From: Sangchul Lee <sc11.lee@samsung.com>
Date: Tue, 16 Jul 2024 08:45:44 +0000 (+0900)
Subject: webrtc_source_mediapacket: Revise the timing of invoking the nego. needed cb
X-Git-Tag: accepted/tizen/unified/20240718.143629^0
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F27%2F314627%2F3;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source_mediapacket: Revise the timing of invoking the nego. needed cb

Previously, in case of h264 format, pushing a h264 buffer is required to
invoke webrtcbin's negotiation needed callback. With setting the particular
property of rtph264pay, it'll be invoked before pushing the first buffer.

[Version] 1.1.16
[Issue Type] Improvement

Change-Id: I9fb47ded651886a9a3dfb23b68b57b787830ca2f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_source_private.h b/include/webrtc_source_private.h
index 7ee3e8fb..5f424ae9 100644
--- a/include/webrtc_source_private.h
+++ b/include/webrtc_source_private.h
@@ -95,6 +95,7 @@ bool _is_videoscale_needed(webrtc_s *webrtc, webrtc_gst_slot_s *source);
 bool _is_videorate_needed(webrtc_s *webrtc, webrtc_gst_slot_s *source);
 GstElement *_prepare_first_capsfilter(webrtc_s *webrtc, webrtc_gst_slot_s *source, int ssrc, int av_idx, GList **element_list, bool *encoded_support, GstCaps **caps_for_render);
 GstElement *_prepare_videorate(webrtc_gst_slot_s *source, int ssrc, GList **element_list);
+void _update_payloader_h264_set_outcaps_before_handling_buffer(GstElement *payloader);
 void _update_payloader_vp8_picture_id_mode(GstElement *payloader);
 int _create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source, int ssrc, bool need_capsfilter, GList **element_list, bool is_audio);
 GstCaps *_make_default_encoded_caps(webrtc_gst_slot_s *source, int ssrc, webrtc_ini_s *ini, gchar **media_type);
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index fe8ebf62..728e7394 100644
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    1.1.15
+Version:    1.1.16
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source_mediapacket.c b/src/webrtc_source_mediapacket.c
index ba7d4eea..a9dd5776 100644
--- a/src/webrtc_source_mediapacket.c
+++ b/src/webrtc_source_mediapacket.c
@@ -323,6 +323,7 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
 		goto error;
 	APPEND_ELEMENT(*element_list, payloader);
 
+	_update_payloader_h264_set_outcaps_before_handling_buffer(payloader);
 	_update_payloader_vp8_picture_id_mode(payloader);
 
 	if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, NULL)))
diff --git a/src/webrtc_source_private.c b/src/webrtc_source_private.c
index e5499874..44f119f2 100644
--- a/src/webrtc_source_private.c
+++ b/src/webrtc_source_private.c
@@ -887,6 +887,29 @@ GstElement *_prepare_videorate(webrtc_gst_slot_s *source, int ssrc, GList **elem
 	return videorateCapsfilter;
 }
 
+void _update_payloader_h264_set_outcaps_before_handling_buffer(GstElement *payloader)
+{
+	const gchar *factory_name;
+	const char *prop_name = "set-outcaps-before-handle-buffer";
+
+	ASSERT(payloader);
+
+	factory_name = GST_OBJECT_NAME(gst_element_get_factory(payloader));
+	if (g_strcmp0(factory_name, "rtph264pay"))
+		return;
+
+	if (!g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(payloader)), prop_name)) {
+		LOG_ERROR("could not find '%s', please check gstreamer rtph264pay", prop_name);
+		return;
+	}
+
+	g_object_set(G_OBJECT(payloader),
+		prop_name, TRUE,
+		NULL);
+
+	LOG_DEBUG("[%s] %s [TRUE]", factory_name, prop_name);
+}
+
 void _update_payloader_vp8_picture_id_mode(GstElement *payloader)
 {
 	const gchar *factory_name;