From: Sangchul Lee <sc11.lee@samsung.com>
Date: Tue, 20 Sep 2022 08:34:36 +0000 (+0900)
Subject: webrtc_source: Add encoding-params(2) to caps in case of OPUS codec
X-Git-Tag: submit/tizen_6.5/20220921.002546^0
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=2e3393e3c2970188c91b66e3a0073d8748c85431;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: Add encoding-params(2) to caps in case of OPUS codec

It represents channels.
In case of gst to gst, it worked fine without it. But in case of gst
to web, it does not work(web API failure).
It is also mandatory according to RFC7587, so it is added.

These codes are related to NULL source type with OPUS codec.

[Version] 0.2.182
[Issue Type] Improvement / Compatibility

Change-Id: I9df43b5f0091b78f4b9356d127b02ec1cc81eba0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 2ae3851f..ab09632c 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:    0.2.181
+Version:    0.2.182
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index cefde50f..65eb2b98 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -579,14 +579,22 @@ static GstCaps *__make_default_encoded_caps(webrtc_gst_slot_s *source, webrtc_in
 //LCOV_EXCL_START
 static GstCaps *__make_transceiver_caps(rtp_payload_info_s *payload_info, int payload_type)
 {
+	GstCaps *caps;
+
 	RET_VAL_IF(payload_info == NULL, NULL, "payload_info is NULL");
 
-	return gst_caps_new_simple("application/x-rtp",
+	caps = gst_caps_new_simple("application/x-rtp",
 				"media", G_TYPE_STRING, payload_info->media_type,
 				"encoding-name", G_TYPE_STRING, payload_info->encoding_name,
 				"clock-rate", G_TYPE_INT, payload_info->clock_rate,
 				"payload", G_TYPE_INT, payload_type,
 				NULL);
+	if (payload_info->codec == WEBRTC_TRANSCEIVER_CODEC_OPUS)
+		gst_caps_set_simple(caps,
+			"encoding-params", G_TYPE_STRING, "2",
+			NULL);
+
+	return caps;
 }
 
 /* Use g_free() to free the media_type parameter. */