From: Sangchul Lee Date: Thu, 7 Mar 2024 03:13:24 +0000 (+0900) Subject: test: Add menu for getting transceiver mid X-Git-Tag: accepted/tizen/8.0/unified/20240308.173324~1 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F00%2F307300%2F1;p=platform%2Fcore%2Fapi%2Fwebrtc.git test: Add menu for getting transceiver mid [Version] 0.4.50 [Issue Type] Test application Change-Id: I66e0c8824a0f7522cfeaaa5c6418988a584356e5 Signed-off-by: Sangchul Lee --- diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec index 71fea89b..4d1d9343 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: 0.4.49 +Version: 0.4.50 Release: 0 Group: Multimedia/API License: Apache-2.0 diff --git a/test/webrtc_test.c b/test/webrtc_test.c index dbc1006e..3fe2a75b 100644 --- a/test/webrtc_test.c +++ b/test/webrtc_test.c @@ -780,6 +780,21 @@ static void _webrtc_media_source_get_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_media_source_get_transceiver_mid(int index, unsigned int source_id, webrtc_media_type_e media_type) +{ + int ret = WEBRTC_ERROR_NONE; + gchar *mid; + + ret = webrtc_media_source_get_transceiver_mid(g_ad.conns[index].webrtc, source_id, media_type, &mid); + RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); + + g_print("webrtc_media_source_get_transceiver_mid() success, source_id[%u], media_type[%s], mid[%s]\n", + source_id, g_webrtc_media_type_str[media_type], mid); + + if (mid) + free(mid); +} + static void _webrtc_media_source_set_transceiver_codec(int index, unsigned int source_id, webrtc_media_type_e media_type, int value) { int ret; @@ -3717,6 +3732,24 @@ static void test_webrtc_media_source(char *cmd) } break; } + case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_MID: { + static unsigned int id; + static unsigned int media_type; + + switch (g_ad.input_count) { + case 0: + id = value; + g_ad.input_count++; + return; + case 1: + media_type = value - 1; + _webrtc_media_source_get_transceiver_mid(0, id, media_type); + id = media_type = 0; + g_ad.input_count = 0; + break; + } + break; + } } reset_menu_state(); diff --git a/test/webrtc_test_menu.c b/test/webrtc_test_menu.c index 93316380..9fbc657f 100644 --- a/test/webrtc_test_menu.c +++ b/test/webrtc_test_menu.c @@ -81,6 +81,7 @@ menu_info_s g_menu_infos[] = { { "ae", CURRENT_STATUS_MEDIA_SOURCE_ADD_TRANSCEIVER_ENCODING, true }, { "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 }, /* webrtc media render */ { "dt", CURRENT_STATUS_SET_DISPLAY_TYPE, true }, { "dm", CURRENT_STATUS_SET_DISPLAY_MODE, true }, @@ -254,6 +255,7 @@ void display_menu_main(void) g_print("ae. *Add transceiver encoding\t"); g_print("re. *Remove transceiver encoding\t"); g_print("te. *Active transceiver encoding\n"); + g_print("tm. *Get transceiver mid\n"); g_print("------------------------------------- Media Render --------------------------------------\n"); g_print("dt. Set display type\t"); g_print("dm. Set display mode\t"); @@ -355,7 +357,12 @@ void display_menu_webrtc_media_source(void) else if (get_appdata()->input_count == 2) g_print("*** input pause or play.(1:pause, 0:play)\n"); break; - case CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE : + case CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE: + case CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE: + case CURRENT_STATUS_MEDIA_SOURCE_GET_ENCODER_BITRATE: + case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION: + case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_CODEC: + case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_MID: if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); else if (get_appdata()->input_count == 1) @@ -369,12 +376,6 @@ void display_menu_webrtc_media_source(void) else if (get_appdata()->input_count == 2) g_print("*** input mute mode.(1:mute 0:unmute)\n"); break; - case CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE: - if (get_appdata()->input_count == 0) - g_print("*** input source id.\n"); - else if (get_appdata()->input_count == 1) - g_print("*** input media type.(1:audio 2:video)\n"); - break; case CURRENT_STATUS_MEDIA_SOURCE_SET_ENCODER_BITRATE: if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); @@ -383,12 +384,6 @@ void display_menu_webrtc_media_source(void) else if (get_appdata()->input_count == 2) g_print("*** input target bitrate.\n"); break; - case CURRENT_STATUS_MEDIA_SOURCE_GET_ENCODER_BITRATE: - if (get_appdata()->input_count == 0) - g_print("*** input source id.\n"); - else if (get_appdata()->input_count == 1) - g_print("*** input media type.(1:audio 2:video)\n"); - break; case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION: if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); @@ -426,12 +421,6 @@ void display_menu_webrtc_media_source(void) else if (get_appdata()->input_count == 2) g_print("*** input transceiver direction.(1:sendonly 2:recvonly 3:sendrecv)\n"); break; - case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION: - if (get_appdata()->input_count == 0) - g_print("*** input source id.\n"); - else if (get_appdata()->input_count == 1) - g_print("*** input media type.(1:audio 2:video)\n"); - break; case CURRENT_STATUS_MEDIA_SOURCE_FOREACH_SUPPORTED_TRANSCEIVER_CODEC: if (get_appdata()->input_count == 0) g_print("*** input media source type.(1:audiotest, 2:videotest, 3:mic, 4:camera, 5:screen, 6:file, 7:media packet, 8:null)\n"); @@ -446,12 +435,6 @@ void display_menu_webrtc_media_source(void) else if (get_appdata()->input_count == 2) g_print("*** input transceiver codec.(1:PCMU 2:PCMA 3:OPUS 4:VP8 5:VP9 6:H264)\n"); break; - case CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_CODEC: - if (get_appdata()->input_count == 0) - g_print("*** input source id.\n"); - else if (get_appdata()->input_count == 1) - g_print("*** input media type.(1:audio 2:video)\n"); - break; case CURRENT_STATUS_FILE_SOURCE_SET_PATH: if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); diff --git a/test/webrtc_test_priv.h b/test/webrtc_test_priv.h index 44195c14..172bd139 100644 --- a/test/webrtc_test_priv.h +++ b/test/webrtc_test_priv.h @@ -120,6 +120,7 @@ enum { CURRENT_STATUS_MEDIA_SOURCE_ADD_TRANSCEIVER_ENCODING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x21, 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, /* 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,