From: Sangchul Lee <sc11.lee@samsung.com>
Date: Fri, 17 Dec 2021 04:43:28 +0000 (+0900)
Subject: webrtc_test: Add menu to set/get RTP packet drop probability
X-Git-Tag: submit/tizen/20211229.072812~6
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=bd5d21d70b69e6548b75730e2150caf6c8a3676c;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_test: Add menu to set/get RTP packet drop probability

sdp. Set RTP packet drop probability
gdp. Get RTP packet drop probability

[Version] 0.3.29
[Issue Type] New feature

Change-Id: I40899c4948614e0b94fd2f8485335b38e21533ca
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index a4bdd979..a4e4faa0 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:    0.3.28
+Version:    0.3.29
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index 413ae4b2..d9a49f71 100644
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -91,6 +91,8 @@ enum {
 	CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY,
 	CURRENT_STATUS_SET_LOCAL_DESCRIPTION,
 	CURRENT_STATUS_SET_REMOTE_DESCRIPTION,
+	CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY,
+	CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY,
 	CURRENT_STATUS_SETTING_SIGNALING_SERVER,
 	CURRENT_STATUS_SETTING_PROXY,
 	CURRENT_STATUS_REQUEST_SESSION,
@@ -1431,6 +1433,27 @@ static void _webrtc_get_ice_transport_policy(int index)
 	g_print("webrtc_get_ice_transport_policy() success, policy[%d]\n", policy);
 }
 
+static void _webrtc_set_rtp_packet_drop_probability(bool sender, float probability)
+{
+	int ret = WEBRTC_ERROR_NONE;
+
+	ret = webrtc_set_rtp_packet_drop_probability(g_conns[0].webrtc, sender, probability);
+	RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+	g_print("webrtc_set_rtp_packet_drop_probability() success, sender[%u] probability[%f]\n", sender, probability);
+}
+
+static void _webrtc_get_rtp_packet_drop_probability(bool sender)
+{
+	int ret = WEBRTC_ERROR_NONE;
+	float probability;
+
+	ret = webrtc_get_rtp_packet_drop_probability(g_conns[0].webrtc, sender, &probability);
+	RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+	g_print("_webrtc_get_rtp_packet_drop_probability() success, sender[%u] probability[%f]\n", sender, probability);
+}
+
 static char * __get_channel_label(webrtc_data_channel_h channel)
 {
 	int ret = WEBRTC_ERROR_NONE;
@@ -4021,6 +4044,12 @@ void _interpret_main_menu(char *cmd)
 		} else if (strncmp(cmd, "gtp", 3) == 0) {
 			_webrtc_get_ice_transport_policy(0);
 
+		} else if (strncmp(cmd, "sdp", 3) == 0) {
+			g_menu_state = CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY;
+
+		} else if (strncmp(cmd, "gdp", 3) == 0) {
+			g_menu_state = CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY;
+
 		} else if (strncmp(cmd, "ssc", 3) == 0) {
 			g_menu_state = CURRENT_STATUS_CREATE_PRIVATE_SIGNALING_SERVER;
 
@@ -4146,6 +4175,8 @@ void display_sub_basic()
 	g_print("ual. Unset audio loopback\n");
 	g_print("vl. Set video loopback\t");
 	g_print("uvl. Unset video loopback\n");
+	g_print("sdp. *Set RTP packet drop probability\t");
+	g_print("gdp. *Get RTP packet drop probability\n");
 	g_print("------------------------------------- Data Channel --------------------------------------\n");
 	g_print("cd. Create data channel\t");
 	g_print("dd. Destroy data channel\n");
@@ -4363,6 +4394,16 @@ static void displaymenu()
 	} else if (g_menu_state == CURRENT_STATUS_SET_LOCAL_DESCRIPTION) {
 		g_print("*** input type of local description.(1:offer, 2:answer)\n");
 
+	} else if (g_menu_state == CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY) {
+		if (g_cnt == 0)
+			g_print("*** input side.(1:sender, 2:receiver)\n");
+		else if (g_cnt == 1)
+			g_print("*** input drop probability.(0 ~ 1.0)\n");
+
+	} else if (g_menu_state == CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY) {
+		if (g_cnt == 0)
+			g_print("*** input side.(1:sender, 2:receiver)\n");
+
 	} else if (g_menu_state == CURRENT_STATUS_SETTING_SIGNALING_SERVER) {
 		g_print("*** input signaling server URL.\n");
 
@@ -4717,6 +4758,41 @@ static void interpret(char *cmd)
 		reset_menu_state();
 		break;
 	}
+	case CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY: {
+		static bool sender;
+		switch (g_cnt) {
+		case 0:
+			value = atoi(cmd);
+			if (value != 1 && value != 2) {
+				g_print("invalid value[%d]\n", value);
+				reset_menu_state();
+				break;
+			}
+			sender = (value == 1);
+			g_cnt++;
+			break;
+		case 1: {
+			float fvalue = strtof(cmd, NULL);
+			_webrtc_set_rtp_packet_drop_probability(sender, fvalue);
+			sender = false;
+			g_cnt = 0;
+			reset_menu_state();
+			break;
+		}
+		}
+		break;
+	}
+	case CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY: {
+		value = atoi(cmd);
+		if (value != 1 && value != 2) {
+			g_print("invalid value[%d]\n", value);
+			reset_menu_state();
+			break;
+		}
+		_webrtc_get_rtp_packet_drop_probability((value == 1));
+		reset_menu_state();
+		break;
+	}
 	case CURRENT_STATUS_SETTING_SIGNALING_SERVER: {
 		_setting_uri(g_signaling_server.url, cmd);
 		reset_menu_state();