Rename internal defines for specific strings

MS_RTP_PAD_* are changed to MS_PAD_* because it is not only used
for RTP node but also used for WebRTC node.

MEDIA_STREAMER_PARAM_AUDIO[VIDEO]_IN_FORMAT is renamed to
MS_PARAM_AUDIO[VIDEO]_IN_FORMAT because these are not exported
defines.

Replace underline by hyphen in string values.

Typo is fixed - 'rctp' to 'rtcp'

[Version] 0.1.86
[Issue Type] Improvement

Change-Id: I6133494c688dfaade2a5bb60f7dcd654e5037f72
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/media_streamer_util.h b/include/media_streamer_util.h
index 08c2a01..503fabb 100644 (file)
--- a/include/media_streamer_util.h
+++ b/include/media_streamer_util.h
@@ -212,12 +212,12 @@ typedef struct _media_streamer_wl_info_s {
 #define MS_ELEMENT_IS_IMAGE(el) g_strrstr(el, "image")
 #define MS_ELEMENT_IS_TEXT(el) g_strrstr(el, "text")
 
-#define MS_RTP_PAD_VIDEO_IN "video_in"
-#define MS_RTP_PAD_AUDIO_IN "audio_in"
-#define MS_RTP_PAD_VIDEO_OUT "video_out"
-#define MS_RTP_PAD_AUDIO_OUT "audio_out"
-#define MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT MS_RTP_PAD_VIDEO_IN"_format"
-#define MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT MS_RTP_PAD_AUDIO_IN"_format"
+#define MS_PAD_VIDEO_IN "video-in"
+#define MS_PAD_AUDIO_IN "audio-in"
+#define MS_PAD_VIDEO_OUT "video-out"
+#define MS_PAD_AUDIO_OUT "audio-out"
+#define MS_PARAM_VIDEO_IN_FORMAT MS_PAD_VIDEO_IN"-format"
+#define MS_PARAM_AUDIO_IN_FORMAT MS_PAD_AUDIO_IN"-format"
 
 #define MS_ELEMENT_IS_RTP(el) g_strrstr(el, "rtp_container")
 #define MS_ELEMENT_IS_ENCODER(el) g_strrstr(el, "encoder")

diff --git a/packaging/capi-media-streamer.spec b/packaging/capi-media-streamer.spec
index 496007e..2c04aa9 100644 (file)
--- a/packaging/capi-media-streamer.spec
+++ b/packaging/capi-media-streamer.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-streamer
 Summary:    A Media Streamer API
-Version:    0.1.85
+Version:    0.1.86
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/media_streamer_gst.c b/src/media_streamer_gst.c
index 4d2aeff..afc6d5d 100644 (file)
--- a/src/media_streamer_gst.c
+++ b/src/media_streamer_gst.c
@@ -1818,13 +1818,13 @@ GstElement *ms_rtp_element_create(void)
        rtp_container = gst_bin_new("rtp_container");
        ms_retvm_if(!rtp_container, (GstElement *) NULL, "Error: creating elements for rtp container");
 
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_VIDEO_OUT, GST_PAD_SRC);
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_AUDIO_OUT, GST_PAD_SRC);
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_VIDEO_IN, GST_PAD_SINK);
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_AUDIO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_VIDEO_OUT, GST_PAD_SRC);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_AUDIO_OUT, GST_PAD_SRC);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_VIDEO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_AUDIO_IN, GST_PAD_SINK);
 
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_VIDEO_IN"_rtp", GST_PAD_SINK);
-       ms_add_no_target_ghostpad(rtp_container, MS_RTP_PAD_AUDIO_IN"_rtp", GST_PAD_SINK);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_VIDEO_IN"-rtp", GST_PAD_SINK);
+       ms_add_no_target_ghostpad(rtp_container, MS_PAD_AUDIO_IN"-rtp", GST_PAD_SINK);
 
        /* Add RTP node parameters as GObject data with destroy function */
        MS_SET_INT_PARAM(rtp_container, MEDIA_STREAMER_PARAM_VIDEO_IN_PORT, RTP_STREAM_DISABLED);
@@ -1832,8 +1832,8 @@ GstElement *ms_rtp_element_create(void)
        MS_SET_INT_PARAM(rtp_container, MEDIA_STREAMER_PARAM_VIDEO_OUT_PORT, RTP_STREAM_DISABLED);
        MS_SET_INT_PARAM(rtp_container, MEDIA_STREAMER_PARAM_AUDIO_OUT_PORT, RTP_STREAM_DISABLED);
        MS_SET_INT_STATIC_STRING_PARAM(rtp_container, MEDIA_STREAMER_PARAM_HOST, "localhost");
-       MS_SET_INT_CAPS_PARAM(rtp_container, MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT, gst_caps_new_any());
-       MS_SET_INT_CAPS_PARAM(rtp_container, MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT, gst_caps_new_any());
+       MS_SET_INT_CAPS_PARAM(rtp_container, MS_PARAM_VIDEO_IN_FORMAT, gst_caps_new_any());
+       MS_SET_INT_CAPS_PARAM(rtp_container, MS_PARAM_AUDIO_IN_FORMAT, gst_caps_new_any());
 
        ms_debug_fleave();
 

diff --git a/src/media_streamer_gst_webrtc.c b/src/media_streamer_gst_webrtc.c
index 8870512..425dd6e 100644 (file)
--- a/src/media_streamer_gst_webrtc.c
+++ b/src/media_streamer_gst_webrtc.c
@@ -541,9 +541,9 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
        ms_debug("type is [%s]", media_type);
 
        if (MS_ELEMENT_IS_VIDEO(media_type)) {
-               src_pad_name = MS_RTP_PAD_VIDEO_OUT;
+               src_pad_name = MS_PAD_VIDEO_OUT;
        } else if (MS_ELEMENT_IS_AUDIO(media_type)) {
-               src_pad_name = MS_RTP_PAD_AUDIO_OUT;
+               src_pad_name = MS_PAD_AUDIO_OUT;
        } else {
                ms_error("Not supported media type[%s]", media_type);
                return;
@@ -619,10 +619,10 @@ GstElement *ms_webrtc_element_create(void)
        webrtc_container = gst_bin_new("webrtc_container");
        ms_retvm_if(!webrtc_container, (GstElement *) NULL, "Failed to create webrtc container");
 
-       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_VIDEO_IN, GST_PAD_SINK);
-       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_VIDEO_OUT, GST_PAD_SRC);
-       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_AUDIO_IN, GST_PAD_SINK);
-       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_AUDIO_OUT, GST_PAD_SRC);
+       ms_add_no_target_ghostpad(webrtc_container, MS_PAD_VIDEO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(webrtc_container, MS_PAD_VIDEO_OUT, GST_PAD_SRC);
+       ms_add_no_target_ghostpad(webrtc_container, MS_PAD_AUDIO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(webrtc_container, MS_PAD_AUDIO_OUT, GST_PAD_SRC);
 
        MS_SET_INT_STATIC_STRING_PARAM(webrtc_container, MEDIA_STREAMER_PARAM_WEBRTC_PEER_TYPE, DEFAULT_WEBRTC_PEER);
        MS_SET_INT_STATIC_STRING_PARAM(webrtc_container, MEDIA_STREAMER_PARAM_WEBRTC_STUN_SERVER, DEFAULT_WEBRTC_STUN_SERVER);

diff --git a/src/media_streamer_node.c b/src/media_streamer_node.c
index f8a8826..92ca0a9 100644 (file)
--- a/src/media_streamer_node.c
+++ b/src/media_streamer_node.c
@@ -184,8 +184,8 @@ static int __ms_rtp_node_set_property(media_streamer_node_s *node, param_s *para
                g_value_unset(val);
                g_value_init(val, G_TYPE_STRING);
                g_value_set_string(val, param_value);
-       } else if (!strcmp(param->param_name, MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT) ||
-                          !strcmp(param->param_name, MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT)) {
+       } else if (!strcmp(param->param_name, MS_PARAM_VIDEO_IN_FORMAT) ||
+                          !strcmp(param->param_name, MS_PARAM_AUDIO_IN_FORMAT)) {
                GstCaps *caps = gst_caps_from_string(param_value);
                if (caps) {
                        g_value_unset(val);
@@ -1416,10 +1416,10 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
 
        val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_VIDEO_IN_PORT);
        if (g_value_get_int(val) > RTP_STREAM_DISABLED) {
-               rtp_el = ms_element_create("udpsrc", MS_RTP_PAD_VIDEO_IN"_rtp");
+               rtp_el = ms_element_create("udpsrc", MS_PAD_VIDEO_IN"-rtp");
                ms_bin_add_element(node->gst_element, rtp_el, FALSE);
 
-               rtcp_el = ms_element_create("udpsrc", MS_RTP_PAD_VIDEO_IN"_rctp");
+               rtcp_el = ms_element_create("udpsrc", MS_PAD_VIDEO_IN"-rtcp");
                ms_bin_add_element(node->gst_element, rtcp_el, FALSE);
 
                if (!gst_element_link_pads(rtp_el, "src", rtpbin, "recv_rtp_sink_0") ||
@@ -1431,16 +1431,16 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
                g_object_set_property(G_OBJECT(rtp_el), MEDIA_STREAMER_PARAM_PORT, val);
                g_object_set(G_OBJECT(rtcp_el), MEDIA_STREAMER_PARAM_PORT, (g_value_get_int(val) + 1), NULL);
 
-               val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT);
+               val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MS_PARAM_VIDEO_IN_FORMAT);
                g_object_set_property(G_OBJECT(rtp_el), "caps", val);
        }
 
        val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_AUDIO_IN_PORT);
        if (g_value_get_int(val) > RTP_STREAM_DISABLED) {
-               rtp_el = ms_element_create("udpsrc", MS_RTP_PAD_AUDIO_IN"_rtp");
+               rtp_el = ms_element_create("udpsrc", MS_PAD_AUDIO_IN"-rtp");
                ms_bin_add_element(node->gst_element, rtp_el, FALSE);
 
-               rtcp_el = ms_element_create("udpsrc", MS_RTP_PAD_AUDIO_IN"_rctp");
+               rtcp_el = ms_element_create("udpsrc", MS_PAD_AUDIO_IN"-rtcp");
                ms_bin_add_element(node->gst_element, rtcp_el, FALSE);
 
                if (!gst_element_link_pads(rtp_el, "src", rtpbin, "recv_rtp_sink_1") ||
@@ -1452,16 +1452,16 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
                g_object_set_property(G_OBJECT(rtp_el), MEDIA_STREAMER_PARAM_PORT, val);
                g_object_set(G_OBJECT(rtcp_el), MEDIA_STREAMER_PARAM_PORT, (g_value_get_int(val) + 1), NULL);
 
-               val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT);
+               val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MS_PARAM_AUDIO_IN_FORMAT);
                g_object_set_property(G_OBJECT(rtp_el), "caps", val);
        }
 
        val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_VIDEO_OUT_PORT);
        if (g_value_get_int(val) > RTP_STREAM_DISABLED) {
-               rtp_el = ms_element_create("udpsink", MS_RTP_PAD_VIDEO_OUT"_rtp");
+               rtp_el = ms_element_create("udpsink", MS_PAD_VIDEO_OUT"-rtp");
                ms_bin_add_element(node->gst_element, rtp_el, FALSE);
 
-               rtcp_el = ms_element_create("udpsink", MS_RTP_PAD_VIDEO_OUT"_rctp");
+               rtcp_el = ms_element_create("udpsink", MS_PAD_VIDEO_OUT"-rtcp");
                ms_bin_add_element(node->gst_element, rtcp_el, FALSE);
 
                video_filter = ms_element_create("capsfilter", NULL);
@@ -1473,10 +1473,10 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
 
                gst_element_link_pads(video_filter, "src", rtpbin, "send_rtp_sink_0");
 
-               ghost_pad = (GstGhostPad *)gst_element_get_static_pad(node->gst_element, MS_RTP_PAD_VIDEO_IN);
+               ghost_pad = (GstGhostPad *)gst_element_get_static_pad(node->gst_element, MS_PAD_VIDEO_IN);
                if (ghost_pad) {
                        if (gst_ghost_pad_set_target(ghost_pad, gst_element_get_static_pad(video_filter, "sink")))
-                               ms_info(" Capsfilter for [%s] in RTP is set and linked", MS_RTP_PAD_VIDEO_IN);
+                               ms_info(" Capsfilter for [%s] in RTP is set and linked", MS_PAD_VIDEO_IN);
                }
 
                if (!gst_element_link_pads(rtpbin, "send_rtp_src_0", rtp_el, "sink") ||
@@ -1495,10 +1495,10 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
 
        val = (GValue *)g_object_get_data(G_OBJECT(node->gst_element), MEDIA_STREAMER_PARAM_AUDIO_OUT_PORT);
        if (g_value_get_int(val) > RTP_STREAM_DISABLED) {
-               rtp_el = ms_element_create("udpsink", MS_RTP_PAD_AUDIO_OUT"_rtp");
+               rtp_el = ms_element_create("udpsink", MS_PAD_AUDIO_OUT"-rtp");
                ms_bin_add_element(node->gst_element, rtp_el, FALSE);
 
-               rtcp_el = ms_element_create("udpsink", MS_RTP_PAD_AUDIO_OUT"_rctp");
+               rtcp_el = ms_element_create("udpsink", MS_PAD_AUDIO_OUT"-rtcp");
                ms_bin_add_element(node->gst_element, rtcp_el, FALSE);
 
                audio_filter = ms_element_create("capsfilter", NULL);
@@ -1509,10 +1509,10 @@ int ms_rtp_node_prepare(media_streamer_node_s *node)
 
                gst_element_link_pads(audio_filter, "src", rtpbin, "send_rtp_sink_1");
 
-               ghost_pad = (GstGhostPad *)gst_element_get_static_pad(node->gst_element, MS_RTP_PAD_AUDIO_IN);
+               ghost_pad = (GstGhostPad *)gst_element_get_static_pad(node->gst_element, MS_PAD_AUDIO_IN);
                if (ghost_pad) {
                        if (gst_ghost_pad_set_target(ghost_pad, gst_element_get_static_pad(audio_filter, "sink")))
-                               ms_info(" Capsfilter for [%s] in RTP is set and linked", MS_RTP_PAD_AUDIO_IN);
+                               ms_info(" Capsfilter for [%s] in RTP is set and linked", MS_PAD_AUDIO_IN);
                }
 
                if (!gst_element_link_pads(rtpbin, "send_rtp_src_1", rtp_el, "sink") ||
@@ -1554,7 +1554,7 @@ static int __ms_webrtc_prepare_ghost_sink_pad(GstElement *webrtc_container, GstE
                g_strcmp0(capsfilter_name, _WEBRTC_VIDEO_CAPSFILTER))
                return MEDIA_STREAMER_ERROR_INVALID_PARAMETER;
 
-       pad_name = !g_strcmp0(capsfilter_name, _WEBRTC_AUDIO_CAPSFILTER) ? MS_RTP_PAD_AUDIO_IN : MS_RTP_PAD_VIDEO_IN;
+       pad_name = !g_strcmp0(capsfilter_name, _WEBRTC_AUDIO_CAPSFILTER) ? MS_PAD_AUDIO_IN : MS_PAD_VIDEO_IN;
 
        if (!(filter = ms_find_element_in_bin_by_name(webrtc_container, capsfilter_name))) {
                ms_debug("No need to export the ghost sink pad for [%s]", pad_name);
@@ -2016,19 +2016,19 @@ static int __ms_rtp_node_set_pad_format(media_streamer_node_s *node, const char
        ms_debug_fenter();
 
        /* It is needed to set 'application/x-rtp' for audio and video udpsrc */
-       if (g_strrstr(pad_name, MS_RTP_PAD_VIDEO_IN)) {
+       if (g_strrstr(pad_name, MS_PAD_VIDEO_IN)) {
                ret = media_format_get_video_info(fmt, &mime, NULL, NULL, NULL, NULL);
                if (MEDIA_FORMAT_ERROR_NONE == ret) {
                        rtp_caps_str = g_strdup_printf("application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=%s", ms_convert_mime_to_rtp_format(mime));
-                       param_s param = {MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT, MEDIA_STREAMER_PARAM_VIDEO_IN_FORMAT};
+                       param_s param = {MS_PARAM_VIDEO_IN_FORMAT, MS_PARAM_VIDEO_IN_FORMAT};
                        ret = ms_node_set_param_value(node, &param, rtp_caps_str);
                }
-       } else if (g_strrstr(pad_name, MS_RTP_PAD_AUDIO_IN)) {
+       } else if (g_strrstr(pad_name, MS_PAD_AUDIO_IN)) {
                int audio_channels, audio_samplerate;
                ret = media_format_get_audio_info(fmt, &mime, &audio_channels, &audio_samplerate, NULL, NULL);
                if (MEDIA_FORMAT_ERROR_NONE == ret) {
                        rtp_caps_str = g_strdup_printf("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1");
-                       param_s param = {MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT, MEDIA_STREAMER_PARAM_AUDIO_IN_FORMAT};
+                       param_s param = {MS_PARAM_AUDIO_IN_FORMAT, MS_PARAM_AUDIO_IN_FORMAT};
                        ret = ms_node_set_param_value(node, &param, rtp_caps_str);
                }
        } else {
@@ -2059,17 +2059,17 @@ static int __ms_webrtc_node_set_pad_format(media_streamer_node_s *node, const ch
 
        ms_debug_fenter();
 
-       if (!g_strrstr(pad_name, MS_RTP_PAD_VIDEO_IN) && !g_strrstr(pad_name, MS_RTP_PAD_AUDIO_IN)) {
+       if (!g_strrstr(pad_name, MS_PAD_VIDEO_IN) && !g_strrstr(pad_name, MS_PAD_AUDIO_IN)) {
                ms_error("Not supported pad_name(%s)", pad_name);
                return MEDIA_STREAMER_ERROR_INVALID_PARAMETER;
        }
 
-       if (g_strrstr(pad_name, MS_RTP_PAD_VIDEO_IN) &&
+       if (g_strrstr(pad_name, MS_PAD_VIDEO_IN) &&
                !media_format_get_video_info(fmt, &mime, NULL, NULL, NULL, NULL)) {
                media = "video";
                payload = 96;
                capsfilter_name = _WEBRTC_VIDEO_CAPSFILTER;
-       } else if (g_strrstr(pad_name, MS_RTP_PAD_AUDIO_IN) &&
+       } else if (g_strrstr(pad_name, MS_PAD_AUDIO_IN) &&
                !media_format_get_audio_info(fmt, &mime, NULL, NULL, NULL, NULL)) {
                media = "audio";
                payload = 97;

diff --git a/test/media_streamer_test.c b/test/media_streamer_test.c
index e461bfd..4d9a176 100644 (file)
--- a/test/media_streamer_test.c
+++ b/test/media_streamer_test.c
@@ -628,19 +628,18 @@ static void start_webrtc_pipeline()
        media_streamer_node_link(video_queue0, "src", video_enc, "sink");
        media_streamer_node_link(video_enc, "src", video_pay, "sink");
        media_streamer_node_link(video_pay, "src", video_queue1, "sink");
-       media_streamer_node_link(video_queue1, "src",webrtc, "video_in");
+       media_streamer_node_link(video_queue1, "src", webrtc, MEDIA_STREAMER_NODE_PAD_VIDEO_SINK);
 
        /* Audio link */
        media_streamer_node_link(audio_src, "src", audio_queue0, "sink");
        media_streamer_node_link(audio_queue0, "src", audio_enc, "sink");
        media_streamer_node_link(audio_enc, "src", audio_pay, "sink");
        media_streamer_node_link(audio_pay, "src", audio_queue1, "sink");
-       media_streamer_node_link(audio_queue1, "src", webrtc, "audio_in");
-
+       media_streamer_node_link(audio_queue1, "src", webrtc, MEDIA_STREAMER_NODE_PAD_AUDIO_SINK);
 
        /* Note that setting pad format to WebRTC node is necessary, WebRTC use this format as media information for SDP */
-       media_streamer_node_set_pad_format(webrtc, "audio_in", afmt_opus);
-       media_streamer_node_set_pad_format(webrtc, "video_in", vfmt_vp8);
+       media_streamer_node_set_pad_format(webrtc, MEDIA_STREAMER_NODE_PAD_AUDIO_SINK, afmt_opus);
+       media_streamer_node_set_pad_format(webrtc, MEDIA_STREAMER_NODE_PAD_VIDEO_SINK, vfmt_vp8);
 
        g_webrtc = webrtc;
 }
@@ -964,8 +963,8 @@ static void set_rtp_params(media_streamer_node_h rtp_node, const char *ip, int v
        bundle_add_str(params, MEDIA_STREAMER_PARAM_HOST, ip);
 
        media_streamer_node_set_params(rtp_node, params);
-       media_streamer_node_set_pad_format(rtp_node, "video_in", vfmt_encoded);
-       media_streamer_node_set_pad_format(rtp_node, "audio_in", afmt_encoded);
+       media_streamer_node_set_pad_format(rtp_node, MEDIA_STREAMER_NODE_PAD_VIDEO_SINK, vfmt_encoded);
+       media_streamer_node_set_pad_format(rtp_node, MEDIA_STREAMER_NODE_PAD_AUDIO_SINK, afmt_encoded);
 
        bundle_free(params);
 }
@@ -1222,7 +1221,7 @@ static void _create_rtp_streamer(media_streamer_node_h rtp_bin)
                /*====================Linking Video Streamer=========================== */
                media_streamer_node_link(video_src, "src", video_enc, "sink");
                media_streamer_node_link(video_enc, "src", video_pay, "sink");
-               media_streamer_node_link(video_pay, "src", rtp_bin, "video_in");
+               media_streamer_node_link(video_pay, "src", rtp_bin, MEDIA_STREAMER_NODE_PAD_VIDEO_SINK);
                /*====================================================================== */
 
                g_print("== success streamer video part \n");
@@ -1255,7 +1254,7 @@ static void _create_rtp_streamer(media_streamer_node_h rtp_bin)
                /*====================Linking Audio Streamer========================== */
                media_streamer_node_link(audio_src, "src", audio_enc, "sink");
                media_streamer_node_link(audio_enc, "src", audio_pay, "sink");
-               media_streamer_node_link(audio_pay, "src", rtp_bin, "audio_in");
+               media_streamer_node_link(audio_pay, "src", rtp_bin, MEDIA_STREAMER_NODE_PAD_AUDIO_SINK);
                /*====================================================================== */
 
                g_print("== success streamer audio part \n");