From: Sangchul Lee Date: Thu, 11 Apr 2024 08:51:19 +0000 (+0900) Subject: test: Add menu for webrtc_start_media_source() X-Git-Tag: accepted/tizen/unified/20240618.010058^0 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F45%2F311745%2F3;p=platform%2Fcore%2Fapi%2Fwebrtc.git test: Add menu for webrtc_start_media_source() [Version] 1.1.1 [Issue Type] Test application Change-Id: Ia6b07b0affa309e732d9b86fb8a989465ffcbb7d Signed-off-by: Sangchul Lee --- diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec index 0f54ce34..1ee63c1a 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: 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 da57b202..2632b9b2 100644 --- 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 c51e132d..34351555 100644 --- 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 2073eb2c..6d7a35f5 100644 --- 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,