test: Add menu for webrtc_start_media_source()

[Version] 1.1.1
[Issue Type] Test application

Change-Id: Ia6b07b0affa309e732d9b86fb8a989465ffcbb7d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 0f54ce34b1f608b85df645697a12ff42a8141a6a..1ee63c1a45746740df04c2c12f21cff2fc36b2db 100644 (file)
--- 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:    1.1.0
+Version:    1.1.1
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index da57b202ab4d5fdce7f0cec0e585cfd57facdaf4..2632b9b247862573a4f06d98dcbe7912b9e96f92 100644 (file)
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -831,6 +831,14 @@ static void _webrtc_media_source_set_transceiver_codec(int index, unsigned int s
                source_id, g_webrtc_media_type_str[media_type], g_webrtc_transceiver_codec_str[codec]);
 }
 
+static void _webrtc_start_media_source(int index, unsigned int source_id)
+{
+       int ret = webrtc_start_media_source(g_ad.conns[index].webrtc, source_id);
+       RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+       g_print("webrtc_start_media_source() success, source_id[%u]\n", source_id);
+}
+
 void _webrtc_set_display_type(int index, int type)
 {
        g_ad.conns[index].render.display_type = type;
@@ -3769,6 +3777,9 @@ static void test_webrtc_media_source(char *cmd)
                }
                break;
        }
+       case CURRENT_STATUS_START_MEDIA_SOURCE:
+               _webrtc_start_media_source(0, value);
+               break;
        }
 
        reset_menu_state();

diff --git a/test/webrtc_test_menu.c b/test/webrtc_test_menu.c
index c51e132da5982d4cc13bd35974b6d207b9a5a745..34351555b4d87e1175f42daeaff0cbcceeb2cc33 100644 (file)
--- a/test/webrtc_test_menu.c
+++ b/test/webrtc_test_menu.c
@@ -82,6 +82,7 @@ menu_info_s g_menu_infos[] = {
        { "re", CURRENT_STATUS_MEDIA_SOURCE_REMOVE_TRANSCEIVER_ENCODING, true },
        { "te", CURRENT_STATUS_MEDIA_SOURCE_ACTIVE_TRANSCEIVER_ENCODING, true },
        { "tm", CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_MID, true },
+       { "ms", CURRENT_STATUS_START_MEDIA_SOURCE, true },
        /* webrtc media render */
        { "dt", CURRENT_STATUS_SET_DISPLAY_TYPE, true },
        { "dm", CURRENT_STATUS_SET_DISPLAY_MODE, true },
@@ -226,7 +227,8 @@ void display_menu_main(void)
        g_print("gdp. *Get RTP packet drop probability\n");
        g_print("------------------------------------- Media Source --------------------------------------\n");
        g_print("a. Add media source\t");
-       g_print("r. Remove media source\n");
+       g_print("r. Remove media source\t");
+       g_print("ms. Start media source\n");
        g_print("p. Pause/play media source\t");
        g_print("o. Get the media source pause\n");
        g_print("mu. Mute/unmute media source\t");
@@ -351,7 +353,8 @@ void display_menu_webrtc_media_source(void)
                        g_print("*** input value to enable AEC. (1:enable 0:disable)\n");
                break;
        case CURRENT_STATUS_REMOVE_MEDIA_SOURCE:
-               g_print("*** input media source id to remove.\n");
+       case CURRENT_STATUS_START_MEDIA_SOURCE:
+               g_print("*** input media source id.\n");
                break;
        case CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE:
                if (get_appdata()->input_count == 0)

diff --git a/test/webrtc_test_priv.h b/test/webrtc_test_priv.h
index 2073eb2ceccf2365495572015fd6a64af6ce210d..6d7a35f5ffe4441bd3a714af751b47fe0f127dc4 100644 (file)
--- a/test/webrtc_test_priv.h
+++ b/test/webrtc_test_priv.h
@@ -121,6 +121,7 @@ enum {
        CURRENT_STATUS_MEDIA_SOURCE_REMOVE_TRANSCEIVER_ENCODING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x22,
        CURRENT_STATUS_MEDIA_SOURCE_ACTIVE_TRANSCEIVER_ENCODING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x23,
        CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_MID = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x24,
+       CURRENT_STATUS_START_MEDIA_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x25,
        /* webrtc media render */
        CURRENT_STATUS_SET_DISPLAY_TYPE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x01,
        CURRENT_STATUS_SET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x02,