From: Sangchul Lee <sc11.lee@samsung.com>
Date: Tue, 13 Sep 2022 07:12:25 +0000 (+0900)
Subject: webrtc_internal: Revise webrtc_media_source_set_payload_type() for media packet source
X-Git-Tag: accepted/tizen/unified/20220919.012611~2
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F33%2F281133%2F1;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_internal: Revise webrtc_media_source_set_payload_type() for media packet source

This function should affect media packet source. It is fixed.

[Version] 0.3.237
[Issue Type] Bug fix

Change-Id: I12b49da60526d2ba93cef07e1ceea9b4e887df33
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 86912306..1c71f0f8 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -730,6 +730,7 @@ int _build_mediapacketsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source);
 GstCaps *_make_mediapacketsrc_raw_caps_from_media_format(webrtc_gst_slot_s *source);
 int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h packet);
 int _set_media_format(webrtc_s *webrtc, unsigned int source_id, media_format_h format);
+int _update_pt_if_media_packet_source(webrtc_s *webrtc, webrtc_gst_slot_s *source);
 
 /* screen source */
 int _get_screen_resolution(int *width, int *height);
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 84ec88cd..bff8802f 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:    0.3.236
+Version:    0.3.237
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_internal.c b/src/webrtc_internal.c
index 45cd57c9..47735368 100644
--- a/src/webrtc_internal.c
+++ b/src/webrtc_internal.c
@@ -117,7 +117,7 @@ int webrtc_media_source_set_payload_type(webrtc_h webrtc, unsigned int source_id
 
 	LOG_INFO("webrtc[%p] source_id[%u] media_type[%u] payload type[%u]", _webrtc, source_id, media_type, pt);
 
-	return WEBRTC_ERROR_NONE;
+	return _update_pt_if_media_packet_source(webrtc, source);
 }
 
 int webrtc_media_source_get_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, unsigned int *pt)
diff --git a/src/webrtc_source_mediapacket.c b/src/webrtc_source_mediapacket.c
index e5af6fd9..74d5329d 100644
--- a/src/webrtc_source_mediapacket.c
+++ b/src/webrtc_source_mediapacket.c
@@ -836,3 +836,34 @@ error:
 	source->media_format = NULL;
 	return ret;
 }
+
+//LCOV_EXCL_START
+int _update_pt_if_media_packet_source(webrtc_s *webrtc, webrtc_gst_slot_s *source)
+{
+	GstCaps *sink_caps;
+	GstElement *capsfilter;
+	const char *media_type;
+	int idx;
+
+	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+	if (source->type != WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET)
+		return WEBRTC_ERROR_NONE;
+
+	if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER))) {
+		LOG_ERROR("failed to gst_bin_get_by_name() for %s", ELEMENT_NAME_RTP_CAPSFILTER);
+		return WEBRTC_ERROR_INVALID_OPERATION;
+	}
+
+	idx = GET_AV_IDX_BY_TYPE(source->media_types);
+	media_type = (idx == AV_IDX_AUDIO) ? _get_audio_media_type(source->av[idx].codec) : _get_video_media_type(source->av[idx].codec);
+
+	if (!(sink_caps = _make_rtp_caps(media_type, source->av[idx].pt, source, NULL)))
+		return WEBRTC_ERROR_INVALID_OPERATION;
+
+	g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
+	gst_caps_unref(sink_caps);
+
+	return WEBRTC_ERROR_NONE;
+}
+//LCOV_EXCL_STOP
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