From: Sangchul Lee Date: Fri, 27 May 2022 11:12:07 +0000 (+0900) Subject: webrtc_test: Remove global variables X-Git-Tag: submit/tizen/20220607.060007^0 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F33%2F275633%2F8;p=platform%2Fcore%2Fapi%2Fwebrtc.git webrtc_test: Remove global variables Some global variables are put inside to app data structure. get_appdata() is added. [Version] 0.3.116 [Issue Type] Refactoring Change-Id: I12ee6a29a5ae8a1b4b1ee1db9c8ca885eef3b56a Signed-off-by: Sangchul Lee --- diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec index dbd24a5d..381bef04 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: 0.3.115 +Version: 0.3.116 Release: 0 Group: Multimedia/API License: Apache-2.0 diff --git a/test/webrtc_test.c b/test/webrtc_test.c index acd96f06..8edb50d7 100644 --- a/test/webrtc_test.c +++ b/test/webrtc_test.c @@ -70,18 +70,7 @@ static const char *g_webrtc_stats_type_str[] = { [WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP] = "remote-outbound-rtp", }; -int g_menu_status; -int g_cnt; -gchar g_proxy[MAX_STRING_LEN]; -appdata_s g_ad; -connection_s g_conns[MAX_CONNECTION_LEN]; -signaling_server_s g_signaling_server; -static webrtc_signaling_server_h g_inner_signaling_server; - -/* for video display */ -static Evas_Object *g_win_id; -static Evas_Object *g_eo_mine; -static Evas_Object *g_text_eo_mine; +static appdata_s g_ad; #if defined(__DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__) || defined(__DEBUG_VALIDATE_ENCODED_FRAME_CB__) GstBuffer *__alloc_buffer_from_packet(media_packet_h packet); @@ -94,6 +83,11 @@ static GstElement* __build_appsrc_vp8_render_pipeline(GstElement **appsrc); static GstElement* __build_appsrc_opus_render_pipeline(GstElement **appsrc); #endif +appdata_s *get_appdata(void) +{ + return &g_ad; +} + static void win_del(void *data, Evas_Object *obj, void *event) { elm_exit(); @@ -205,7 +199,7 @@ static int app_create(void *data) if (win == NULL) return -1; ad->win = win; - g_win_id = win; + ad->win_id = win; create_render_rect_and_bg(ad->win); /* Create evas image object for EVAS surface. @@ -215,7 +209,7 @@ static int app_create(void *data) * | eo (remote1) | eo (remote2) | * * |____________________|____________________| */ for (i = 0; i < MAX_CONNECTION_LEN + 1; i++) { - eo = (i == 0) ? &g_eo_mine : &g_conns[i - 1].render.eo; + eo = (i == 0) ? &g_ad.eo_mine : &g_ad.conns[i - 1].render.eo; *eo = create_image_object(ad->win); if (!*eo) { g_print("failed to create evas image object\n"); @@ -288,24 +282,24 @@ static int app_terminate(void *data) int i; for (i = 0; i < MAX_CONNECTION_LEN; i++) { - if (g_conns[i].render.eo) { - evas_object_del(g_conns[i].render.eo); - g_conns[i].render.eo = NULL; + if (g_ad.conns[i].render.eo) { + evas_object_del(g_ad.conns[i].render.eo); + g_ad.conns[i].render.eo = NULL; } - if (g_conns[i].render.text_eo) { - evas_object_del(g_conns[i].render.text_eo); - g_conns[i].render.text_eo = NULL; + if (g_ad.conns[i].render.text_eo) { + evas_object_del(g_ad.conns[i].render.text_eo); + g_ad.conns[i].render.text_eo = NULL; } } - if (g_eo_mine) { - evas_object_del(g_eo_mine); - g_eo_mine = NULL; + if (g_ad.eo_mine) { + evas_object_del(g_ad.eo_mine); + g_ad.eo_mine = NULL; } - if (g_win_id) { - evas_object_del(g_win_id); - g_win_id = NULL; + if (ad->win_id) { + evas_object_del(ad->win_id); + ad->win_id = NULL; } ad->win = NULL; @@ -339,7 +333,7 @@ static void __release_packet_source(int conn_idx, unsigned int source_idx) if (source_idx >= MAX_MEDIA_PACKET_SOURCE_LEN) return; - media_packet_source_s *packet_source = &g_conns[conn_idx].packet_sources[source_idx]; + media_packet_source_s *packet_source = &g_ad.conns[conn_idx].packet_sources[source_idx]; packet_source->source_id = 0; if (packet_source->format) { @@ -375,13 +369,13 @@ static void _webrtc_create(int index) { int ret = WEBRTC_ERROR_NONE; - RET_IF(g_conns[index].webrtc, "already created"); + RET_IF(g_ad.conns[index].webrtc, "already created"); - ret = webrtc_create(&g_conns[index].webrtc); + ret = webrtc_create(&g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - g_conns[index].index = index; - g_print("webrtc[%p, index:%d] is created\n", g_conns[index].webrtc, index); + g_ad.conns[index].index = index; + g_print("webrtc[%p, index:%d] is created\n", g_ad.conns[index].webrtc, index); } static void _webrtc_destroy(int index) @@ -389,63 +383,63 @@ static void _webrtc_destroy(int index) int ret = WEBRTC_ERROR_NONE; int i; - RET_IF(!g_conns[index].webrtc, "webrtc is NULL"); + RET_IF(!g_ad.conns[index].webrtc, "webrtc is NULL"); for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i]) { - webrtc_destroy_data_channel(g_conns[index].channels[i]); - g_conns[index].channels[i] = NULL; + if (g_ad.conns[index].channels[i]) { + webrtc_destroy_data_channel(g_ad.conns[index].channels[i]); + g_ad.conns[index].channels[i] = NULL; } - g_conns[index].recv_channels[i] = NULL; + g_ad.conns[index].recv_channels[i] = NULL; } - ret = webrtc_destroy(g_conns[index].webrtc); + ret = webrtc_destroy(g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - g_print("webrtc[%p, index:%d] is destroyed\n", g_conns[index].webrtc, index); + g_print("webrtc[%p, index:%d] is destroyed\n", g_ad.conns[index].webrtc, index); - g_conns[index].webrtc = NULL; - if (g_conns[index].offer) { - free(g_conns[index].offer); - g_conns[index].offer = NULL; + g_ad.conns[index].webrtc = NULL; + if (g_ad.conns[index].offer) { + free(g_ad.conns[index].offer); + g_ad.conns[index].offer = NULL; } - if (g_conns[index].answer) { - free(g_conns[index].answer); - g_conns[index].answer = NULL; + if (g_ad.conns[index].answer) { + free(g_ad.conns[index].answer); + g_ad.conns[index].answer = NULL; } - if (g_conns[index].remote_desc) { - free(g_conns[index].remote_desc); - g_conns[index].remote_desc = NULL; + if (g_ad.conns[index].remote_desc) { + free(g_ad.conns[index].remote_desc); + g_ad.conns[index].remote_desc = NULL; } - g_list_free_full(g_conns[index].ice_candidates, free); - g_conns[index].ice_candidates = NULL; + g_list_free_full(g_ad.conns[index].ice_candidates, free); + g_ad.conns[index].ice_candidates = NULL; - g_conns[index].channel_index = 0; + g_ad.conns[index].channel_index = 0; for (i = 0; i < MAX_MEDIA_PACKET_SOURCE_LEN; i++) __release_packet_source(index, i); - if (g_conns[index].source.stream_info) { - sound_manager_destroy_stream_information(g_conns[index].source.stream_info); - g_conns[index].source.stream_info = NULL; + if (g_ad.conns[index].source.stream_info) { + sound_manager_destroy_stream_information(g_ad.conns[index].source.stream_info); + g_ad.conns[index].source.stream_info = NULL; } - if (g_conns[index].render.stream_info) { - sound_manager_destroy_stream_information(g_conns[index].render.stream_info); - g_conns[index].render.stream_info = NULL; + if (g_ad.conns[index].render.stream_info) { + sound_manager_destroy_stream_information(g_ad.conns[index].render.stream_info); + g_ad.conns[index].render.stream_info = NULL; } - g_conns[index].render.loopback_track_id = 0; + g_ad.conns[index].render.loopback_track_id = 0; #ifndef TIZEN_TV - if (g_conns[index].encoded_audio_frame_cb_is_set) - g_conns[index].encoded_audio_frame_cb_is_set = false; + if (g_ad.conns[index].encoded_audio_frame_cb_is_set) + g_ad.conns[index].encoded_audio_frame_cb_is_set = false; - if (g_conns[index].encoded_video_frame_cb_is_set) - g_conns[index].encoded_video_frame_cb_is_set = false; + if (g_ad.conns[index].encoded_video_frame_cb_is_set) + g_ad.conns[index].encoded_video_frame_cb_is_set = false; - if (g_conns[index].render.espp) { - g_conns[index].render.espp = NULL; + if (g_ad.conns[index].render.espp) { + g_ad.conns[index].render.espp = NULL; g_print("espp close & destroy\n"); } #endif @@ -455,23 +449,23 @@ static void _webrtc_start(int index) { int ret = WEBRTC_ERROR_NONE; - RET_IF(!g_conns[index].webrtc, "webrtc is NULL"); + RET_IF(!g_ad.conns[index].webrtc, "webrtc is NULL"); - ret = webrtc_start(g_conns[index].webrtc); + ret = webrtc_start(g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_start() success\n"); #ifndef TIZEN_TV - if (g_conns[index].encoded_audio_frame_cb_is_set || - g_conns[index].encoded_video_frame_cb_is_set) { - if (!g_conns[index].render.espp) + if (g_ad.conns[index].encoded_audio_frame_cb_is_set || + g_ad.conns[index].encoded_video_frame_cb_is_set) { + if (!g_ad.conns[index].render.espp) g_print("espp create & open\n"); } - if (g_conns[index].encoded_audio_frame_cb_is_set) + if (g_ad.conns[index].encoded_audio_frame_cb_is_set) g_print("espp set audio stream info\n"); - if (g_conns[index].encoded_video_frame_cb_is_set) + if (g_ad.conns[index].encoded_video_frame_cb_is_set) g_print("espp set video stream info\n"); #endif } @@ -481,40 +475,40 @@ static void _webrtc_stop(int index) int ret = WEBRTC_ERROR_NONE; int i; - RET_IF(!g_conns[index].webrtc, "webrtc is NULL"); + RET_IF(!g_ad.conns[index].webrtc, "webrtc is NULL"); - ret = webrtc_stop(g_conns[index].webrtc); + ret = webrtc_stop(g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_stop() success\n"); for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].recv_channels[i] != NULL) - g_conns[index].recv_channels[i] = NULL; + if (g_ad.conns[index].recv_channels[i] != NULL) + g_ad.conns[index].recv_channels[i] = NULL; } #ifdef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ - if (g_conns[index].video_render_pipeline) { - if (gst_element_set_state(g_conns[index].video_render_pipeline, GST_STATE_NULL) == GST_STATE_CHANGE_FAILURE) + if (g_ad.conns[index].video_render_pipeline) { + if (gst_element_set_state(g_ad.conns[index].video_render_pipeline, GST_STATE_NULL) == GST_STATE_CHANGE_FAILURE) g_printerr("failed to set state to NULL\n"); - gst_object_unref(g_conns[index].video_render_pipeline); - g_conns[index].video_render_pipeline = NULL; - g_conns[index].appsrc_for_video = NULL; + gst_object_unref(g_ad.conns[index].video_render_pipeline); + g_ad.conns[index].video_render_pipeline = NULL; + g_ad.conns[index].appsrc_for_video = NULL; g_print("video render pipeline is released\n"); } - if (g_conns[index].audio_render_pipeline) { - if (gst_element_set_state(g_conns[index].audio_render_pipeline, GST_STATE_NULL) == GST_STATE_CHANGE_FAILURE) + if (g_ad.conns[index].audio_render_pipeline) { + if (gst_element_set_state(g_ad.conns[index].audio_render_pipeline, GST_STATE_NULL) == GST_STATE_CHANGE_FAILURE) g_printerr("failed to set state to NULL\n"); - gst_object_unref(g_conns[index].audio_render_pipeline); - g_conns[index].audio_render_pipeline = NULL; - g_conns[index].appsrc_for_audio = NULL; + gst_object_unref(g_ad.conns[index].audio_render_pipeline); + g_ad.conns[index].audio_render_pipeline = NULL; + g_ad.conns[index].appsrc_for_audio = NULL; g_print("audio render pipeline is released\n"); } #endif #ifndef TIZEN_TV - if (g_conns[index].encoded_audio_frame_cb_is_set || - g_conns[index].encoded_video_frame_cb_is_set) { - if (g_conns[index].render.espp) { - g_conns[index].render.espp = NULL; + if (g_ad.conns[index].encoded_audio_frame_cb_is_set || + g_ad.conns[index].encoded_video_frame_cb_is_set) { + if (g_ad.conns[index].render.espp) { + g_ad.conns[index].render.espp = NULL; g_print("espp destroy\n"); } } @@ -526,7 +520,7 @@ static void _webrtc_get_state(int index) int ret = WEBRTC_ERROR_NONE; webrtc_state_e state; - ret = webrtc_get_state(g_conns[index].webrtc, &state); + ret = webrtc_get_state(g_ad.conns[index].webrtc, &state); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("state: [%s]\n", g_webrtc_state_str[state]); @@ -537,7 +531,7 @@ static int _get_empty_packet_sources_index(int index) int i; for (i = 0; i < MAX_MEDIA_PACKET_SOURCE_LEN; i++) { - if (g_conns[index].packet_sources[i].source_id == 0) + if (g_ad.conns[index].packet_sources[i].source_id == 0) return i; } @@ -553,7 +547,7 @@ static int _find_packet_sources_index(int index, unsigned int source_id) return -1; for (i = 0; i < MAX_MEDIA_PACKET_SOURCE_LEN; i++) { - if (g_conns[index].packet_sources[i].source_id == source_id) + if (g_ad.conns[index].packet_sources[i].source_id == source_id) return i; } @@ -700,17 +694,17 @@ static void _webrtc_add_media_source(int index, int type, unsigned int *source_i i = _get_empty_packet_sources_index(index); RET_IF(i == -1, "media packet source can be added up to %d", MAX_MEDIA_PACKET_SOURCE_LEN); } - ret = webrtc_add_media_source(g_conns[index].webrtc, type, &_source_id); + ret = webrtc_add_media_source(g_ad.conns[index].webrtc, type, &_source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); if (type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) { - g_conns[index].packet_sources[i].source_id = _source_id; - g_mutex_init(&g_conns[index].packet_sources[i].mutex); - g_cond_init(&g_conns[index].packet_sources[i].cond); + g_ad.conns[index].packet_sources[i].source_id = _source_id; + g_mutex_init(&g_ad.conns[index].packet_sources[i].mutex); + g_cond_init(&g_ad.conns[index].packet_sources[i].cond); } break; case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO ... WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_VIDEO: - ret = webrtc_add_media_source_internal(g_conns[index].webrtc, type, &_source_id); + ret = webrtc_add_media_source_internal(g_ad.conns[index].webrtc, type, &_source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); break; default: @@ -728,18 +722,18 @@ static void _webrtc_add_mic_source(int index, bool aec, unsigned int *source_id) int ret = WEBRTC_ERROR_NONE; unsigned int _source_id = 0; - ret = webrtc_add_media_source(g_conns[index].webrtc, WEBRTC_MEDIA_SOURCE_TYPE_MIC, &_source_id); + ret = webrtc_add_media_source(g_ad.conns[index].webrtc, WEBRTC_MEDIA_SOURCE_TYPE_MIC, &_source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - if (g_conns[index].source.stream_info) { - sound_manager_destroy_stream_information(g_conns[index].source.stream_info); - g_conns[index].source.stream_info = NULL; + if (g_ad.conns[index].source.stream_info) { + sound_manager_destroy_stream_information(g_ad.conns[index].source.stream_info); + g_ad.conns[index].source.stream_info = NULL; } - if (__get_mic_sound_stream_info(aec, &g_conns[index].source.stream_info) < 0) { + if (__get_mic_sound_stream_info(aec, &g_ad.conns[index].source.stream_info) < 0) { g_printerr("failed to __get_mic_sound_stream_info()\n"); } else { - ret = webrtc_mic_source_set_sound_stream_info(g_conns[index].webrtc, _source_id, g_conns[index].source.stream_info); + ret = webrtc_mic_source_set_sound_stream_info(g_ad.conns[index].webrtc, _source_id, g_ad.conns[index].source.stream_info); if (ret != WEBRTC_ERROR_NONE) g_printerr("failed to webrtc_mic_source_set_sound_stream_info(), ret[0x%x]\n", ret); } @@ -754,7 +748,7 @@ static void _webrtc_remove_media_source(int index, unsigned int source_id) int ret = WEBRTC_ERROR_NONE; int i; - ret = webrtc_remove_media_source(g_conns[index].webrtc, source_id); + ret = webrtc_remove_media_source(g_ad.conns[index].webrtc, source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_remove_media_source() success, source_id[%u]\n", source_id); @@ -767,7 +761,7 @@ static void _webrtc_media_source_set_pause(int index, unsigned int source_id, un { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_set_pause(g_conns[index].webrtc, source_id, media_type, pause); + ret = webrtc_media_source_set_pause(g_ad.conns[index].webrtc, source_id, media_type, pause); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_pause() success, source_id[%u], media_type[%s], pause[%d]\n", @@ -779,7 +773,7 @@ static void _webrtc_media_source_get_pause(int index, unsigned int source_id, un int ret = WEBRTC_ERROR_NONE; bool paused = false; - ret = webrtc_media_source_get_pause(g_conns[index].webrtc, source_id, media_type, &paused); + ret = webrtc_media_source_get_pause(g_ad.conns[index].webrtc, source_id, media_type, &paused); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_get_pause() success, source_id[%u], media_type[%s], paused[%d]\n", @@ -790,7 +784,7 @@ static void _webrtc_media_source_set_video_resolution(int index, unsigned int so { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_set_video_resolution(g_conns[index].webrtc, source_id, width, height); + ret = webrtc_media_source_set_video_resolution(g_ad.conns[index].webrtc, source_id, width, height); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_video_resolution() success, source_id[%u], [%dx%d]\n", source_id, width, height); @@ -802,7 +796,7 @@ static void _webrtc_media_source_get_video_resolution(int index, unsigned int so int width; int height; - ret = webrtc_media_source_get_video_resolution(g_conns[index].webrtc, source_id, &width, &height); + ret = webrtc_media_source_get_video_resolution(g_ad.conns[index].webrtc, source_id, &width, &height); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_get_video_resolution() success, source_id[%u], [%dx%d]\n", source_id, width, height); @@ -812,7 +806,7 @@ static void _webrtc_media_source_set_video_framerate(int index, unsigned int sou { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_set_video_framerate(g_conns[index].webrtc, source_id, framerate); + ret = webrtc_media_source_set_video_framerate(g_ad.conns[index].webrtc, source_id, framerate); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_video_framerate() success, source_id[%u], framerate[%d]\n", source_id, framerate); @@ -823,7 +817,7 @@ static void _webrtc_media_source_get_video_framerate(int index, unsigned int sou int ret = WEBRTC_ERROR_NONE; int framerate; - ret = webrtc_media_source_get_video_framerate(g_conns[index].webrtc, source_id, &framerate); + ret = webrtc_media_source_get_video_framerate(g_ad.conns[index].webrtc, source_id, &framerate); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_get_video_framerate() success, source_id[%u], framerate[%d]\n", source_id, framerate); @@ -913,7 +907,7 @@ static void _webrtc_media_packet_source_set_format(int index, unsigned int sourc if (__create_formats(type, &format) == -1) return; - ret = webrtc_media_packet_source_set_format(g_conns[index].webrtc, source_id, format); + ret = webrtc_media_packet_source_set_format(g_ad.conns[index].webrtc, source_id, format); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_media_format() success, source_id[%u], format[%p]\n", source_id, format); @@ -921,10 +915,10 @@ static void _webrtc_media_packet_source_set_format(int index, unsigned int sourc i = _find_packet_sources_index(index, source_id); RET_IF(i == -1, "could not find media packet source id[%u]", source_id); - if (g_conns[index].packet_sources[i].format) - media_format_unref(g_conns[index].packet_sources[i].format); - g_conns[index].packet_sources[i].format = format; - g_conns[index].packet_sources[i].webrtc = g_conns[index].webrtc; + if (g_ad.conns[index].packet_sources[i].format) + media_format_unref(g_ad.conns[index].packet_sources[i].format); + g_ad.conns[index].packet_sources[i].format = format; + g_ad.conns[index].packet_sources[i].webrtc = g_ad.conns[index].webrtc; } static void _webrtc_media_source_get_transceiver_direction(int index, unsigned int source_id, webrtc_media_type_e media_type) @@ -932,7 +926,7 @@ static void _webrtc_media_source_get_transceiver_direction(int index, unsigned i int ret = WEBRTC_ERROR_NONE; webrtc_transceiver_direction_e direction; - ret = webrtc_media_source_get_transceiver_direction(g_conns[index].webrtc, source_id, media_type, &direction); + ret = webrtc_media_source_get_transceiver_direction(g_ad.conns[index].webrtc, source_id, media_type, &direction); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_get_transceiver_direction() success, source_id[%u], media_type[%s], direction[%s]\n", @@ -943,7 +937,7 @@ static void _webrtc_media_source_set_transceiver_direction(int index, unsigned i { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_set_transceiver_direction(g_conns[index].webrtc, source_id, media_type, direction); + ret = webrtc_media_source_set_transceiver_direction(g_ad.conns[index].webrtc, source_id, media_type, direction); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_transceiver_direction() success, source_id[%u], media_type[%s], direction[%s]\n", @@ -952,7 +946,7 @@ static void _webrtc_media_source_set_transceiver_direction(int index, unsigned i static void _webrtc_set_display_type(int index, int type) { - g_conns[index].render.display_type = type; + g_ad.conns[index].render.display_type = type; g_print("display type[%d] is set, it'll be applied when starting rendering video.\n", type); } @@ -961,7 +955,7 @@ static void _webrtc_set_display_mode(int index, unsigned int track_id, unsigned { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_set_display_mode(g_conns[index].webrtc, track_id, mode); + ret = webrtc_set_display_mode(g_ad.conns[index].webrtc, track_id, mode); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_display_mode() success, track_id[%u], mode[%u]\n", track_id, mode); @@ -972,7 +966,7 @@ static void _webrtc_get_display_mode(int index, unsigned int track_id) int ret = WEBRTC_ERROR_NONE; webrtc_display_mode_e mode; - ret = webrtc_get_display_mode(g_conns[index].webrtc, track_id, &mode); + ret = webrtc_get_display_mode(g_ad.conns[index].webrtc, track_id, &mode); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_get_display_mode() success, track_id[%u], mode[%u]\n", track_id, mode); @@ -982,7 +976,7 @@ static void _webrtc_set_display_visible(int index, unsigned int track_id, int vi { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_set_display_visible(g_conns[index].webrtc, track_id, (bool)visible); + ret = webrtc_set_display_visible(g_ad.conns[index].webrtc, track_id, (bool)visible); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_display_visible() success, track_id[%u], visible[%u]\n", track_id, visible); @@ -993,7 +987,7 @@ static void _webrtc_get_display_visible(int index, unsigned int track_id) int ret = WEBRTC_ERROR_NONE; bool visible; - ret = webrtc_get_display_visible(g_conns[index].webrtc, track_id, &visible); + ret = webrtc_get_display_visible(g_ad.conns[index].webrtc, track_id, &visible); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_get_display_visible() success, track_id[%u], visible[%u]\n", track_id, visible); @@ -1004,12 +998,12 @@ static void _webrtc_media_source_set_audio_loopback(int index, unsigned int sour int ret = WEBRTC_ERROR_NONE; unsigned int track_id; - if (!g_conns[index].render.stream_info) { - ret = sound_manager_create_stream_information(SOUND_STREAM_TYPE_MEDIA, NULL, NULL, &g_conns[index].render.stream_info); + if (!g_ad.conns[index].render.stream_info) { + ret = sound_manager_create_stream_information(SOUND_STREAM_TYPE_MEDIA, NULL, NULL, &g_ad.conns[index].render.stream_info); RET_IF(ret != SOUND_MANAGER_ERROR_NONE, "failed to sound_manager_create_stream_information(), ret[0x%x]", ret); } - ret = webrtc_media_source_set_audio_loopback(g_conns[index].webrtc, source_id, g_conns[index].render.stream_info, &track_id); + ret = webrtc_media_source_set_audio_loopback(g_ad.conns[index].webrtc, source_id, g_ad.conns[index].render.stream_info, &track_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_audio_loopback() success, source_id[%u] track_id[%u]\n", source_id, track_id); @@ -1019,7 +1013,7 @@ static void _webrtc_media_source_unset_audio_loopback(int index, unsigned int so { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_unset_audio_loopback(g_conns[index].webrtc, source_id); + ret = webrtc_media_source_unset_audio_loopback(g_ad.conns[index].webrtc, source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_unset_audio_loopback() success, source_id[%u]\n", source_id); @@ -1030,21 +1024,21 @@ static void _webrtc_media_source_set_video_loopback(int index, unsigned int sour int ret = WEBRTC_ERROR_NONE; unsigned int track_id; - ret = webrtc_media_source_set_video_loopback(g_conns[index].webrtc, source_id, WEBRTC_DISPLAY_TYPE_EVAS, g_eo_mine, &track_id); + ret = webrtc_media_source_set_video_loopback(g_ad.conns[index].webrtc, source_id, WEBRTC_DISPLAY_TYPE_EVAS, g_ad.eo_mine, &track_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_video_loopback() success, source_id[%u] track_id[%u]\n", source_id, track_id); - g_conns[index].render.loopback_track_id = track_id; + g_ad.conns[index].render.loopback_track_id = track_id; } static void _webrtc_media_source_unset_video_loopback(int index, unsigned int source_id) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_source_unset_video_loopback(g_conns[index].webrtc, source_id); + ret = webrtc_media_source_unset_video_loopback(g_ad.conns[index].webrtc, source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - evas_object_hide(g_eo_mine); + evas_object_hide(g_ad.eo_mine); g_print("webrtc_media_source_unset_video_loopback() success, source_id[%u]\n", source_id); } @@ -1073,31 +1067,31 @@ static void _webrtc_data_channel_send_string(const char *string, bool send_as_by int ret; int i, j; gchar *string_with_peer_id; - string_with_peer_id = g_strdup_printf("[%d] %s", g_signaling_server.local_peer_id, string); + string_with_peer_id = g_strdup_printf("[%d] %s", g_ad.signaling_server.local_peer_id, string); for (i = 0; i < MAX_CONNECTION_LEN; i++) { - if (!g_conns[i].webrtc) + if (!g_ad.conns[i].webrtc) continue; for (j = 0; j < MAX_CHANNEL_LEN; j++) { - if (!g_conns[i].channels[j]) + if (!g_ad.conns[i].channels[j]) continue; - g_print("[%d] data_channel[%p], index[%d]: ", i, g_conns[i].channels[j], j); + g_print("[%d] data_channel[%p], index[%d]: ", i, g_ad.conns[i].channels[j], j); if (send_as_bytes) { - ret = webrtc_data_channel_send_bytes(g_conns[i].channels[j], string_with_peer_id, strlen(string_with_peer_id)); + ret = webrtc_data_channel_send_bytes(g_ad.conns[i].channels[j], string_with_peer_id, strlen(string_with_peer_id)); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_send_bytes(), string[%s]\n", string_with_peer_id); else g_print("webrtc_data_channel_send_bytes() success, string[%s]\n", string_with_peer_id); } else { - ret = webrtc_data_channel_send_string(g_conns[i].channels[j], string_with_peer_id); + ret = webrtc_data_channel_send_string(g_ad.conns[i].channels[j], string_with_peer_id); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_send_string(), string[%s]\n", string_with_peer_id); else g_print("webrtc_data_channel_send_string() success, string[%s]\n", string_with_peer_id); } } - if (g_conns[i].render.loopback_track_id > 0) - __render_text_message(&g_text_eo_mine, 0, string_with_peer_id); + if (g_ad.conns[i].render.loopback_track_id > 0) + __render_text_message(&g_ad.text_eo_mine, 0, string_with_peer_id); } g_free(string_with_peer_id); @@ -1118,9 +1112,9 @@ static void _webrtc_data_channel_send_file(int index, const char *file_path) unsigned int buffered_amount; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - g_print("data_channel[%p], index[%d]: ", g_conns[index].channels[i], i); + g_print("data_channel[%p], index[%d]: ", g_ad.conns[index].channels[i], i); fd = open(file_path, O_RDONLY); if (fd == -1) { @@ -1138,13 +1132,13 @@ static void _webrtc_data_channel_send_file(int index, const char *file_path) expected_size = g_strdup_printf("expected size:%llu", (unsigned long long)st.st_size); expected_name = g_strdup_printf("expected name:%s", basename((char *)file_path)); - ret = webrtc_data_channel_send_string(g_conns[index].channels[i], expected_size); + ret = webrtc_data_channel_send_string(g_ad.conns[index].channels[i], expected_size); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_send_string(), string[%s]\n", expected_size); else g_print("webrtc_data_channel_send_string() success, string[%s]\n", expected_size); - ret = webrtc_data_channel_send_string(g_conns[index].channels[i], expected_name); + ret = webrtc_data_channel_send_string(g_ad.conns[index].channels[i], expected_name); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_send_string(), string[%s]\n", expected_name); else @@ -1158,11 +1152,11 @@ static void _webrtc_data_channel_send_file(int index, const char *file_path) sum_size += read_size; g_print("%s : [%llu / %llu]bytes is read.", file_path, sum_size, (unsigned long long)st.st_size); - ret = webrtc_data_channel_send_bytes(g_conns[index].channels[i], buffer, (unsigned int)read_size); + ret = webrtc_data_channel_send_bytes(g_ad.conns[index].channels[i], buffer, (unsigned int)read_size); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_send_bytes(), file_path[%s], size[%zd]\n", file_path, read_size); - ret = webrtc_data_channel_get_buffered_amount(g_conns[index].channels[i], &buffered_amount); + ret = webrtc_data_channel_get_buffered_amount(g_ad.conns[index].channels[i], &buffered_amount); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_get_buffered_amount()\n"); if (buffered_amount > 0) @@ -1176,7 +1170,7 @@ static void _webrtc_data_channel_send_file(int index, const char *file_path) static void _webrtc_media_source_set_mute(int index, unsigned int source_id, webrtc_media_type_e media_type, bool mute) { - int ret = webrtc_media_source_set_mute(g_conns[index].webrtc, source_id, media_type, mute); + int ret = webrtc_media_source_set_mute(g_ad.conns[index].webrtc, source_id, media_type, mute); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_set_mute() success, source_id[%d], media_type[%s], mute[%d]\n", @@ -1186,7 +1180,7 @@ static void _webrtc_media_source_set_mute(int index, unsigned int source_id, web static void _webrtc_media_source_get_mute(int index, unsigned int source_id, webrtc_media_type_e media_type) { bool muted = false; - int ret = webrtc_media_source_get_mute(g_conns[index].webrtc, source_id, media_type, &muted); + int ret = webrtc_media_source_get_mute(g_ad.conns[index].webrtc, source_id, media_type, &muted); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_source_get_mute() success, source_id[%d], media_type[%s], muted[%d]\n", @@ -1197,7 +1191,7 @@ static void _webrtc_screen_source_set_crop(int index, unsigned int source_id, in { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_screen_source_set_crop(g_conns[index].webrtc, source_id, x, y, w, h, is_vertical, width, height); + ret = webrtc_screen_source_set_crop(g_ad.conns[index].webrtc, source_id, x, y, w, h, is_vertical, width, height); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_screen_source_set_crop(), source_id[%d], ret[0x%x]\n", source_id, ret); @@ -1208,7 +1202,7 @@ static void _webrtc_screen_source_unset_crop(int index, unsigned int source_id) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_screen_source_unset_crop(g_conns[index].webrtc, source_id); + ret = webrtc_screen_source_unset_crop(g_ad.conns[index].webrtc, source_id); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_screen_source_unset_crop(), source_id[%d], ret[0x%x]\n", source_id, ret); } @@ -1336,7 +1330,7 @@ static void _webrtc_set_stun_server(int index, char *uri) if (!uri) return; - ret = webrtc_set_stun_server(g_conns[index].webrtc, uri); + ret = webrtc_set_stun_server(g_ad.conns[index].webrtc, uri); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_stun_server() success, uri[%s]\n", uri); @@ -1347,7 +1341,7 @@ static void _webrtc_get_stun_server(int index) int ret = WEBRTC_ERROR_NONE; char *stun_server; - ret = webrtc_get_stun_server(g_conns[index].webrtc, &stun_server); + ret = webrtc_get_stun_server(g_ad.conns[index].webrtc, &stun_server); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("STUN server[%s]", stun_server); @@ -1361,7 +1355,7 @@ static void _webrtc_add_turn_server(int index, char *uri) if (!uri) return; - ret = webrtc_add_turn_server(g_conns[index].webrtc, uri); + ret = webrtc_add_turn_server(g_ad.conns[index].webrtc, uri); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_add_turn_server() success, uri[%s]\n", uri); @@ -1377,7 +1371,7 @@ static void _webrtc_get_turn_servers(int index) { int ret = 0; - ret = webrtc_foreach_turn_server(g_conns[index].webrtc, foreach_turn_server, NULL); + ret = webrtc_foreach_turn_server(g_ad.conns[index].webrtc, foreach_turn_server, NULL); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_foreach_turn_server() success\n"); @@ -1387,7 +1381,7 @@ static void _webrtc_set_bundle_policy(int index, int policy) { int ret = 0; - ret = webrtc_set_bundle_policy(g_conns[index].webrtc, policy); + ret = webrtc_set_bundle_policy(g_ad.conns[index].webrtc, policy); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_bundle_policy() success\n"); @@ -1398,7 +1392,7 @@ static void _webrtc_get_bundle_policy(int index) int ret = 0; webrtc_bundle_policy_e policy; - ret = webrtc_get_bundle_policy(g_conns[index].webrtc, &policy); + ret = webrtc_get_bundle_policy(g_ad.conns[index].webrtc, &policy); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_get_bundle_policy() success, policy[%d]\n", policy); @@ -1408,7 +1402,7 @@ static void _webrtc_set_ice_transport_policy(int index, int policy) { int ret = 0; - ret = webrtc_set_ice_transport_policy(g_conns[index].webrtc, policy); + ret = webrtc_set_ice_transport_policy(g_ad.conns[index].webrtc, policy); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_ice_transport_policy() success\n"); @@ -1419,7 +1413,7 @@ static void _webrtc_get_ice_transport_policy(int index) int ret = 0; webrtc_ice_transport_policy_e policy; - ret = webrtc_get_ice_transport_policy(g_conns[index].webrtc, &policy); + ret = webrtc_get_ice_transport_policy(g_ad.conns[index].webrtc, &policy); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_get_ice_transport_policy() success, policy[%d]\n", policy); @@ -1429,7 +1423,7 @@ static void _webrtc_set_rtp_packet_drop_probability(bool sender, float probabili { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_set_rtp_packet_drop_probability(g_conns[0].webrtc, sender, probability); + ret = webrtc_set_rtp_packet_drop_probability(g_ad.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); @@ -1440,7 +1434,7 @@ 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 = webrtc_get_rtp_packet_drop_probability(g_ad.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); @@ -1667,10 +1661,10 @@ static void _webrtc_data_channel_set_buffered_amount_low_cb(int index, unsigned int i; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - ret = webrtc_data_channel_set_buffered_amount_low_cb(g_conns[index].channels[i], threshold, __data_channel_buffered_amount_low_cb, &g_conns[index]); + ret = webrtc_data_channel_set_buffered_amount_low_cb(g_ad.conns[index].channels[i], threshold, __data_channel_buffered_amount_low_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_set_buffered_amount_low_cb(), index[%d]\n", i); else @@ -1685,10 +1679,10 @@ static void _webrtc_data_channel_get_buffered_amount_low_threshold(int index) unsigned int threshold; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - ret = webrtc_data_channel_get_buffered_amount_low_threshold(g_conns[index].channels[i], &threshold); + ret = webrtc_data_channel_get_buffered_amount_low_threshold(g_ad.conns[index].channels[i], &threshold); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_get_buffered_amount_low_threshold(), index[%d]\n", i); else @@ -1702,10 +1696,10 @@ static void _webrtc_data_channel_unset_buffered_amount_low_cb(int index) int i; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - ret = webrtc_data_channel_unset_buffered_amount_low_cb(g_conns[index].channels[i]); + ret = webrtc_data_channel_unset_buffered_amount_low_cb(g_ad.conns[index].channels[i]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_unset_buffered_amount_low_cb(), index[%d]\n", i); else @@ -1720,10 +1714,10 @@ static void _webrtc_data_channel_get_buffered_amount(int index) unsigned int buffered_amount; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - ret = webrtc_data_channel_get_buffered_amount(g_conns[index].channels[i], &buffered_amount); + ret = webrtc_data_channel_get_buffered_amount(g_ad.conns[index].channels[i], &buffered_amount); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_data_channel_get_buffered_amount(), index[%d]\n", i); else @@ -1805,7 +1799,7 @@ static void _webrtc_foreach_stats(int index, int selected) return; } - ret = webrtc_foreach_stats(g_conns[index].webrtc, type_mask, __stats_cb, NULL); + ret = webrtc_foreach_stats(g_ad.conns[index].webrtc, type_mask, __stats_cb, NULL); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_foreach_stats(), index[%d]\n", index); else @@ -1827,7 +1821,7 @@ static void _websocket_connection_send_text_for_room(connection_s *conn, int rem RET_IF(!message, "message is NULL"); message_for_room = g_strdup_printf ("ROOM_PEER_MSG %d %s", remote_peer_id, message); - soup_websocket_connection_send_text(g_signaling_server.ws_conn, message_for_room); + soup_websocket_connection_send_text(g_ad.signaling_server.ws_conn, message_for_room); g_free(message_for_room); } @@ -1870,22 +1864,22 @@ static void __signaling_message_cb(webrtc_signaling_message_type_e type, const c if (type == SIGNALING_MESSAGE_TYPE_CONNECTED) { g_print("\n[from SERVER > CONNECTED %s]\n", message); - g_signaling_server.is_connected = true; - g_signaling_server.server_status = SERVER_STATUS_CONNECTED; - g_signaling_server.local_peer_id = atoi(message); + g_ad.signaling_server.is_connected = true; + g_ad.signaling_server.server_status = SERVER_STATUS_CONNECTED; + g_ad.signaling_server.local_peer_id = atoi(message); } else if (type == SIGNALING_MESSAGE_TYPE_DISCONNECTED) { g_print("\n[from SERVER > DISCONNECTED %s]\n", message); - g_signaling_server.is_connected = false; - g_signaling_server.server_status = SERVER_STATUS_DISCONNECTED; + g_ad.signaling_server.is_connected = false; + g_ad.signaling_server.server_status = SERVER_STATUS_DISCONNECTED; } else if (type == SIGNALING_MESSAGE_TYPE_SESSION_ESTABLISHED) { g_print("\n[from SERVER > SESSION_ESTABLISHED with %s]\n", message); - g_signaling_server.server_status = SERVER_STATUS_SESSION_ESTABLISHED; + g_ad.signaling_server.server_status = SERVER_STATUS_SESSION_ESTABLISHED; } else if (type == SIGNALING_MESSAGE_TYPE_SESSION_CLOSED) { g_print("\n[from SERVER > SESSION_CLOSED with %s]\n", message); - g_signaling_server.server_status = SERVER_STATUS_SESSION_CLOSED; + g_ad.signaling_server.server_status = SERVER_STATUS_SESSION_CLOSED; } else if (type == SIGNALING_MESSAGE_TYPE_SDP) { g_print("\n[from SERVER > SDP]\n%s\n", message); @@ -1903,18 +1897,18 @@ static void _webrtc_signaling_connect(const char *ip, int port) { int ret = WEBRTC_ERROR_NONE; - if (strlen(g_signaling_server.url) > 0) { - g_printerr("server[%s] is already set by 'ss'\n", g_signaling_server.url); + if (strlen(g_ad.signaling_server.url) > 0) { + g_printerr("server[%s] is already set by 'ss'\n", g_ad.signaling_server.url); return; } - ret = webrtc_signaling_connect(ip, port, __signaling_message_cb, &g_conns[0], &g_signaling_server.signaling_client); + ret = webrtc_signaling_connect(ip, port, __signaling_message_cb, &g_ad.conns[0], &g_ad.signaling_server.signaling_client); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - if (g_signaling_server.private_ip) - free(g_signaling_server.private_ip); - g_signaling_server.private_ip = strdup(ip); - g_signaling_server.port = port; + if (g_ad.signaling_server.private_ip) + free(g_ad.signaling_server.private_ip); + g_ad.signaling_server.private_ip = strdup(ip); + g_ad.signaling_server.port = port; g_print("webrtc_signaling_connect() success\n"); } @@ -1923,7 +1917,7 @@ static void _webrtc_signaling_disconnect(void) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_disconnect(g_signaling_server.signaling_client); + ret = webrtc_signaling_disconnect(g_ad.signaling_server.signaling_client); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_disconnect() success\n"); @@ -1933,7 +1927,7 @@ static void _webrtc_signaling_request_session(int peer_id) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_request_session(g_signaling_server.signaling_client, peer_id); + ret = webrtc_signaling_request_session(g_ad.signaling_server.signaling_client, peer_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_request_session() success\n"); @@ -1944,9 +1938,9 @@ static void _webrtc_signaling_send_message(const char *message) int ret = WEBRTC_ERROR_NONE; RET_IF(!message, "message is NULL"); - RET_IF(!g_signaling_server.signaling_client, "signaling_client is NULL"); + RET_IF(!g_ad.signaling_server.signaling_client, "signaling_client is NULL"); - ret = webrtc_signaling_send_message(g_signaling_server.signaling_client, message); + ret = webrtc_signaling_send_message(g_ad.signaling_server.signaling_client, message); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_send_message() success\n"); @@ -1960,12 +1954,12 @@ static void __ice_candidate_cb(webrtc_h webrtc, const char *candidate, void *use RET_IF(!conn, "conn is NULL"); g_print("\n[to SERVER > ICE]\n%s\n", candidate); - if (g_signaling_server.ws_conn) { + if (g_ad.signaling_server.ws_conn) { if (conn->is_for_room) _websocket_connection_send_text_for_room(conn, conn->remote_peer_id, candidate); else - soup_websocket_connection_send_text(g_signaling_server.ws_conn, candidate); - } else if (g_signaling_server.signaling_client && g_signaling_server.is_connected) { + soup_websocket_connection_send_text(g_ad.signaling_server.ws_conn, candidate); + } else if (g_ad.signaling_server.signaling_client && g_ad.signaling_server.is_connected) { _webrtc_signaling_send_message(candidate); } } @@ -2015,25 +2009,25 @@ static void __ice_connection_state_change_cb(webrtc_h webrtc, webrtc_ice_connect static void _webrtc_set_all_negotiation_state_change_cbs(int index) { - int ret = webrtc_set_peer_connection_state_change_cb(g_conns[index].webrtc, __peer_connection_state_change_cb, &g_conns[index]); + int ret = webrtc_set_peer_connection_state_change_cb(g_ad.conns[index].webrtc, __peer_connection_state_change_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_peer_connection_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_peer_connection_state_change_cb() success\n"); - ret = webrtc_set_signaling_state_change_cb(g_conns[index].webrtc, __signaling_state_change_cb, &g_conns[index]); + ret = webrtc_set_signaling_state_change_cb(g_ad.conns[index].webrtc, __signaling_state_change_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_signaling_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_signaling_state_change_cb() success\n"); - ret = webrtc_set_ice_gathering_state_change_cb(g_conns[index].webrtc, __ice_gathering_state_change_cb, &g_conns[index]); + ret = webrtc_set_ice_gathering_state_change_cb(g_ad.conns[index].webrtc, __ice_gathering_state_change_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_ice_gathering_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_ice_gathering_state_change_cb() success\n"); - ret = webrtc_set_ice_connection_state_change_cb(g_conns[index].webrtc, __ice_connection_state_change_cb, &g_conns[index]); + ret = webrtc_set_ice_connection_state_change_cb(g_ad.conns[index].webrtc, __ice_connection_state_change_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_ice_connection_state_change_cb(), ret[0x%x]\n", ret); else @@ -2042,25 +2036,25 @@ static void _webrtc_set_all_negotiation_state_change_cbs(int index) static void _webrtc_unset_all_negotiation_state_change_cbs(int index) { - int ret = webrtc_unset_peer_connection_state_change_cb(g_conns[index].webrtc); + int ret = webrtc_unset_peer_connection_state_change_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_peer_connection_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_peer_connection_state_change_cb() success\n"); - ret = webrtc_unset_signaling_state_change_cb(g_conns[index].webrtc); + ret = webrtc_unset_signaling_state_change_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_signaling_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_signaling_state_change_cb() success\n"); - ret = webrtc_unset_ice_gathering_state_change_cb(g_conns[index].webrtc); + ret = webrtc_unset_ice_gathering_state_change_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_ice_gathering_state_change_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_ice_gathering_state_change_cb() success\n"); - ret = webrtc_unset_ice_connection_state_change_cb(g_conns[index].webrtc); + ret = webrtc_unset_ice_connection_state_change_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_ice_connection_state_change_cb(), ret[0x%x]\n", ret); else @@ -2075,19 +2069,19 @@ static void _webrtc_get_all_negotiation_states(int index) webrtc_ice_gathering_state_e ice_gathering_state = WEBRTC_ICE_GATHERING_STATE_NEW; webrtc_ice_connection_state_e ice_conn_state = WEBRTC_ICE_CONNECTION_STATE_NEW; - ret = webrtc_get_peer_connection_state(g_conns[index].webrtc, &peer_conn_state); + ret = webrtc_get_peer_connection_state(g_ad.conns[index].webrtc, &peer_conn_state); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_get_peer_connection_state()\n"); - ret = webrtc_get_signaling_state(g_conns[index].webrtc, &signaling_state); + ret = webrtc_get_signaling_state(g_ad.conns[index].webrtc, &signaling_state); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_get_signaling_state()\n"); - ret = webrtc_get_ice_gathering_state(g_conns[index].webrtc, &ice_gathering_state); + ret = webrtc_get_ice_gathering_state(g_ad.conns[index].webrtc, &ice_gathering_state); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_get_ice_gathering_state()\n"); - ret = webrtc_get_ice_connection_state(g_conns[index].webrtc, &ice_conn_state); + ret = webrtc_get_ice_connection_state(g_ad.conns[index].webrtc, &ice_conn_state); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_get_ice_connection_state()\n"); @@ -2122,8 +2116,8 @@ static void __track_added_cb(webrtc_h webrtc, webrtc_media_type_e type, unsigned if (type == WEBRTC_MEDIA_TYPE_VIDEO) { #ifndef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ if (conn->render.display_type == WEBRTC_DISPLAY_TYPE_OVERLAY) { - g_print("Video track is added, set display - overlay, object[%p]\n", g_win_id); - webrtc_set_display(webrtc, id, WEBRTC_DISPLAY_TYPE_OVERLAY, g_win_id); + g_print("Video track is added, set display - overlay, object[%p]\n", g_ad.win_id); + webrtc_set_display(webrtc, id, WEBRTC_DISPLAY_TYPE_OVERLAY, g_ad.win_id); } else if (conn->render.display_type == WEBRTC_DISPLAY_TYPE_EVAS) { g_print("Video track is added, set display - evas object[%p]\n", conn->render.eo); @@ -2166,39 +2160,39 @@ static void __track_added_cb(webrtc_h webrtc, webrtc_media_type_e type, unsigned static void _webrtc_set_all_basic_cbs(int index) { /* COMMON */ - int ret = webrtc_set_error_cb(g_conns[index].webrtc, __error_cb, g_conns[index].webrtc); + int ret = webrtc_set_error_cb(g_ad.conns[index].webrtc, __error_cb, g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_error_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_error_cb() success\n"); - ret = webrtc_set_state_changed_cb(g_conns[index].webrtc, __state_changed_cb, &g_conns[index]); + ret = webrtc_set_state_changed_cb(g_ad.conns[index].webrtc, __state_changed_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_state_changed_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_state_changed_cb() success\n"); - ret = webrtc_set_negotiation_needed_cb(g_conns[index].webrtc, __negotiation_needed_cb, g_conns[index].webrtc); + ret = webrtc_set_negotiation_needed_cb(g_ad.conns[index].webrtc, __negotiation_needed_cb, g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_negotiation_needed_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_negotiation_needed_cb() success\n"); - ret = webrtc_set_ice_candidate_cb(g_conns[index].webrtc, __ice_candidate_cb, &g_conns[index]); + ret = webrtc_set_ice_candidate_cb(g_ad.conns[index].webrtc, __ice_candidate_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_ice_candidate_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_ice_candidate_cb() success\n"); /* MEDIA RENDER */ - ret = webrtc_set_track_added_cb(g_conns[index].webrtc, __track_added_cb, &g_conns[index]); + ret = webrtc_set_track_added_cb(g_ad.conns[index].webrtc, __track_added_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_track_added_cb(), ret[0x%x]\n", ret); else g_print("webrtc_set_track_added_cb() success\n"); /* DATA CHANNEL */ - ret = webrtc_set_data_channel_cb(g_conns[index].webrtc, __data_channel_cb, &g_conns[index]); + ret = webrtc_set_data_channel_cb(g_ad.conns[index].webrtc, __data_channel_cb, &g_ad.conns[index]); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_set_data_channel_cb(), ret[0x%x]\n", ret); else @@ -2208,39 +2202,39 @@ static void _webrtc_set_all_basic_cbs(int index) static void _webrtc_unset_all_basic_cbs(int index) { /* COMMON */ - int ret = webrtc_unset_error_cb(g_conns[index].webrtc); + int ret = webrtc_unset_error_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_error_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_error_cb() success\n"); - ret = webrtc_unset_state_changed_cb(g_conns[index].webrtc); + ret = webrtc_unset_state_changed_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_state_changed_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_state_changed_cb() success\n"); - ret = webrtc_unset_negotiation_needed_cb(g_conns[index].webrtc); + ret = webrtc_unset_negotiation_needed_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_negotiation_needed_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_negotiation_needed_cb() success\n"); - ret = webrtc_unset_ice_candidate_cb(g_conns[index].webrtc); + ret = webrtc_unset_ice_candidate_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_ice_candidate_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_ice_candidate_cb() success\n"); /* MEDIA RENDER */ - ret = webrtc_unset_track_added_cb(g_conns[index].webrtc); + ret = webrtc_unset_track_added_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_track_added_cb(), ret[0x%x]\n", ret); else g_print("webrtc_unset_track_added_cb() success\n"); /* DATA CHANNEL */ - ret = webrtc_unset_data_channel_cb(g_conns[index].webrtc); + ret = webrtc_unset_data_channel_cb(g_ad.conns[index].webrtc); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_unset_data_channel_cb(), ret[0x%x]\n", ret); else @@ -2349,15 +2343,15 @@ static void _webrtc_set_encoded_audio_frame_cb(int index) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_set_encoded_audio_frame_cb(g_conns[index].webrtc, __encoded_frame_cb, &g_conns[index]); + ret = webrtc_set_encoded_audio_frame_cb(g_ad.conns[index].webrtc, __encoded_frame_cb, &g_ad.conns[index]); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_encoded_audio_frame_cb() success\n"); #ifndef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ #ifndef TIZEN_TV - if (!g_conns[index].encoded_audio_frame_cb_is_set) { - g_conns[index].encoded_audio_frame_cb_is_set = true; - if (!g_conns[index].render.espp) + if (!g_ad.conns[index].encoded_audio_frame_cb_is_set) { + g_ad.conns[index].encoded_audio_frame_cb_is_set = true; + if (!g_ad.conns[index].render.espp) g_print("espp create & open\n"); g_print("espp set audio stream info\n"); @@ -2370,14 +2364,14 @@ static void _webrtc_unset_encoded_audio_frame_cb(int index) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_unset_encoded_audio_frame_cb(g_conns[index].webrtc); + ret = webrtc_unset_encoded_audio_frame_cb(g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_unset_encoded_audio_frame_cb() success\n"); #ifndef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ #ifndef TIZEN_TV - if (g_conns[index].encoded_audio_frame_cb_is_set) { - g_conns[index].encoded_audio_frame_cb_is_set = false; + if (g_ad.conns[index].encoded_audio_frame_cb_is_set) { + g_ad.conns[index].encoded_audio_frame_cb_is_set = false; g_print("espp close & destroy\n"); } #endif @@ -2388,16 +2382,16 @@ static void _webrtc_set_encoded_video_frame_cb(int index) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_set_encoded_video_frame_cb(g_conns[index].webrtc, __encoded_frame_cb, &g_conns[index]); + ret = webrtc_set_encoded_video_frame_cb(g_ad.conns[index].webrtc, __encoded_frame_cb, &g_ad.conns[index]); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_set_encoded_video_frame_cb() success\n"); #ifndef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ #ifndef TIZEN_TV - if (!g_conns[index].encoded_video_frame_cb_is_set) { - g_conns[index].encoded_video_frame_cb_is_set = true; - if (!g_conns[index].render.espp) { + if (!g_ad.conns[index].encoded_video_frame_cb_is_set) { + g_ad.conns[index].encoded_video_frame_cb_is_set = true; + if (!g_ad.conns[index].render.espp) { g_print("espp create & open\n"); } g_print("espp set video stream info\n"); @@ -2410,15 +2404,15 @@ static void _webrtc_unset_encoded_video_frame_cb(int index) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_unset_encoded_video_frame_cb(g_conns[index].webrtc); + ret = webrtc_unset_encoded_video_frame_cb(g_ad.conns[index].webrtc); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_unset_encoded_video_frame_cb() success\n"); #ifndef __DEBUG_VALIDATE_ENCODED_FRAME_CB__ #ifndef TIZEN_TV - if (g_conns[index].encoded_video_frame_cb_is_set) { - g_conns[index].encoded_video_frame_cb_is_set = false; + if (g_ad.conns[index].encoded_video_frame_cb_is_set) { + g_ad.conns[index].encoded_video_frame_cb_is_set = false; g_print("espp close & destroy\n"); } #endif @@ -2452,7 +2446,7 @@ static void _webrtc_media_packet_source_set_buffer_state_changed_cb(int index, u { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_packet_source_set_buffer_state_changed_cb(g_conns[index].webrtc, source_id, __media_packet_source_buffer_state_changed_cb, &g_conns[index]); + ret = webrtc_media_packet_source_set_buffer_state_changed_cb(g_ad.conns[index].webrtc, source_id, __media_packet_source_buffer_state_changed_cb, &g_ad.conns[index]); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_packet_source_set_buffer_state_changed_cb() success\n"); @@ -2462,7 +2456,7 @@ static void _webrtc_media_packet_source_unset_buffer_state_changed_cb(int index, { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_media_packet_source_unset_buffer_state_changed_cb(g_conns[index].webrtc, source_id); + ret = webrtc_media_packet_source_unset_buffer_state_changed_cb(g_ad.conns[index].webrtc, source_id); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_media_packet_source_unset_buffer_state_changed_cb() success\n"); @@ -2476,8 +2470,8 @@ static void _webrtc_set_all_cbs(int index) _webrtc_set_all_negotiation_state_change_cbs(index); for (i = 0; i < MAX_MEDIA_PACKET_SOURCE_LEN; i++) { - if (g_conns[index].packet_sources[i].source_id > 0) - _webrtc_media_packet_source_set_buffer_state_changed_cb(0, g_conns[index].packet_sources[i].source_id); + if (g_ad.conns[index].packet_sources[i].source_id > 0) + _webrtc_media_packet_source_set_buffer_state_changed_cb(0, g_ad.conns[index].packet_sources[i].source_id); } } @@ -2489,8 +2483,8 @@ static void _webrtc_unset_all_cbs(int index) _webrtc_unset_all_negotiation_state_change_cbs(index); for (i = 0; i < MAX_MEDIA_PACKET_SOURCE_LEN; i++) { - if (g_conns[index].packet_sources[i].source_id > 0) - _webrtc_media_packet_source_unset_buffer_state_changed_cb(0, g_conns[index].packet_sources[i].source_id); + if (g_ad.conns[index].packet_sources[i].source_id > 0) + _webrtc_media_packet_source_unset_buffer_state_changed_cb(0, g_ad.conns[index].packet_sources[i].source_id); } } @@ -2499,21 +2493,21 @@ static void _webrtc_create_data_channel(int index) int ret = WEBRTC_ERROR_NONE; gchar *label; - RET_IF(g_conns[index].channel_index == MAX_CHANNEL_LEN, "%d channels are already created, skip it", MAX_CHANNEL_LEN); + RET_IF(g_ad.conns[index].channel_index == MAX_CHANNEL_LEN, "%d channels are already created, skip it", MAX_CHANNEL_LEN); - label = g_strdup_printf("data_channel_%d_%d", index, g_conns[index].channel_index); + label = g_strdup_printf("data_channel_%d_%d", index, g_ad.conns[index].channel_index); - ret = webrtc_create_data_channel(g_conns[index].webrtc, label, NULL, &g_conns[index].channels[g_conns[index].channel_index]); + ret = webrtc_create_data_channel(g_ad.conns[index].webrtc, label, NULL, &g_ad.conns[index].channels[g_ad.conns[index].channel_index]); if (ret != WEBRTC_ERROR_NONE) { g_print("failed to webrtc_create_data_channel()\n"); } else { g_print("webrtc_create_data_channel() success, channel[%p], index[%d]\n", - g_conns[index].channels[g_conns[index].channel_index], g_conns[index].channel_index); - webrtc_data_channel_set_open_cb(g_conns[index].channels[g_conns[index].channel_index], __data_channel_open_cb, &g_conns[index]); - webrtc_data_channel_set_message_cb(g_conns[index].channels[g_conns[index].channel_index], __data_channel_message_cb, &g_conns[index]); - webrtc_data_channel_set_error_cb(g_conns[index].channels[g_conns[index].channel_index], __data_channel_error_cb, &g_conns[index]); - webrtc_data_channel_set_close_cb(g_conns[index].channels[g_conns[index].channel_index], __data_channel_close_cb, &g_conns[index]); - g_conns[index].channel_index++; + g_ad.conns[index].channels[g_ad.conns[index].channel_index], g_ad.conns[index].channel_index); + webrtc_data_channel_set_open_cb(g_ad.conns[index].channels[g_ad.conns[index].channel_index], __data_channel_open_cb, &g_ad.conns[index]); + webrtc_data_channel_set_message_cb(g_ad.conns[index].channels[g_ad.conns[index].channel_index], __data_channel_message_cb, &g_ad.conns[index]); + webrtc_data_channel_set_error_cb(g_ad.conns[index].channels[g_ad.conns[index].channel_index], __data_channel_error_cb, &g_ad.conns[index]); + webrtc_data_channel_set_close_cb(g_ad.conns[index].channels[g_ad.conns[index].channel_index], __data_channel_close_cb, &g_ad.conns[index]); + g_ad.conns[index].channel_index++; } g_free(label); @@ -2525,20 +2519,20 @@ static void _webrtc_destroy_data_channel(int index) int i; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - ret = webrtc_destroy_data_channel(g_conns[index].channels[i]); + ret = webrtc_destroy_data_channel(g_ad.conns[index].channels[i]); if (ret != WEBRTC_ERROR_NONE) { g_print("failed to _webrtc_destroy_data_channel(), index[%d]\n", i); } else { g_print("_webrtc_destroy_data_channel() success, send channel index[%d]\n", i); - g_conns[index].channels[i] = NULL; + g_ad.conns[index].channels[i] = NULL; } } - g_conns[index].channel_index = 0; + g_ad.conns[index].channel_index = 0; for (i = 0; i < MAX_CHANNEL_LEN; i++) - g_conns[index].recv_channels[i] = NULL; + g_ad.conns[index].recv_channels[i] = NULL; } static void _webrtc_data_channel_get_label(int index) @@ -2547,10 +2541,10 @@ static void _webrtc_data_channel_get_label(int index) char *label = NULL; for (i = 0; i < MAX_CHANNEL_LEN; i++) { - if (g_conns[index].channels[i] == NULL) + if (g_ad.conns[index].channels[i] == NULL) continue; - label = __get_channel_label(g_conns[index].channels[i]); + label = __get_channel_label(g_ad.conns[index].channels[i]); if (!label) continue; @@ -2568,7 +2562,7 @@ static void _setting_uri(gchar *dest_arr, char *uri) if (!uri) return; - if (g_signaling_server.signaling_client) { + if (g_ad.signaling_server.signaling_client) { g_printerr("already set by 'scc'\n"); return; } @@ -2584,14 +2578,14 @@ static void _setting_uri(gchar *dest_arr, char *uri) static void _request_session(int remote_peer_id) { - if (!g_signaling_server.ws_conn && !g_signaling_server.is_connected) { - g_printerr("server[%s] is not connected\n", g_signaling_server.url); + if (!g_ad.signaling_server.ws_conn && !g_ad.signaling_server.is_connected) { + g_printerr("server[%s] is not connected\n", g_ad.signaling_server.url); return; } - if (g_signaling_server.ws_conn) { + if (g_ad.signaling_server.ws_conn) { gchar *msg; - if (soup_websocket_connection_get_state(g_signaling_server.ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) { + if (soup_websocket_connection_get_state(g_ad.signaling_server.ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) { g_printerr("websocket is not opened\n"); return; } @@ -2599,7 +2593,7 @@ static void _request_session(int remote_peer_id) msg = g_strdup_printf("SESSION %d", remote_peer_id); g_print("\n[to SERVER > %s]\n", msg); - soup_websocket_connection_send_text(g_signaling_server.ws_conn, msg); + soup_websocket_connection_send_text(g_ad.signaling_server.ws_conn, msg); g_free(msg); } else { @@ -2613,14 +2607,14 @@ static void _request_join_room(char *room_name) RET_IF(!room_name, "room_name is NULL"); - if (!g_signaling_server.ws_conn && !g_signaling_server.is_connected) { - g_printerr("server[%s] is not connected\n", g_signaling_server.url); + if (!g_ad.signaling_server.ws_conn && !g_ad.signaling_server.is_connected) { + g_printerr("server[%s] is not connected\n", g_ad.signaling_server.url); return; } - RET_IF(!g_signaling_server.ws_conn, "ws_conn is NULL"); + RET_IF(!g_ad.signaling_server.ws_conn, "ws_conn is NULL"); - if (soup_websocket_connection_get_state(g_signaling_server.ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) { + if (soup_websocket_connection_get_state(g_ad.signaling_server.ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) { g_printerr("websocket is not opened\n"); return; } @@ -2628,7 +2622,7 @@ static void _request_join_room(char *room_name) msg = g_strdup_printf("ROOM %s", room_name); g_print("\n[to SERVER > %s]\n", msg); - soup_websocket_connection_send_text(g_signaling_server.ws_conn, msg); + soup_websocket_connection_send_text(g_ad.signaling_server.ws_conn, msg); g_free(msg); } @@ -2636,17 +2630,17 @@ static void _send_local_description(bool is_offer) { char *desc; - if (!g_signaling_server.ws_conn && !g_signaling_server.is_connected) { - g_printerr("server[%s] is not connected\n", g_signaling_server.url); + if (!g_ad.signaling_server.ws_conn && !g_ad.signaling_server.is_connected) { + g_printerr("server[%s] is not connected\n", g_ad.signaling_server.url); return; } - desc = is_offer ? g_conns[0].offer : g_conns[0].answer; + desc = is_offer ? g_ad.conns[0].offer : g_ad.conns[0].answer; g_print("\n[to SERVER > local description]\n%s\n", desc); - if (g_signaling_server.ws_conn) - soup_websocket_connection_send_text(g_signaling_server.ws_conn, desc); + if (g_ad.signaling_server.ws_conn) + soup_websocket_connection_send_text(g_ad.signaling_server.ws_conn, desc); else _webrtc_signaling_send_message(desc); } @@ -3644,21 +3638,21 @@ static void _start_pushing_packet(int index, int source_id) int i = _find_packet_sources_index(index, source_id); RET_IF(i == -1, "could not find media packet source id[%u]", source_id); - RET_IF(!g_conns[index].packet_sources[i].format, "please call webrtc_media_packet_source_set_format() first, press 'f'"); + RET_IF(!g_ad.conns[index].packet_sources[i].format, "please call webrtc_media_packet_source_set_format() first, press 'f'"); - if (!g_conns[index].packet_sources[i].src_pipeline) { - g_conns[index].packet_sources[i].src_pipeline = __make_src_pipeline(&g_conns[index].packet_sources[i]); - RET_IF(!g_conns[index].packet_sources[i].src_pipeline, "src_pipeline is NULL"); + if (!g_ad.conns[index].packet_sources[i].src_pipeline) { + g_ad.conns[index].packet_sources[i].src_pipeline = __make_src_pipeline(&g_ad.conns[index].packet_sources[i]); + RET_IF(!g_ad.conns[index].packet_sources[i].src_pipeline, "src_pipeline is NULL"); } - state_change_ret = gst_element_set_state(g_conns[index].packet_sources[i].src_pipeline, GST_STATE_PLAYING); + state_change_ret = gst_element_set_state(g_ad.conns[index].packet_sources[i].src_pipeline, GST_STATE_PLAYING); #ifdef __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__ - if (g_conns[index].packet_sources[i].render_pipeline) - state_change_ret = gst_element_set_state(g_conns[index].packet_sources[i].render_pipeline, GST_STATE_PLAYING); + if (g_ad.conns[index].packet_sources[i].render_pipeline) + state_change_ret = gst_element_set_state(g_ad.conns[index].packet_sources[i].render_pipeline, GST_STATE_PLAYING); #endif RET_IF(state_change_ret == GST_STATE_CHANGE_FAILURE, "failed to set state to PLAYING"); - g_conns[index].packet_sources[i].is_stop_requested = false; + g_ad.conns[index].packet_sources[i].is_stop_requested = false; g_print("_start_pushing_packet()\n"); } @@ -3668,17 +3662,17 @@ static void _stop_pushing_packet(int index, int source_id) int i = _find_packet_sources_index(index, source_id); RET_IF(i == -1, "could not find media packet source id[%u]", source_id); - RET_IF(!g_conns[index].packet_sources[i].src_pipeline, "already stopped for pushing packet"); + RET_IF(!g_ad.conns[index].packet_sources[i].src_pipeline, "already stopped for pushing packet"); - if (g_conns[index].packet_sources[i].is_overflowed) - g_cond_signal(&g_conns[index].packet_sources[i].cond); + if (g_ad.conns[index].packet_sources[i].is_overflowed) + g_cond_signal(&g_ad.conns[index].packet_sources[i].cond); - g_conns[index].packet_sources[i].is_stop_requested = true; + g_ad.conns[index].packet_sources[i].is_stop_requested = true; - gst_element_set_state(g_conns[index].packet_sources[i].src_pipeline, GST_STATE_PAUSED); + gst_element_set_state(g_ad.conns[index].packet_sources[i].src_pipeline, GST_STATE_PAUSED); #ifdef __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__ - if (g_conns[index].packet_sources[i].render_pipeline) - gst_element_set_state(g_conns[index].packet_sources[i].render_pipeline, GST_STATE_PAUSED); + if (g_ad.conns[index].packet_sources[i].render_pipeline) + gst_element_set_state(g_ad.conns[index].packet_sources[i].render_pipeline, GST_STATE_PAUSED); #endif g_print("_stop_pushing_packet()\n"); @@ -3688,7 +3682,7 @@ static void _set_media_path(int index, unsigned int source_id, const char *path) { int ret = 0; - ret = webrtc_file_source_set_path(g_conns[index].webrtc, source_id, path); + ret = webrtc_file_source_set_path(g_ad.conns[index].webrtc, source_id, path); if (ret != WEBRTC_ERROR_NONE) g_print("failed to webrtc_file_source_set_path(), source_id[%d], ret[0x%x]\n", source_id, ret); else @@ -3699,7 +3693,7 @@ static void _webrtc_file_source_set_looping(int index, unsigned int source_id, b { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_file_source_set_looping(g_conns[index].webrtc, source_id, looping); + ret = webrtc_file_source_set_looping(g_ad.conns[index].webrtc, source_id, looping); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_file_source_set_looping() success, source_id[%u] looping_state[%u]\n", source_id, looping); @@ -3710,7 +3704,7 @@ static void _webrtc_file_source_get_looping(int index, unsigned int source_id) int ret = WEBRTC_ERROR_NONE; bool looping_state; - ret = webrtc_file_source_get_looping(g_conns[index].webrtc, source_id, &looping_state); + ret = webrtc_file_source_get_looping(g_ad.conns[index].webrtc, source_id, &looping_state); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_file_source_get_looping() success, source_id[%u] looping_state[%u]\n", source_id, looping_state); @@ -3756,21 +3750,21 @@ static void __auto_configure_add_peer(gchar *peer_id, bool is_offer) for (i = 0; i < MAX_CONNECTION_LEN; i++) { unsigned int source_id = 0; - if (g_conns[i].webrtc) + if (g_ad.conns[i].webrtc) continue; - g_conns[i].remote_peer_id = atoi((const char *)peer_id); - g_conns[i].is_for_room = true; - g_conns[i].is_offer = is_offer; + g_ad.conns[i].remote_peer_id = atoi((const char *)peer_id); + g_ad.conns[i].is_for_room = true; + g_ad.conns[i].is_offer = is_offer; if (i > 0) /* follow the source type of the first one */ - g_conns[i].room_source_type = g_conns[0].room_source_type; + g_ad.conns[i].room_source_type = g_ad.conns[0].room_source_type; _webrtc_create(i); _webrtc_set_all_basic_cbs(i); _webrtc_set_all_negotiation_state_change_cbs(i); _webrtc_create_data_channel(i); - switch (g_conns[i].room_source_type) { + switch (g_ad.conns[i].room_source_type) { case 1: _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST, NULL); _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST, &source_id); @@ -3808,12 +3802,12 @@ static connection_s *__get_conn_from_peer_id(gchar *peer_id, int *index) } for (i = 0; i < MAX_CONNECTION_LEN; i++) { - if (g_conns[i].remote_peer_id <= 0) + if (g_ad.conns[i].remote_peer_id <= 0) continue; - if (g_conns[i].remote_peer_id == atoi((const char *)peer_id)) { + if (g_ad.conns[i].remote_peer_id == atoi((const char *)peer_id)) { if (index) *index = i; - return &g_conns[i]; + return &g_ad.conns[i]; } } @@ -3921,7 +3915,7 @@ static void __handle_room_related_message(const gchar *text) if (g_str_has_prefix(text, "ROOM_OK")) { g_print("\n[from SERVER > %s]\n", text); - g_signaling_server.server_status = SERVER_STATUS_ROOM_ESTABLISHED; + g_ad.signaling_server.server_status = SERVER_STATUS_ROOM_ESTABLISHED; /* parse text, get the previous peers in the room and negotiate to each other */ if (strlen(text) > strlen("ROOM_OK")) { @@ -3957,7 +3951,7 @@ static void __handle_room_related_message(const gchar *text) } else { g_print("\n[from SERVER > %s]\n", text); - g_signaling_server.server_status = SERVER_STATUS_ERROR_FOUND; + g_ad.signaling_server.server_status = SERVER_STATUS_ERROR_FOUND; } } @@ -3970,7 +3964,7 @@ static void __websocket_message_cb(SoupWebsocketConnection *ws_conn, SoupWebsock RET_IF(!conn, "conn is NULL"); RET_IF(!ws_conn, "ws_conn is NULL"); - RET_IF(g_signaling_server.ws_conn != ws_conn, "not matched ws_conn[%p, %p]", g_signaling_server.ws_conn, ws_conn); + RET_IF(g_ad.signaling_server.ws_conn != ws_conn, "not matched ws_conn[%p, %p]", g_ad.signaling_server.ws_conn, ws_conn); RET_IF(type != SOUP_WEBSOCKET_DATA_TEXT, "invalid data type(%d)", type); data = g_bytes_get_data(message, &size); @@ -3978,18 +3972,18 @@ static void __websocket_message_cb(SoupWebsocketConnection *ws_conn, SoupWebsock /* NOTE: Logics below can be different in each server */ if (g_strcmp0(text, "HELLO") == 0) { - g_print("\n[from SERVER > %s] registered done, local_peer_id[%d]\n", text, g_signaling_server.local_peer_id); + g_print("\n[from SERVER > %s] registered done, local_peer_id[%d]\n", text, g_ad.signaling_server.local_peer_id); } else if (g_strcmp0(text, "SESSION_OK") == 0) { g_print("\n[from SERVER > %s]\n", text); - g_signaling_server.server_status = SERVER_STATUS_SESSION_ESTABLISHED; + g_ad.signaling_server.server_status = SERVER_STATUS_SESSION_ESTABLISHED; } else if (g_str_has_prefix(text, "ROOM_")) { __handle_room_related_message(text); } else if (g_str_has_prefix(text, "ERROR")) { g_print("\n[from SERVER > %s]\n", text); - g_signaling_server.server_status = SERVER_STATUS_ERROR_FOUND; + g_ad.signaling_server.server_status = SERVER_STATUS_ERROR_FOUND; } else { __handle_json_structured_message(conn, text); @@ -4043,15 +4037,15 @@ static void __websocket_connected_cb(SoupSession *session, GAsyncResult *res, vo g_error_free(error); return; } - g_print("\n[%s] is connected, conn[%p]\n", g_signaling_server.url, conn); + g_print("\n[%s] is connected, conn[%p]\n", g_ad.signaling_server.url, conn); - g_signaling_server.ws_conn = ws_conn; - g_signaling_server.server_status = SERVER_STATUS_CONNECTED; + g_ad.signaling_server.ws_conn = ws_conn; + g_ad.signaling_server.server_status = SERVER_STATUS_CONNECTED; - g_signal_connect(ws_conn, "closed", G_CALLBACK(__websocket_closed_cb), &g_signaling_server); + g_signal_connect(ws_conn, "closed", G_CALLBACK(__websocket_closed_cb), &g_ad.signaling_server); g_signal_connect(ws_conn, "message", G_CALLBACK(__websocket_message_cb), conn); - g_signaling_server.local_peer_id = __send_greeting_to_register(ws_conn); + g_ad.signaling_server.local_peer_id = __send_greeting_to_register(ws_conn); } static void _connect_signaling_server(void) @@ -4061,14 +4055,14 @@ static void _connect_signaling_server(void) SoupURI *proxy_uri; const char *https_aliases[] = {"wss", NULL}; - RET_IF(g_signaling_server.signaling_client, "already set by 'scc'"); + RET_IF(g_ad.signaling_server.signaling_client, "already set by 'scc'"); - if (strlen(g_proxy) == 0) { + if (strlen(g_ad.proxy) == 0) { session = soup_session_new_with_options(SOUP_SESSION_SSL_STRICT, TRUE, SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL); } else { - g_print("use proxy [%s]\n", g_proxy); - proxy_uri = soup_uri_new(g_proxy); + g_print("use proxy [%s]\n", g_ad.proxy); + proxy_uri = soup_uri_new(g_ad.proxy); session = soup_session_new_with_options(SOUP_SESSION_SSL_STRICT, TRUE, SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE, SOUP_SESSION_PROXY_URI, proxy_uri, @@ -4077,29 +4071,29 @@ static void _connect_signaling_server(void) soup_uri_free(proxy_uri); } - message = soup_message_new(SOUP_METHOD_GET, g_signaling_server.url); + message = soup_message_new(SOUP_METHOD_GET, g_ad.signaling_server.url); - g_print("connecting to signaling server[%s]...\n", g_signaling_server.url); + g_print("connecting to signaling server[%s]...\n", g_ad.signaling_server.url); soup_session_websocket_connect_async(session, message, NULL, NULL, NULL, - (GAsyncReadyCallback) __websocket_connected_cb, &g_conns[0]); + (GAsyncReadyCallback) __websocket_connected_cb, &g_ad.conns[0]); } static void _webrtc_signaling_server_create(int port) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_server_create(port, &g_inner_signaling_server); + ret = webrtc_signaling_server_create(port, &(g_ad.inner_signaling_server)); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - g_print("webrtc_signaling_server_create() success, port[%d] g_inner_signaling_server[%p]\n", port, g_inner_signaling_server); + g_print("webrtc_signaling_server_create() success, port[%d] g_inner_signaling_server[%p]\n", port, g_ad.inner_signaling_server); } static void _webrtc_signaling_server_destroy(void) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_server_destroy(g_inner_signaling_server); + ret = webrtc_signaling_server_destroy(g_ad.inner_signaling_server); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_server_destroy() success\n"); @@ -4109,7 +4103,7 @@ static void _webrtc_signaling_server_start(void) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_server_start(g_inner_signaling_server); + ret = webrtc_signaling_server_start(g_ad.inner_signaling_server); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_server_start() success\n"); @@ -4119,7 +4113,7 @@ static void _webrtc_signaling_server_stop(void) { int ret = WEBRTC_ERROR_NONE; - ret = webrtc_signaling_server_stop(g_inner_signaling_server); + ret = webrtc_signaling_server_stop(g_ad.inner_signaling_server); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); g_print("webrtc_signaling_server_stop() success\n"); @@ -4129,14 +4123,14 @@ void quit_program(void) { int i; for (i = 0; i < MAX_CONNECTION_LEN; i++) { - if (g_conns[i].webrtc != NULL) { + if (g_ad.conns[i].webrtc != NULL) { _webrtc_stop(i); _webrtc_destroy(i); } } g_print("quit program\n"); - g_menu_status = CURRENT_STATUS_TERMINATE; + g_ad.menu_status = CURRENT_STATUS_TERMINATE; elm_exit(); } @@ -4157,7 +4151,7 @@ static bool interpret_main_menu_cmd(char *cmd) menu_info_s *menu_info = g_hash_table_lookup(g_ad.menu_items, cmd); if (!menu_info) break; - g_menu_status = menu_info->status; + g_ad.menu_status = menu_info->status; return menu_info->key_input_needed; } default: @@ -4178,7 +4172,7 @@ static void test_webrtc_common(char *cmd) { int value; - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_CREATE: _webrtc_create(0); reset_menu_state(); @@ -4244,22 +4238,22 @@ static void test_webrtc_common(char *cmd) } case CURRENT_STATUS_CREATE_OFFER: case CURRENT_STATUS_CREATE_OFFER_ASYNC: { - _webrtc_create_offer(&g_conns[0], g_menu_status == CURRENT_STATUS_CREATE_OFFER ? false : true); + _webrtc_create_offer(&g_ad.conns[0], g_ad.menu_status == CURRENT_STATUS_CREATE_OFFER ? false : true); reset_menu_state(); break; } case CURRENT_STATUS_CREATE_ANSWER: case CURRENT_STATUS_CREATE_ANSWER_ASYNC: { - _webrtc_create_answer(&g_conns[0], g_menu_status == CURRENT_STATUS_CREATE_ANSWER ? false: true); + _webrtc_create_answer(&g_ad.conns[0], g_ad.menu_status == CURRENT_STATUS_CREATE_ANSWER ? false: true); reset_menu_state(); break; } case CURRENT_STATUS_SET_LOCAL_DESCRIPTION: { value = atoi(cmd); if (value == 1) - _webrtc_set_local_description(&g_conns[0], true); + _webrtc_set_local_description(&g_ad.conns[0], true); else if (value == 2) - _webrtc_set_local_description(&g_conns[0], false); + _webrtc_set_local_description(&g_ad.conns[0], false); else g_print("invalid value[%d]\n", value); @@ -4267,12 +4261,12 @@ static void test_webrtc_common(char *cmd) break; } case CURRENT_STATUS_SET_REMOTE_DESCRIPTION: { - _webrtc_set_remote_description(&g_conns[0]); + _webrtc_set_remote_description(&g_ad.conns[0]); reset_menu_state(); break; } case CURRENT_STATUS_ADD_ICE_CANDIDATE: { - _webrtc_add_ice_candidate(&g_conns[0], NULL); + _webrtc_add_ice_candidate(&g_ad.conns[0], NULL); reset_menu_state(); break; } @@ -4293,7 +4287,7 @@ static void test_webrtc_common(char *cmd) } case CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY: { static bool sender; - switch (g_cnt) { + switch (g_ad.input_count) { case 0: value = atoi(cmd); if (value != 1 && value != 2) { @@ -4302,13 +4296,13 @@ static void test_webrtc_common(char *cmd) break; } sender = (value == 1); - g_cnt++; + g_ad.input_count++; break; case 1: { float fvalue = strtof(cmd, NULL); _webrtc_set_rtp_packet_drop_probability(sender, fvalue); sender = false; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4333,14 +4327,14 @@ static void test_webrtc_media_source(char *cmd) { int value; - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_ADD_MEDIA_SOURCE: { value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: if (value == 3) { /* mic source */ - g_cnt++; + g_ad.input_count++; } else { _webrtc_add_media_source(0, value - 1, NULL); reset_menu_state(); @@ -4348,7 +4342,7 @@ static void test_webrtc_media_source(char *cmd) break; case 1: _webrtc_add_mic_source(0, !(value == 0), NULL); - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4367,23 +4361,23 @@ static void test_webrtc_media_source(char *cmd) int new_state; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: media_type = value - 1; - g_cnt++; + g_ad.input_count++; break; case 2: new_state = value; - if (g_menu_status == CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE) + if (g_ad.menu_status == CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE) _webrtc_media_source_set_pause(0, id, media_type, new_state); else _webrtc_media_source_set_mute(0, id, media_type, new_state); id = media_type = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4394,18 +4388,18 @@ static void test_webrtc_media_source(char *cmd) static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: - if (g_menu_status == CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE) + if (g_ad.menu_status == CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE) _webrtc_media_source_get_pause(0, id, value - 1); else _webrtc_media_source_get_mute(0, id, value - 1); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4417,20 +4411,20 @@ static void test_webrtc_media_source(char *cmd) static int height; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: width = value; - g_cnt++; + g_ad.input_count++; break; case 2: height = value; _webrtc_media_source_set_video_resolution(0, id, width, height); id = width = height = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4447,16 +4441,16 @@ static void test_webrtc_media_source(char *cmd) static int framerate; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: framerate = value; _webrtc_media_source_set_video_framerate(0, id, framerate); id = framerate = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4472,15 +4466,15 @@ static void test_webrtc_media_source(char *cmd) static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: _set_media_path(0, id, cmd); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4489,15 +4483,15 @@ static void test_webrtc_media_source(char *cmd) case CURRENT_STATUS_FILE_SOURCE_SET_LOOPING: { static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: _webrtc_file_source_set_looping(0, id, value); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4525,15 +4519,15 @@ static void test_webrtc_media_source(char *cmd) static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: _webrtc_media_packet_source_set_format(0, id, value); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4545,20 +4539,20 @@ static void test_webrtc_media_source(char *cmd) static unsigned int direction; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: media_type = value - 1; - g_cnt++; + g_ad.input_count++; break; case 2: direction = value - 1; _webrtc_media_source_set_transceiver_direction(0, id, media_type, direction); id = media_type = direction = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4569,16 +4563,16 @@ static void test_webrtc_media_source(char *cmd) static unsigned int media_type; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: media_type = value - 1; _webrtc_media_source_get_transceiver_direction(0, id, media_type); id = media_type = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4602,33 +4596,33 @@ static void test_webrtc_media_source(char *cmd) static int is_vertical; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: x = value; - g_cnt++; + g_ad.input_count++; break; case 2: y = value; - g_cnt++; + g_ad.input_count++; break; case 3: w = value; - g_cnt++; + g_ad.input_count++; break; case 4: h = value; - g_cnt++; + g_ad.input_count++; break; case 5: is_vertical = value; int width, height; _webrtc_screen_source_set_crop(0, id, x, y, w, h, is_vertical, &width, &height); x = y = w = h = is_vertical = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4647,7 +4641,7 @@ static void test_webrtc_media_render(char *cmd) { int value; - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_SET_DISPLAY_TYPE: { value = atoi(cmd); _webrtc_set_display_type(0, value - 1); @@ -4658,15 +4652,15 @@ static void test_webrtc_media_render(char *cmd) static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: _webrtc_set_display_mode(0, id, value - 1); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4682,15 +4676,15 @@ static void test_webrtc_media_render(char *cmd) static unsigned int id; value = atoi(cmd); - switch (g_cnt) { + switch (g_ad.input_count) { case 0: id = value; - g_cnt++; + g_ad.input_count++; break; case 1: _webrtc_set_display_visible(0, id, value); id = 0; - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4751,7 +4745,7 @@ static void test_webrtc_media_render(char *cmd) static void test_webrtc_data_channel(char *cmd) { - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_DATA_CHANNEL_CREATE: { _webrtc_create_data_channel(0); reset_menu_state(); @@ -4807,7 +4801,7 @@ static void test_webrtc_data_channel(char *cmd) static void test_webrtc_stats(char *cmd) { - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_FOREACH_STATS: { _webrtc_foreach_stats(0, atoi(cmd)); reset_menu_state(); @@ -4820,14 +4814,14 @@ static void app_setting_and_signaling(char *cmd) { int value; - switch (g_menu_status) { + switch (g_ad.menu_status) { case CURRENT_STATUS_SETTING_SIGNALING_SERVER: { - _setting_uri(g_signaling_server.url, cmd); + _setting_uri(g_ad.signaling_server.url, cmd); reset_menu_state(); break; } case CURRENT_STATUS_SETTING_PROXY: { - _setting_uri(g_proxy, cmd); + _setting_uri(g_ad.proxy, cmd); reset_menu_state(); break; } @@ -4843,7 +4837,7 @@ static void app_setting_and_signaling(char *cmd) break; } case CURRENT_STATUS_REQUEST_JOIN_ROOM: { - switch (g_cnt) { + switch (g_ad.input_count) { case 0: value = atoi(cmd); if (value <= 0 || value > 3) { @@ -4851,12 +4845,12 @@ static void app_setting_and_signaling(char *cmd) reset_menu_state(); break; } - g_conns[0].room_source_type = value; - g_cnt++; + g_ad.conns[0].room_source_type = value; + g_ad.input_count++; break; case 1: _request_join_room(cmd); - g_cnt = 0; + g_ad.input_count = 0; reset_menu_state(); break; } @@ -4898,21 +4892,21 @@ static void app_setting_and_signaling(char *cmd) case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT: { static char *ip = NULL; - if (strlen(g_signaling_server.url) > 0) { - g_printerr("server[%s] is already set by 'ss'\n", g_signaling_server.url); + if (strlen(g_ad.signaling_server.url) > 0) { + g_printerr("server[%s] is already set by 'ss'\n", g_ad.signaling_server.url); reset_menu_state(); break; } - switch (g_cnt) { + switch (g_ad.input_count) { case 0: ip = strdup(cmd); - g_cnt++; + g_ad.input_count++; break; case 1: value = atoi(cmd); _webrtc_signaling_connect(ip, value); - g_cnt = 0; + g_ad.input_count = 0; if (ip) { free(ip); ip = NULL; @@ -4935,34 +4929,34 @@ static void interpret(char *cmd) if (!cmd) return; - if (g_menu_status == CURRENT_STATUS_MAINMENU) { + if (g_ad.menu_status == CURRENT_STATUS_MAINMENU) { if (interpret_main_menu_cmd(cmd)) goto out; } - if (g_menu_status & TEST_MENU_WEBRTC_COMMON) + if (g_ad.menu_status & TEST_MENU_WEBRTC_COMMON) test_webrtc_common(cmd); - else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) + else if (g_ad.menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) test_webrtc_media_source(cmd); - else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) + else if (g_ad.menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) test_webrtc_media_render(cmd); - else if (g_menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) + else if (g_ad.menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) test_webrtc_data_channel(cmd); - else if (g_menu_status & TEST_MENU_WEBRTC_STATS) + else if (g_ad.menu_status & TEST_MENU_WEBRTC_STATS) test_webrtc_stats(cmd); - else if (g_menu_status & TEST_MENU_APP_SIGNALING) + else if (g_ad.menu_status & TEST_MENU_APP_SIGNALING) app_setting_and_signaling(cmd); - else if (g_menu_status == CURRENT_STATUS_QUIT) + else if (g_ad.menu_status == CURRENT_STATUS_QUIT) quit_program(); out: - if (g_menu_status != CURRENT_STATUS_TERMINATE) + if (g_ad.menu_status != CURRENT_STATUS_TERMINATE) g_timeout_add(100, timeout_menu_display_cb, 0); } diff --git a/test/webrtc_test_menu.c b/test/webrtc_test_menu.c index cb60ab58..e346a82d 100644 --- a/test/webrtc_test_menu.c +++ b/test/webrtc_test_menu.c @@ -118,19 +118,19 @@ void display_handle_status(int index) webrtc_state_e state; char *stun_server = NULL; - if (g_conns[index].webrtc == NULL) + if (get_appdata()->conns[index].webrtc == NULL) return; - ret = webrtc_get_state(g_conns[index].webrtc, &state); + ret = webrtc_get_state(get_appdata()->conns[index].webrtc, &state); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - ret = webrtc_get_stun_server(g_conns[index].webrtc, &stun_server); + ret = webrtc_get_stun_server(get_appdata()->conns[index].webrtc, &stun_server); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - ret = webrtc_foreach_turn_server(g_conns[index].webrtc, foreach_turn_server, NULL); + ret = webrtc_foreach_turn_server(get_appdata()->conns[index].webrtc, foreach_turn_server, NULL); RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret); - g_print(" webrtc[%p]", g_conns[index].webrtc); + g_print(" webrtc[%p]", get_appdata()->conns[index].webrtc); g_print(" state[%s]", g_webrtc_state_str[state]); if (stun_server) { g_print(" STUN server[%s]", stun_server); @@ -142,22 +142,22 @@ void display_handle_status(int index) void display_setting_status(void) { - int len_proxy = strlen(g_proxy); - int len_server = strlen(g_signaling_server.url); + int len_proxy = strlen(get_appdata()->proxy); + int len_server = strlen(get_appdata()->signaling_server.url); int i; if (len_proxy > 0) - g_print(" proxy[%s]", g_proxy); + g_print(" proxy[%s]\n", get_appdata()->proxy); if (len_server > 0) - g_print(" server[%s][%s]\n", g_signaling_server.url, g_server_status_str[g_signaling_server.server_status]); - if (g_signaling_server.private_ip && g_signaling_server.port > 0) - g_print(" server[%s:%d][%s]\n", g_signaling_server.private_ip, g_signaling_server.port, g_server_status_str[g_signaling_server.server_status]); - if (g_signaling_server.local_peer_id > 0) - g_print(" local peer id : %d\n", g_signaling_server.local_peer_id); + g_print(" server[%s][%s]\n", get_appdata()->signaling_server.url, g_server_status_str[get_appdata()->signaling_server.server_status]); + if (get_appdata()->signaling_server.private_ip && get_appdata()->signaling_server.port > 0) + g_print(" server[%s:%d][%s]\n", get_appdata()->signaling_server.private_ip, get_appdata()->signaling_server.port, g_server_status_str[get_appdata()->signaling_server.server_status]); + if (get_appdata()->signaling_server.local_peer_id > 0) + g_print(" local peer id : %d\n", get_appdata()->signaling_server.local_peer_id); for (i = 0; i < MAX_CONNECTION_LEN; i++) { - if (g_conns[i].remote_peer_id == 0) + if (get_appdata()->conns[i].remote_peer_id == 0) continue; - g_print(" [%d] remote peer id : %d\n", i, g_conns[i].remote_peer_id); + g_print(" [%d] remote peer id : %d\n", i, get_appdata()->conns[i].remote_peer_id); } g_print("-----------------------------------------------------------------------------------------\n"); } @@ -263,7 +263,7 @@ void display_menu_main(void) void display_menu_webrtc_common(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_SET_STUN_SERVER: g_print("*** input STUN server address.\n"); break; @@ -280,13 +280,13 @@ void display_menu_webrtc_common(void) g_print("*** input type of local description.(1:offer, 2:answer)\n"); break; case CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input side.(1:sender, 2:receiver)\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input drop probability.(0 ~ 1.0)\n"); break; case CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input side.(1:sender, 2:receiver)\n"); break; } @@ -295,92 +295,92 @@ void display_menu_webrtc_common(void) void display_menu_webrtc_media_source(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_ADD_MEDIA_SOURCE: - if (g_cnt == 0) + 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:custom audio, 9:custom video)\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) 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"); break; case CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media type.(1:audio 2:video)\n"); - else if (g_cnt == 2) + 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 : - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media type.(1:audio 2:video)\n"); break; case CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media type.(1:audio 2:video)\n"); - else if (g_cnt == 2) + 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 (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + 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 (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input width.\n"); - else if (g_cnt == 2) + else if (get_appdata()->input_count == 2) g_print("*** input height.\n"); break; case CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_RESOLUTION: g_print("*** input source id.\n"); break; case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_FRAMERATE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input framerate.\n"); break; case CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_FRAMERATE: g_print("*** input source id.\n"); break; case CURRENT_STATUS_MEDIA_SOURCE_SET_TRANSCEIVER_DIRECTION: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media type.(1:audio 2:video)\n"); - else if (g_cnt == 2) + 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 (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + 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 (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media path.\n"); break; case CURRENT_STATUS_FILE_SOURCE_SET_LOOPING: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input looping state.(1:true 0:false)\n"); break; case CURRENT_STATUS_FILE_SOURCE_GET_LOOPING: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); break; case CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB: @@ -390,9 +390,9 @@ void display_menu_webrtc_media_source(void) g_print("*** input media packet source id to unset buffer state changed callback.\n"); break; case CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_FORMAT: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input media format.(1:I420 2:NV12 3:PCM_S16LE 4:H264)\n"); break; case CURRENT_STATUS_START_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE: @@ -402,21 +402,21 @@ void display_menu_webrtc_media_source(void) g_print("*** input media packet source id to stop pushing packet.\n"); break; case CURRENT_STATUS_SET_CROP_SCREEN_SOURCE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input x.\n"); - else if (g_cnt == 2) + else if (get_appdata()->input_count == 2) g_print("*** input y.\n"); - else if (g_cnt == 3) + else if (get_appdata()->input_count == 3) g_print("*** input width.\n"); - else if (g_cnt == 4) + else if (get_appdata()->input_count == 4) g_print("*** input height.\n"); - else if (g_cnt == 5) + else if (get_appdata()->input_count == 5) g_print("*** input whether screen rotates (0: horizontal, 1: vertical).\n"); break; case CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source id.\n"); break; } @@ -425,23 +425,23 @@ void display_menu_webrtc_media_source(void) void display_menu_webrtc_media_render(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_SET_DISPLAY_TYPE: g_print("*** input display type.(1:overlay 2:evas)\n"); break; case CURRENT_STATUS_SET_DISPLAY_MODE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input track id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input display mode.(1:letter-box 2:origin size 3:full)\n"); break; case CURRENT_STATUS_GET_DISPLAY_MODE: g_print("*** input track id.\n"); break; case CURRENT_STATUS_SET_DISPLAY_VISIBLE: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input track id.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input display visible.(1:true 0:false)\n"); break; case CURRENT_STATUS_GET_DISPLAY_VISIBLE: @@ -465,7 +465,7 @@ void display_menu_webrtc_media_render(void) void display_menu_webrtc_data_channel(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_DATA_CHANNEL_SEND_STRING: g_print("*** input string to send.\n"); break; @@ -484,9 +484,9 @@ void display_menu_webrtc_data_channel(void) void display_menu_webrtc_stats(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_FOREACH_STATS: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input stats type.(1:all, 2:codec, 3:inbound-rtp/remote-outbound-rtp, 4:outbound-rtp/remote-inbound-rtp)\n"); break; } @@ -495,7 +495,7 @@ void display_menu_webrtc_stats(void) void display_menu_app_signaling(void) { - switch (g_menu_status) { + switch (get_appdata()->menu_status) { case CURRENT_STATUS_SETTING_SIGNALING_SERVER: g_print("*** input signaling server URL.\n"); break; @@ -506,9 +506,9 @@ void display_menu_app_signaling(void) g_print("*** input remote peer id.\n"); break; case CURRENT_STATUS_REQUEST_JOIN_ROOM: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input source type.(1:audiotest/videotest 2:mic/camera 3:mic only)\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input room name to join.\n"); break; case CURRENT_STATUS_SEND_LOCAL_DESCRIPTION: @@ -518,9 +518,9 @@ void display_menu_app_signaling(void) g_print("*** input port.\n"); break; case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT: - if (g_cnt == 0) + if (get_appdata()->input_count == 0) g_print("*** input server ip.\n"); - else if (g_cnt == 1) + else if (get_appdata()->input_count == 1) g_print("*** input port.\n"); break; } @@ -529,30 +529,30 @@ void display_menu_app_signaling(void) void displaymenu(void) { - if (g_menu_status == CURRENT_STATUS_MAINMENU) { + if (get_appdata()->menu_status == CURRENT_STATUS_MAINMENU) { display_menu_main(); } else { - if (g_menu_status & TEST_MENU_WEBRTC_COMMON) { + if (get_appdata()->menu_status & TEST_MENU_WEBRTC_COMMON) { display_menu_webrtc_common(); - } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) { + } else if (get_appdata()->menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) { display_menu_webrtc_media_source(); - } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) { + } else if (get_appdata()->menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) { display_menu_webrtc_media_render(); - } else if (g_menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) { + } else if (get_appdata()->menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) { display_menu_webrtc_data_channel(); - } else if (g_menu_status & TEST_MENU_WEBRTC_STATS) { + } else if (get_appdata()->menu_status & TEST_MENU_WEBRTC_STATS) { display_menu_webrtc_stats(); - } else if (g_menu_status & TEST_MENU_APP_SIGNALING) { + } else if (get_appdata()->menu_status & TEST_MENU_APP_SIGNALING) { display_menu_app_signaling(); } else { - g_print("%s() > unknown menu status[0x%x]\n", __FUNCTION__, g_menu_status); + g_print("%s() > unknown menu status[0x%x]\n", __FUNCTION__, get_appdata()->menu_status); quit_program(); } } @@ -560,5 +560,5 @@ void displaymenu(void) void reset_menu_state(void) { - g_menu_status = CURRENT_STATUS_MAINMENU; + get_appdata()->menu_status = CURRENT_STATUS_MAINMENU; } \ No newline at end of file diff --git a/test/webrtc_test_priv.h b/test/webrtc_test_priv.h index 8ef90e46..941f37a3 100644 --- a/test/webrtc_test_priv.h +++ b/test/webrtc_test_priv.h @@ -161,13 +161,6 @@ enum { #define MAX_CHANNEL_LEN 10 #define MAX_MEDIA_PACKET_SOURCE_LEN 4 -typedef struct { - GHashTable *menu_items; - Evas_Object *win; - int win_width; - int win_height; -} appdata_s; - typedef struct { unsigned int source_id; media_format_h format; @@ -262,16 +255,29 @@ typedef struct { bool key_input_needed; } menu_info_s; +typedef struct { + GHashTable *menu_items; + int menu_status; + int input_count; + + Evas_Object *win; + int win_width; + int win_height; + Evas_Object *win_id; + Evas_Object *eo_mine; + Evas_Object *text_eo_mine; + + connection_s conns[MAX_CONNECTION_LEN]; + signaling_server_s signaling_server; + webrtc_signaling_server_h inner_signaling_server; + gchar proxy[MAX_STRING_LEN]; +} appdata_s; + extern menu_info_s g_menu_infos[]; extern const char *g_server_status_str[]; extern const char *g_webrtc_state_str[]; -extern int g_menu_status; -extern int g_cnt; -extern gchar g_proxy[]; -extern appdata_s g_ad; -extern connection_s g_conns[]; -extern signaling_server_s g_signaling_server; +appdata_s *get_appdata(void); void display_handle_status(int index); void display_setting_status(void); void display_menu_main(void);