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webrtc: Simplify fmtp handling in codec stats
Parsing the whole caps as SDP media only to retrieve the fmtp field afterwards seems a bit superfluous. By looking up the a-fmtp attribute directly the number of allocations in this function gets down a bit. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8125>
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1 changed files with 4 additions and 9 deletions
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@ -815,8 +815,8 @@ _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
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GstStructure *caps_s = gst_caps_get_structure (caps, 0);
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gint pt;
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const gchar *encoding_name, *media, *encoding_params;
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GstSDPMedia sdp_media = { 0 };
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guint channels = 0;
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const gchar *fmtp;
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if (gst_structure_get_int (caps_s, "payload", &pt))
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gst_structure_set (stats, "payload-type", G_TYPE_UINT, pt, NULL);
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@ -852,15 +852,10 @@ _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
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else
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gst_structure_set (stats, "codec-type", G_TYPE_STRING, "encode", NULL);
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gst_sdp_media_init (&sdp_media);
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if (gst_sdp_media_set_media_from_caps (caps, &sdp_media) == GST_SDP_OK) {
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const gchar *fmtp = gst_sdp_media_get_attribute_val (&sdp_media, "fmtp");
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fmtp = gst_structure_get_string (caps_s, "a-fmtp");
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if (fmtp) {
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gst_structure_set (stats, "sdp-fmtp-line", G_TYPE_STRING, fmtp, NULL);
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}
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}
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gst_sdp_media_uninit (&sdp_media);
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/* FIXME: transportId */
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}
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