));
_get_or_create_transport_stream (webrtc, rtp_session_id,
_message_media_is_datachannel (sd->sdp->sdp, rtp_session_id));
if (sd->source == SDP_REMOTE) {
guint j;