Name: capi-media-webrtc
Summary: A WebRTC library in Tizen Native API
-Version: 0.3.103
+Version: 0.3.104
Release: 0
Group: Multimedia/API
License: Apache-2.0
CMAKE_MINIMUM_REQUIRED(VERSION 2.6)
SET(fw_test "${fw_name}-test")
+SET(test_name "webrtc_test")
INCLUDE_DIRECTORIES(../include)
FOREACH(src ${sources})
GET_FILENAME_COMPONENT(src_name ${src} NAME_WE)
+ LIST(APPEND src_list "${src_name}")
MESSAGE("${src_name}")
- ADD_EXECUTABLE(${src_name} ${src})
- TARGET_LINK_LIBRARIES(${src_name} capi-media-webrtc ${${fw_test}_LDFLAGS})
ENDFOREACH()
+ADD_EXECUTABLE(${test_name} ${src_list})
+TARGET_LINK_LIBRARIES(${test_name} capi-media-webrtc ${${fw_test}_LDFLAGS})
* limitations under the License.
*/
-#include <webrtc_internal.h>
-#include <media_format.h>
-#include <media_packet_internal.h>
-#include <sound_manager.h>
-#ifndef TIZEN_TV
-#include <esplusplayer_capi.h>
-#endif
-#include <appcore-efl.h>
-#include <Elementary.h>
+#include "webrtc_test_priv.h"
+
#include <tbm_surface_internal.h>
-#include <glib.h>
-#include <gst/gst.h>
-#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <sys/stat.h>
#include <fcntl.h>
#endif
#define PACKAGE "webrtc_test"
-//#define __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__
-//#define __DEBUG_VALIDATE_ENCODED_FRAME_CB__
-
#ifdef LOG_TAG
#undef LOG_TAG
#endif
#define LOG_TAG "WEBRTC_TEST"
-#define RET_IF(expr, fmt, arg...) \
-do { \
- if ((expr)) { \
- g_printerr("failed to %s(), "fmt"\n", __func__, ##arg); \
- return; \
- } \
-} while (0)
-
-#define MAX_STRING_LEN 512
-#define MAX_CHANNEL_LEN 10
-#define MAX_CONNECTION_LEN 3
-#define MAX_MEDIA_PACKET_SOURCE_LEN 4
#define MAX_EXPECTED_SIZE 1024 * 1024 * 1024
#define USE_GSTBUFFER_WITHOUT_COPY true
#define FONT_SIZE 30
-#define TEST_MENU_WEBRTC_COMMON 0x00001000
-#define TEST_MENU_WEBRTC_MEDIA_SOURCE 0x00002000
-#define TEST_MENU_WEBRTC_MEDIA_RENDER 0x00004000
-#define TEST_MENU_WEBRTC_DATA_CHANNEL 0x00008000
-#define TEST_MENU_WEBRTC_STATS 0x00010000
-#define TEST_MENU_APP_SIGNALING 0x00020000
-
-enum {
- CURRENT_STATUS_MAINMENU,
- CURRENT_STATUS_TERMINATE,
- CURRENT_STATUS_QUIT,
- /* webrtc common */
- CURRENT_STATUS_CREATE = TEST_MENU_WEBRTC_COMMON | 0x01,
- CURRENT_STATUS_START = TEST_MENU_WEBRTC_COMMON | 0x02,
- CURRENT_STATUS_STOP = TEST_MENU_WEBRTC_COMMON | 0x03,
- CURRENT_STATUS_DESTROY = TEST_MENU_WEBRTC_COMMON | 0x04,
- CURRENT_STATUS_GET_STATE = TEST_MENU_WEBRTC_COMMON | 0x05,
- CURRENT_STATUS_SET_STUN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x06,
- CURRENT_STATUS_GET_STUN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x07,
- CURRENT_STATUS_ADD_TURN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x08,
- CURRENT_STATUS_GET_TURN_SERVERS = TEST_MENU_WEBRTC_COMMON | 0x09,
- CURRENT_STATUS_SET_BUNDLE_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0A,
- CURRENT_STATUS_GET_BUNDLE_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0B,
- CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0C,
- CURRENT_STATUS_GET_ICE_TRANSPORT_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0D,
- CURRENT_STATUS_CREATE_OFFER = TEST_MENU_WEBRTC_COMMON | 0x0E,
- CURRENT_STATUS_CREATE_ANSWER = TEST_MENU_WEBRTC_COMMON | 0x0F,
- CURRENT_STATUS_CREATE_OFFER_ASYNC = TEST_MENU_WEBRTC_COMMON | 0x10,
- CURRENT_STATUS_CREATE_ANSWER_ASYNC = TEST_MENU_WEBRTC_COMMON | 0x11,
- CURRENT_STATUS_SET_LOCAL_DESCRIPTION = TEST_MENU_WEBRTC_COMMON | 0x12,
- CURRENT_STATUS_SET_REMOTE_DESCRIPTION = TEST_MENU_WEBRTC_COMMON | 0x13,
- CURRENT_STATUS_ADD_ICE_CANDIDATE = TEST_MENU_WEBRTC_COMMON | 0x14,
- CURRENT_STATUS_SET_ALL_CALLBACKS = TEST_MENU_WEBRTC_COMMON | 0x15,
- CURRENT_STATUS_UNSET_ALL_CALLBACKS = TEST_MENU_WEBRTC_COMMON | 0x16,
- CURRENT_STATUS_GET_ALL_NEGOTIATION_STATES = TEST_MENU_WEBRTC_COMMON | 0x17,
- CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY = TEST_MENU_WEBRTC_COMMON | 0x18,
- CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY = TEST_MENU_WEBRTC_COMMON | 0x19,
- /* webrtc media source */
- CURRENT_STATUS_ADD_MEDIA_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x01,
- CURRENT_STATUS_REMOVE_MEDIA_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x02,
- CURRENT_STATUS_MEDIA_SOURCE_SET_TRANSCEIVER_DIRECTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x03,
- CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x04,
- CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x05,
- CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x06,
- CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x07,
- CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x08,
- CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x09,
- CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_RESOLUTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0A,
- CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_FRAMERATE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0B,
- CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_FRAMERATE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0C,
- CURRENT_STATUS_FILE_SOURCE_SET_PATH = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0D,
- CURRENT_STATUS_FILE_SOURCE_SET_LOOPING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0E,
- CURRENT_STATUS_FILE_SOURCE_GET_LOOPING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0F,
- CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_FORMAT = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x10,
- CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x11,
- CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x12,
- CURRENT_STATUS_START_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x13,
- CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x14,
- CURRENT_STATUS_SET_CROP_SCREEN_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x15,
- CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x16,
- /* webrtc media render */
- CURRENT_STATUS_SET_DISPLAY_TYPE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x01,
- CURRENT_STATUS_SET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x02,
- CURRENT_STATUS_GET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x03,
- CURRENT_STATUS_SET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x04,
- CURRENT_STATUS_GET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x05,
- CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x06,
- CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x07,
- CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x08,
- CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x09,
- CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0A,
- CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0B,
- CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0C,
- CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0D,
- /* webrtc data channel */
- CURRENT_STATUS_DATA_CHANNEL_CREATE = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x01,
- CURRENT_STATUS_DATA_CHANNEL_DESTROY = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x02,
- CURRENT_STATUS_DATA_CHANNEL_GET_LABEL = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x03,
- CURRENT_STATUS_DATA_CHANNEL_SEND_STRING = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x04,
- CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x05,
- CURRENT_STATUS_DATA_CHANNEL_SEND_FILE = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x06,
- CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x07,
- CURRENT_STATUS_DATA_CHANNEL_UNSET_BUFFERED_AMOUNT_LOW_CB = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x08,
- CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT_LOW_THRESHOLD = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x09,
- CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x0A,
- /* webrtc stats */
- CURRENT_STATUS_FOREACH_STATS = TEST_MENU_WEBRTC_STATS | 0x01,
- /* app. setting & signaling */
- CURRENT_STATUS_SETTING_SIGNALING_SERVER = TEST_MENU_APP_SIGNALING | 0x01,
- CURRENT_STATUS_CONNECT_SIGNALING_SERVER = TEST_MENU_APP_SIGNALING | 0x02,
- CURRENT_STATUS_SETTING_PROXY = TEST_MENU_APP_SIGNALING | 0x03,
- CURRENT_STATUS_REQUEST_SESSION = TEST_MENU_APP_SIGNALING | 0x04,
- CURRENT_STATUS_REQUEST_JOIN_ROOM = TEST_MENU_APP_SIGNALING | 0x05,
- CURRENT_STATUS_SEND_LOCAL_DESCRIPTION = TEST_MENU_APP_SIGNALING | 0x06,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE = TEST_MENU_APP_SIGNALING | 0x07,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DESTROY = TEST_MENU_APP_SIGNALING | 0x08,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_START = TEST_MENU_APP_SIGNALING | 0x09,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_STOP = TEST_MENU_APP_SIGNALING | 0x0A,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT = TEST_MENU_APP_SIGNALING | 0x0B,
- CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DISCONNECT = TEST_MENU_APP_SIGNALING | 0x0C,
-};
-
-typedef struct {
- const char *cmd;
- int status;
- bool key_input_needed;
-} menu_info_s;
-
-menu_info_s g_menu_infos[] = {
- { "none", CURRENT_STATUS_MAINMENU, false },
- { "none", CURRENT_STATUS_TERMINATE, false },
- { "q", CURRENT_STATUS_QUIT, false },
- /* webrtc common */
- { "c", CURRENT_STATUS_CREATE, false },
- { "s", CURRENT_STATUS_START, false },
- { "t", CURRENT_STATUS_STOP, false },
- { "d", CURRENT_STATUS_DESTROY, false },
- { "g", CURRENT_STATUS_GET_STATE, false },
- { "st", CURRENT_STATUS_SET_STUN_SERVER, true },
- { "gt", CURRENT_STATUS_GET_STUN_SERVER, false },
- { "su", CURRENT_STATUS_ADD_TURN_SERVER, true },
- { "gu", CURRENT_STATUS_GET_TURN_SERVERS, false },
- { "sbp", CURRENT_STATUS_SET_BUNDLE_POLICY, true },
- { "gbp", CURRENT_STATUS_GET_BUNDLE_POLICY, false },
- { "stp", CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY, true },
- { "gtp", CURRENT_STATUS_GET_ICE_TRANSPORT_POLICY, false },
- { "co", CURRENT_STATUS_CREATE_OFFER, false },
- { "ca", CURRENT_STATUS_CREATE_ANSWER, false },
- { "coa", CURRENT_STATUS_CREATE_OFFER_ASYNC, false },
- { "caa", CURRENT_STATUS_CREATE_ANSWER_ASYNC, false },
- { "sl", CURRENT_STATUS_SET_LOCAL_DESCRIPTION, true },
- { "sr", CURRENT_STATUS_SET_REMOTE_DESCRIPTION, false },
- { "ac", CURRENT_STATUS_ADD_ICE_CANDIDATE, false },
- { "sac", CURRENT_STATUS_SET_ALL_CALLBACKS, false },
- { "uac", CURRENT_STATUS_UNSET_ALL_CALLBACKS, false },
- { "gan", CURRENT_STATUS_GET_ALL_NEGOTIATION_STATES, false },
- { "sdp", CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY, true },
- { "gdp", CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY, true },
- /* webrtc media source */
- { "a", CURRENT_STATUS_ADD_MEDIA_SOURCE, true },
- { "r", CURRENT_STATUS_REMOVE_MEDIA_SOURCE, true },
- { "td", CURRENT_STATUS_MEDIA_SOURCE_SET_TRANSCEIVER_DIRECTION, true },
- { "gd", CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION, true },
- { "p", CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE, true },
- { "o", CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE, true },
- { "mu", CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE, true },
- { "mg", CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE, true },
- { "v", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION, true },
- { "l", CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_RESOLUTION, true },
- { "f", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_FRAMERATE, true },
- { "m", CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_FRAMERATE, true },
- { "pa", CURRENT_STATUS_FILE_SOURCE_SET_PATH, true },
- { "sfl", CURRENT_STATUS_FILE_SOURCE_SET_LOOPING, true },
- { "gfl", CURRENT_STATUS_FILE_SOURCE_GET_LOOPING, true },
- { "sf", CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_FORMAT, true },
- { "sm", CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB, true },
- { "um", CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB, true },
- { "sp", CURRENT_STATUS_START_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE, true },
- { "tp", CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE, true },
- { "scs", CURRENT_STATUS_SET_CROP_SCREEN_SOURCE, true },
- { "ucs", CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE, true },
- /* webrtc media render */
- { "dt", CURRENT_STATUS_SET_DISPLAY_TYPE, true },
- { "dm", CURRENT_STATUS_SET_DISPLAY_MODE, true },
- { "gm", CURRENT_STATUS_GET_DISPLAY_MODE, true },
- { "dv", CURRENT_STATUS_SET_DISPLAY_VISIBLE, true },
- { "gv", CURRENT_STATUS_GET_DISPLAY_VISIBLE, true },
- { "sa", CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB, false },
- { "ua", CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB, false },
- { "sv", CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB, false },
- { "uv", CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB, false },
- { "al", CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK, true },
- { "ual", CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK, true },
- { "vl", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK, true },
- { "uvl", CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK, true },
- /* webrtc data channel */
- { "cd", CURRENT_STATUS_DATA_CHANNEL_CREATE, false },
- { "dd", CURRENT_STATUS_DATA_CHANNEL_DESTROY, false },
- { "dl", CURRENT_STATUS_DATA_CHANNEL_GET_LABEL, false },
- { "zs", CURRENT_STATUS_DATA_CHANNEL_SEND_STRING, true },
- { "zb", CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES, true },
- { "zf", CURRENT_STATUS_DATA_CHANNEL_SEND_FILE, true },
- { "sbc", CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB, true },
- { "ubc", CURRENT_STATUS_DATA_CHANNEL_UNSET_BUFFERED_AMOUNT_LOW_CB, false },
- { "gbt", CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT_LOW_THRESHOLD, false },
- { "ba", CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT, false },
- /* webrtc stats */
- { "sts", CURRENT_STATUS_FOREACH_STATS, true },
- /* app. setting & signaling */
- { "ss", CURRENT_STATUS_SETTING_SIGNALING_SERVER, true },
- { "cs", CURRENT_STATUS_CONNECT_SIGNALING_SERVER, false },
- { "px", CURRENT_STATUS_SETTING_PROXY, true },
- { "rs", CURRENT_STATUS_REQUEST_SESSION, true },
- { "rj", CURRENT_STATUS_REQUEST_JOIN_ROOM, true },
- { "sd", CURRENT_STATUS_SEND_LOCAL_DESCRIPTION, true },
- { "ssc", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE, true },
- { "ssd", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DESTROY, false },
- { "sss", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_START, false },
- { "sst", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_STOP, false },
- { "scc", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT, true },
- { "scd", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DISCONNECT, false },
- { NULL, -1, false },
-};
-
-enum {
- SERVER_STATUS_DISCONNECTED,
- SERVER_STATUS_CONNECTED,
- SERVER_STATUS_SESSION_ESTABLISHED,
- SERVER_STATUS_SESSION_CLOSED,
- SERVER_STATUS_ROOM_ESTABLISHED,
- SERVER_STATUS_ERROR_FOUND
-};
-
const char *g_server_status_str[] = {
[SERVER_STATUS_DISCONNECTED] = "DISCONNECTED",
[SERVER_STATUS_CONNECTED] = "CONNECTED",
[SERVER_STATUS_ERROR_FOUND] = "ERROR_FOUND",
};
-static const char *g_webrtc_state_str[] = {
+const char *g_webrtc_state_str[] = {
[WEBRTC_STATE_IDLE] = "IDLE",
[WEBRTC_STATE_NEGOTIATING] = "NEGOTIATING",
[WEBRTC_STATE_PLAYING] = "PLAYING",
[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;
-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;
-
- webrtc_h webrtc;
- GstElement *src_pipeline;
-#ifdef __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__
- GstElement *render_pipeline;
- GstElement *appsrc;
-#endif
- GstElement *src;
- GstElement *sink;
- GstElement *demux;
- GstBus *bus;
- guint bus_watcher;
- gulong handoff_signal_id;
- gulong pad_added_signal_id;
- bool is_overflowed;
- bool is_stop_requested;
- GCond cond;
- GMutex mutex;
- bool got_eos;
-} media_packet_source_s;
-
-typedef struct _connection_s {
- int index;
- int remote_peer_id;
-
- bool is_for_room;
- bool is_offer;
- int room_source_type;
-
- webrtc_h webrtc;
- webrtc_data_channel_h channels[MAX_CHANNEL_LEN];
- int channel_index;
- webrtc_data_channel_h recv_channels[MAX_CHANNEL_LEN];
- char *offer;
- char *answer;
- char *remote_desc;
- GList *ice_candidates;
-
- /* receive data & dump file */
- gint64 sum_size;
- gchar *expected_name;
- gint64 expected_size;
- char* receive_buffer;
-
- struct {
- sound_stream_info_h stream_info;
- } source;
-
- struct {
- sound_stream_info_h stream_info;
- webrtc_display_type_e display_type;
- Evas_Object *eo;
- Evas_Object *text_eo;
- unsigned int loopback_track_id;
-#ifndef TIZEN_TV
- esplusplayer_handle espp;
-#endif
- } render;
-
-#ifndef TIZEN_TV
- bool encoded_video_frame_cb_is_set;
- bool encoded_audio_frame_cb_is_set;
-#endif
-#ifdef __DEBUG_VALIDATE_ENCODED_FRAME_CB__
- GstElement *audio_render_pipeline;
- GstElement *video_render_pipeline;
- GstElement *appsrc_for_audio;
- GstElement *appsrc_for_video;
-#endif
- media_packet_source_s packet_sources[MAX_MEDIA_PACKET_SOURCE_LEN];
-} connection_s;
-
-typedef struct _signaling_server_s {
- gchar url[MAX_STRING_LEN];
- SoupWebsocketConnection *ws_conn;
- int server_status;
- gint32 local_peer_id;
-
- /* for private network - internal API */
- webrtc_signaling_client_h signaling_client;
- char *private_ip;
- int port;
- bool is_connected;
-} signaling_server_s;
-
-static gchar g_proxy[MAX_STRING_LEN];
-
-static appdata_s g_ad;
-static connection_s g_conns[MAX_CONNECTION_LEN];
-static signaling_server_s g_signaling_server;
-static int g_menu_status;
-static int g_cnt;
-
-static webrtc_signaling_server_h g_inner_signaling_server;
-
#if defined(__DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__) || defined(__DEBUG_VALIDATE_ENCODED_FRAME_CB__)
GstBuffer *__alloc_buffer_from_packet(media_packet_h packet);
#endif
g_print("webrtc_add_turn_server() success, uri[%s]\n", uri);
}
-static bool __foreach_turn_server(const char *turn_server, gpointer user_data)
+bool foreach_turn_server(const char *turn_server, gpointer user_data)
{
g_print("- turn server %s\n", turn_server);
return true;
{
int ret = 0;
- ret = webrtc_foreach_turn_server(g_conns[index].webrtc, __foreach_turn_server, NULL);
+ ret = webrtc_foreach_turn_server(g_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");
g_print("webrtc_signaling_server_stop() success\n");
}
-static void quit_program()
+void quit_program(void)
{
int i;
for (i = 0; i < MAX_CONNECTION_LEN; i++) {
return true;
}
-static void display_handle_status(int index)
-{
- int ret = WEBRTC_ERROR_NONE;
- webrtc_state_e state;
- char *stun_server = NULL;
-
- if (g_conns[index].webrtc == NULL)
- return;
-
- ret = webrtc_get_state(g_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_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
-
- ret = webrtc_foreach_turn_server(g_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(" state[%s]", g_webrtc_state_str[state]);
- if (stun_server) {
- g_print(" STUN server[%s]", stun_server);
- free(stun_server);
- }
-
- g_print("\n-----------------------------------------------------------------------------------------\n");
-}
-
-static void display_setting_status(void)
-{
- int len_proxy = strlen(g_proxy);
- int len_server = strlen(g_signaling_server.url);
- int i;
-
- if (len_proxy > 0)
- g_print(" proxy[%s]", g_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);
- for (i = 0; i < MAX_CONNECTION_LEN; i++) {
- if (g_conns[i].remote_peer_id == 0)
- continue;
- g_print(" [%d] remote peer id : %d\n", i, g_conns[i].remote_peer_id);
- }
- g_print("-----------------------------------------------------------------------------------------\n");
-}
-
-static void display_menu_main(void)
-{
- g_print("\n");
- g_print("=========================================================================================\n");
- g_print(" Native WebRTC Test (press q to quit, * for internal API)\n");
- g_print("-----------------------------------------------------------------------------------------\n");
- display_handle_status(0);
- g_print("c. Create\t");
- g_print("d. Destroy\t");
- g_print("s. Start\t");
- g_print("t. Stop\t\t");
- g_print("g. Get state\n");
- g_print("sac. Set all callbacks\t");
- g_print("uac. Unset all callbacks\n");
- g_print("gan. Gets all the negotiation states\n");
- g_print("st. Set STUN server\t");
- g_print("gt. Get STUN server\t");
- g_print("su. Add TURN server\t");
- g_print("gu. Get TURN servers\n");
- g_print("sbp. Set bundle policy\t");
- g_print("gbp. Get bundle policy\n");
- g_print("stp. Set ICE transport policy\t");
- g_print("gtp. Get ICE transport policy\n");
- g_print("co. Create offer\t");
- g_print("ca. Create answer\n");
- g_print("coa. Create offer(async)\t");
- g_print("caa. Create answer(async)\n");
- g_print("sl. Set local description\t");
- g_print("sr. Set remote description\n");
- g_print("ac. Add ICE candidate\n");
- g_print("sdp. *Set RTP packet drop probability\t");
- g_print("gdp. *Get RTP packet drop probability\n");
- g_print("------------------------------------- Media Source --------------------------------------\n");
- g_print("a. Add media source\t");
- g_print("r. Remove media source\n");
- g_print("p. Pause/play media source\t");
- g_print("o. Get the media source pause\n");
- g_print("mu. Mute/unmute media source\t");
- g_print("mg. Get the media source mute\n");
- g_print("v. Set video resolution\t");
- g_print("l. Get video resolution\n");
- g_print("f. Set video framerate\t");
- g_print("m. Get video framerate\n");
- g_print("td. Set transceiver direction\t");
- g_print("gd. Get transceiver direction\n");
- g_print("pa. Set media path to file source\n");
- g_print("sfl. Set file source looping\t");
- g_print("gfl. Set file source looping\n");
- g_print("sf. Set media format to media packet source\n");
- g_print("sp. Start pushing packet to media packet source\t");
- g_print("tp. Stop pushing packet to media packet source\n");
- g_print("scs. *Set crop screen source\t");
- g_print("ucs. *Unset crop screen source\n");
- g_print("------------------------------------- Media Render --------------------------------------\n");
- g_print("dt. Set display type\t");
- g_print("dm. Set display mode\t");
- g_print("gm. Get display mode\n");
- g_print("dv. Set display visible\t");
- g_print("gv. Get display visible\n");
- g_print("al. Set audio loopback\t");
- g_print("ual. Unset audio loopback\n");
- g_print("vl. Set video loopback\t");
- g_print("uvl. Unset video loopback\n");
- g_print("sa. Set encoded audio frame callback\t");
- g_print("ua. Unset encoded audio frame callback\n");
- g_print("sv. Set encoded video frame callback\t");
- g_print("uv. Unset encoded video frame callback\n");
- g_print("------------------------------------- Data Channel --------------------------------------\n");
- g_print("cd. Create data channel\t");
- g_print("dd. Destroy data channel\n");
- g_print("dl. Get data channel label\n");
- g_print("zs. Send string via data channel\n");
- g_print("zb. Send string as bytes data via data channel\t");
- g_print("zf. Send file via data channel\n");
- g_print("ba. Get buffered amount\n");
- g_print("sbc. Set buffered amount low callback\t");
- g_print("ubc. Unset buffered amount low callback\n");
- g_print("gbt. Get buffered amount low threshold\n");
- g_print("---------------------------------------- Stats ------------------------------------------\n");
- g_print("sts. Get stats\n");
- g_print("------------------------------- App. Setting & Signaling --------------------------------\n");
- display_setting_status();
- g_print("px. Set proxy URL\n");
- g_print("ss. Set signaling server URL\n");
- g_print("cs. Connect to the signaling server\n");
- g_print("rs. Request session of remote peer id\n");
- g_print("rj. Request join room\n");
- g_print("sd. Send local description\n");
- g_print("ssc. *Create signaling server\t");
- g_print("ssd. *Destroy signaling server\n");
- g_print("sss. *Start signaling server\t");
- g_print("sst. *Stop signaling server\n");
- g_print("scc. *Connect to signaling server\t");
- g_print("scd. *Disconnect from signaling server\n");
- g_print("-----------------------------------------------------------------------------------------\n");
- g_print("=========================================================================================\n");
- g_print(" >>> ");
-}
-
-static void display_menu_webrtc_common(void)
-{
- switch (g_menu_status) {
- case CURRENT_STATUS_SET_STUN_SERVER:
- g_print("*** input STUN server address.\n");
- break;
- case CURRENT_STATUS_ADD_TURN_SERVER:
- g_print("*** input TURN server address.\n");
- break;
- case CURRENT_STATUS_SET_BUNDLE_POLICY:
- g_print("*** input bundle policy.(0:none, 1:max-bundle)\n");
- break;
- case CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY:
- g_print("*** input ICE transport policy.(0:all, 1:relay)\n");
- break;
- case CURRENT_STATUS_SET_LOCAL_DESCRIPTION:
- 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)
- g_print("*** input side.(1:sender, 2:receiver)\n");
- else if (g_cnt == 1)
- g_print("*** input drop probability.(0 ~ 1.0)\n");
- break;
- case CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY:
- if (g_cnt == 0)
- g_print("*** input side.(1:sender, 2:receiver)\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void display_menu_webrtc_media_source(void)
-{
- switch (g_menu_status) {
- case CURRENT_STATUS_ADD_MEDIA_SOURCE:
- 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");
- 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)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- else if (g_cnt == 2)
- g_print("*** input pause or play.(1:pause, 0:play)\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE :
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- else if (g_cnt == 2)
- g_print("*** input mute mode.(1:mute 0:unmute)\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input width.\n");
- else if (g_cnt == 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)
- g_print("*** input source id.\n");
- else if (g_cnt == 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)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- else if (g_cnt == 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)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media type.(1:audio 2:video)\n");
- break;
- case CURRENT_STATUS_FILE_SOURCE_SET_PATH:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input media path.\n");
- break;
- case CURRENT_STATUS_FILE_SOURCE_SET_LOOPING:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input looping state.(1:true 0:false)\n");
- break;
- case CURRENT_STATUS_FILE_SOURCE_GET_LOOPING:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- break;
- case CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB:
- g_print("*** input media packet source id to set buffer state changed callback.\n");
- break;
- case CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB:
- 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)
- g_print("*** input source id.\n");
- else if (g_cnt == 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:
- g_print("*** input media packet source id to start pushing packet.\n");
- break;
- case CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE:
- g_print("*** input media packet source id to stop pushing packet.\n");
- break;
- case CURRENT_STATUS_SET_CROP_SCREEN_SOURCE:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- else if (g_cnt == 1)
- g_print("*** input x.\n");
- else if (g_cnt == 2)
- g_print("*** input y.\n");
- else if (g_cnt == 3)
- g_print("*** input width.\n");
- else if (g_cnt == 4)
- g_print("*** input height.\n");
- else if (g_cnt == 5)
- g_print("*** input whether screen rotates (0: horizontal, 1: vertical).\n");
- break;
- case CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE:
- if (g_cnt == 0)
- g_print("*** input source id.\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void display_menu_webrtc_media_render(void)
-{
- switch (g_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)
- g_print("*** input track id.\n");
- else if (g_cnt == 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)
- g_print("*** input track id.\n");
- else if (g_cnt == 1)
- g_print("*** input display visible.(1:true 0:false)\n");
- break;
- case CURRENT_STATUS_GET_DISPLAY_VISIBLE:
- g_print("*** input track id.\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK:
- g_print("*** input source id.\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK:
- g_print("*** input source id.\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK:
- g_print("*** input source id.\n");
- break;
- case CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK:
- g_print("*** input source id.\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void display_menu_webrtc_data_channel(void)
-{
- switch (g_menu_status) {
- case CURRENT_STATUS_DATA_CHANNEL_SEND_STRING:
- g_print("*** input string to send.\n");
- break;
- case CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES:
- g_print("*** input string to send.(it will be converted to bytes data)\n");
- break;
- case CURRENT_STATUS_DATA_CHANNEL_SEND_FILE:
- g_print("*** input file path to send.\n");
- break;
- case CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB:
- g_print("*** input data channel buffered amount low threshold.\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void display_menu_webrtc_stats(void)
-{
- switch (g_menu_status) {
- case CURRENT_STATUS_FOREACH_STATS:
- if (g_cnt == 0)
- g_print("*** input stats type.(1:all, 2:codec, 3:inbound-rtp/remote-outbound-rtp, 4:outbound-rtp/remote-inbound-rtp)\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void display_menu_app_signaling(void)
-{
- switch (g_menu_status) {
- case CURRENT_STATUS_SETTING_SIGNALING_SERVER:
- g_print("*** input signaling server URL.\n");
- break;
- case CURRENT_STATUS_SETTING_PROXY:
- g_print("*** input proxy URL.\n");
- break;
- case CURRENT_STATUS_REQUEST_SESSION:
- g_print("*** input remote peer id.\n");
- break;
- case CURRENT_STATUS_REQUEST_JOIN_ROOM:
- if (g_cnt == 0)
- g_print("*** input source type.(1:audiotest/videotest 2:mic/camera 3:mic only)\n");
- else if (g_cnt == 1)
- g_print("*** input room name to join.\n");
- break;
- case CURRENT_STATUS_SEND_LOCAL_DESCRIPTION:
- g_print("*** input type of local description to send to the server.(1:offer, 2:answer)\n");
- break;
- case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE:
- g_print("*** input port.\n");
- break;
- case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT:
- if (g_cnt == 0)
- g_print("*** input server ip.\n");
- else if (g_cnt == 1)
- g_print("*** input port.\n");
- break;
- }
- g_print(" >>> ");
-}
-
-static void displaymenu(void)
-{
- if (g_menu_status == CURRENT_STATUS_MAINMENU) {
- display_menu_main();
-
- } else {
- if (g_menu_status & TEST_MENU_WEBRTC_COMMON) {
- display_menu_webrtc_common();
-
- } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) {
- display_menu_webrtc_media_source();
-
- } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) {
- display_menu_webrtc_media_render();
-
- } else if (g_menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) {
- display_menu_webrtc_data_channel();
-
- } else if (g_menu_status & TEST_MENU_WEBRTC_STATS) {
- display_menu_webrtc_stats();
-
- } else if (g_menu_status & TEST_MENU_APP_SIGNALING) {
- display_menu_app_signaling();
-
- } else {
- g_print("%s() > unknown menu status[0x%x]\n", __FUNCTION__, g_menu_status);
- quit_program();
- }
- }
-}
-
static gboolean timeout_menu_display_cb(void *data)
{
displaymenu();
return FALSE;
}
-static void reset_menu_state(void)
-{
- g_menu_status = CURRENT_STATUS_MAINMENU;
-}
-
static void test_webrtc_common(char *cmd)
{
int value;
--- /dev/null
+/*
+ * Copyright (c) 2022 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "webrtc_test_priv.h"
+
+menu_info_s g_menu_infos[] = {
+ { "none", CURRENT_STATUS_MAINMENU, false },
+ { "none", CURRENT_STATUS_TERMINATE, false },
+ { "q", CURRENT_STATUS_QUIT, false },
+ /* webrtc common */
+ { "c", CURRENT_STATUS_CREATE, false },
+ { "s", CURRENT_STATUS_START, false },
+ { "t", CURRENT_STATUS_STOP, false },
+ { "d", CURRENT_STATUS_DESTROY, false },
+ { "g", CURRENT_STATUS_GET_STATE, false },
+ { "st", CURRENT_STATUS_SET_STUN_SERVER, true },
+ { "gt", CURRENT_STATUS_GET_STUN_SERVER, false },
+ { "su", CURRENT_STATUS_ADD_TURN_SERVER, true },
+ { "gu", CURRENT_STATUS_GET_TURN_SERVERS, false },
+ { "sbp", CURRENT_STATUS_SET_BUNDLE_POLICY, true },
+ { "gbp", CURRENT_STATUS_GET_BUNDLE_POLICY, false },
+ { "stp", CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY, true },
+ { "gtp", CURRENT_STATUS_GET_ICE_TRANSPORT_POLICY, false },
+ { "co", CURRENT_STATUS_CREATE_OFFER, false },
+ { "ca", CURRENT_STATUS_CREATE_ANSWER, false },
+ { "coa", CURRENT_STATUS_CREATE_OFFER_ASYNC, false },
+ { "caa", CURRENT_STATUS_CREATE_ANSWER_ASYNC, false },
+ { "sl", CURRENT_STATUS_SET_LOCAL_DESCRIPTION, true },
+ { "sr", CURRENT_STATUS_SET_REMOTE_DESCRIPTION, false },
+ { "ac", CURRENT_STATUS_ADD_ICE_CANDIDATE, false },
+ { "sac", CURRENT_STATUS_SET_ALL_CALLBACKS, false },
+ { "uac", CURRENT_STATUS_UNSET_ALL_CALLBACKS, false },
+ { "gan", CURRENT_STATUS_GET_ALL_NEGOTIATION_STATES, false },
+ { "sdp", CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY, true },
+ { "gdp", CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY, true },
+ /* webrtc media source */
+ { "a", CURRENT_STATUS_ADD_MEDIA_SOURCE, true },
+ { "r", CURRENT_STATUS_REMOVE_MEDIA_SOURCE, true },
+ { "td", CURRENT_STATUS_MEDIA_SOURCE_SET_TRANSCEIVER_DIRECTION, true },
+ { "gd", CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION, true },
+ { "p", CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE, true },
+ { "o", CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE, true },
+ { "mu", CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE, true },
+ { "mg", CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE, true },
+ { "v", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION, true },
+ { "l", CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_RESOLUTION, true },
+ { "f", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_FRAMERATE, true },
+ { "m", CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_FRAMERATE, true },
+ { "pa", CURRENT_STATUS_FILE_SOURCE_SET_PATH, true },
+ { "sfl", CURRENT_STATUS_FILE_SOURCE_SET_LOOPING, true },
+ { "gfl", CURRENT_STATUS_FILE_SOURCE_GET_LOOPING, true },
+ { "sf", CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_FORMAT, true },
+ { "sm", CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB, true },
+ { "um", CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB, true },
+ { "sp", CURRENT_STATUS_START_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE, true },
+ { "tp", CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE, true },
+ { "scs", CURRENT_STATUS_SET_CROP_SCREEN_SOURCE, true },
+ { "ucs", CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE, true },
+ /* webrtc media render */
+ { "dt", CURRENT_STATUS_SET_DISPLAY_TYPE, true },
+ { "dm", CURRENT_STATUS_SET_DISPLAY_MODE, true },
+ { "gm", CURRENT_STATUS_GET_DISPLAY_MODE, true },
+ { "dv", CURRENT_STATUS_SET_DISPLAY_VISIBLE, true },
+ { "gv", CURRENT_STATUS_GET_DISPLAY_VISIBLE, true },
+ { "sa", CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB, false },
+ { "ua", CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB, false },
+ { "sv", CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB, false },
+ { "uv", CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB, false },
+ { "al", CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK, true },
+ { "ual", CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK, true },
+ { "vl", CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK, true },
+ { "uvl", CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK, true },
+ /* webrtc data channel */
+ { "cd", CURRENT_STATUS_DATA_CHANNEL_CREATE, false },
+ { "dd", CURRENT_STATUS_DATA_CHANNEL_DESTROY, false },
+ { "dl", CURRENT_STATUS_DATA_CHANNEL_GET_LABEL, false },
+ { "zs", CURRENT_STATUS_DATA_CHANNEL_SEND_STRING, true },
+ { "zb", CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES, true },
+ { "zf", CURRENT_STATUS_DATA_CHANNEL_SEND_FILE, true },
+ { "sbc", CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB, true },
+ { "ubc", CURRENT_STATUS_DATA_CHANNEL_UNSET_BUFFERED_AMOUNT_LOW_CB, false },
+ { "gbt", CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT_LOW_THRESHOLD, false },
+ { "ba", CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT, false },
+ /* webrtc stats */
+ { "sts", CURRENT_STATUS_FOREACH_STATS, true },
+ /* app. setting & signaling */
+ { "ss", CURRENT_STATUS_SETTING_SIGNALING_SERVER, true },
+ { "cs", CURRENT_STATUS_CONNECT_SIGNALING_SERVER, false },
+ { "px", CURRENT_STATUS_SETTING_PROXY, true },
+ { "rs", CURRENT_STATUS_REQUEST_SESSION, true },
+ { "rj", CURRENT_STATUS_REQUEST_JOIN_ROOM, true },
+ { "sd", CURRENT_STATUS_SEND_LOCAL_DESCRIPTION, true },
+ { "ssc", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE, true },
+ { "ssd", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DESTROY, false },
+ { "sss", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_START, false },
+ { "sst", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_STOP, false },
+ { "scc", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT, true },
+ { "scd", CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DISCONNECT, false },
+ { NULL, -1, false },
+};
+
+void display_handle_status(int index)
+{
+ int ret = WEBRTC_ERROR_NONE;
+ webrtc_state_e state;
+ char *stun_server = NULL;
+
+ if (g_conns[index].webrtc == NULL)
+ return;
+
+ ret = webrtc_get_state(g_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_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+
+ ret = webrtc_foreach_turn_server(g_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(" state[%s]", g_webrtc_state_str[state]);
+ if (stun_server) {
+ g_print(" STUN server[%s]", stun_server);
+ free(stun_server);
+ }
+
+ g_print("\n-----------------------------------------------------------------------------------------\n");
+}
+
+void display_setting_status(void)
+{
+ int len_proxy = strlen(g_proxy);
+ int len_server = strlen(g_signaling_server.url);
+ int i;
+
+ if (len_proxy > 0)
+ g_print(" proxy[%s]", g_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);
+ for (i = 0; i < MAX_CONNECTION_LEN; i++) {
+ if (g_conns[i].remote_peer_id == 0)
+ continue;
+ g_print(" [%d] remote peer id : %d\n", i, g_conns[i].remote_peer_id);
+ }
+ g_print("-----------------------------------------------------------------------------------------\n");
+}
+
+void display_menu_main(void)
+{
+ g_print("\n");
+ g_print("=========================================================================================\n");
+ g_print(" Native WebRTC Test (press q to quit, * for internal API)\n");
+ g_print("-----------------------------------------------------------------------------------------\n");
+ display_handle_status(0);
+ g_print("c. Create\t");
+ g_print("d. Destroy\t");
+ g_print("s. Start\t");
+ g_print("t. Stop\t\t");
+ g_print("g. Get state\n");
+ g_print("sac. Set all callbacks\t");
+ g_print("uac. Unset all callbacks\n");
+ g_print("gan. Gets all the negotiation states\n");
+ g_print("st. Set STUN server\t");
+ g_print("gt. Get STUN server\t");
+ g_print("su. Add TURN server\t");
+ g_print("gu. Get TURN servers\n");
+ g_print("sbp. Set bundle policy\t");
+ g_print("gbp. Get bundle policy\n");
+ g_print("stp. Set ICE transport policy\t");
+ g_print("gtp. Get ICE transport policy\n");
+ g_print("co. Create offer\t");
+ g_print("ca. Create answer\n");
+ g_print("coa. Create offer(async)\t");
+ g_print("caa. Create answer(async)\n");
+ g_print("sl. Set local description\t");
+ g_print("sr. Set remote description\n");
+ g_print("ac. Add ICE candidate\n");
+ g_print("sdp. *Set RTP packet drop probability\t");
+ g_print("gdp. *Get RTP packet drop probability\n");
+ g_print("------------------------------------- Media Source --------------------------------------\n");
+ g_print("a. Add media source\t");
+ g_print("r. Remove media source\n");
+ g_print("p. Pause/play media source\t");
+ g_print("o. Get the media source pause\n");
+ g_print("mu. Mute/unmute media source\t");
+ g_print("mg. Get the media source mute\n");
+ g_print("v. Set video resolution\t");
+ g_print("l. Get video resolution\n");
+ g_print("f. Set video framerate\t");
+ g_print("m. Get video framerate\n");
+ g_print("td. Set transceiver direction\t");
+ g_print("gd. Get transceiver direction\n");
+ g_print("pa. Set media path to file source\n");
+ g_print("sfl. Set file source looping\t");
+ g_print("gfl. Set file source looping\n");
+ g_print("sf. Set media format to media packet source\n");
+ g_print("sp. Start pushing packet to media packet source\t");
+ g_print("tp. Stop pushing packet to media packet source\n");
+ g_print("scs. *Set crop screen source\t");
+ g_print("ucs. *Unset crop screen source\n");
+ g_print("------------------------------------- Media Render --------------------------------------\n");
+ g_print("dt. Set display type\t");
+ g_print("dm. Set display mode\t");
+ g_print("gm. Get display mode\n");
+ g_print("dv. Set display visible\t");
+ g_print("gv. Get display visible\n");
+ g_print("al. Set audio loopback\t");
+ g_print("ual. Unset audio loopback\n");
+ g_print("vl. Set video loopback\t");
+ g_print("uvl. Unset video loopback\n");
+ g_print("sa. Set encoded audio frame callback\t");
+ g_print("ua. Unset encoded audio frame callback\n");
+ g_print("sv. Set encoded video frame callback\t");
+ g_print("uv. Unset encoded video frame callback\n");
+ g_print("------------------------------------- Data Channel --------------------------------------\n");
+ g_print("cd. Create data channel\t");
+ g_print("dd. Destroy data channel\n");
+ g_print("dl. Get data channel label\n");
+ g_print("zs. Send string via data channel\n");
+ g_print("zb. Send string as bytes data via data channel\t");
+ g_print("zf. Send file via data channel\n");
+ g_print("ba. Get buffered amount\n");
+ g_print("sbc. Set buffered amount low callback\t");
+ g_print("ubc. Unset buffered amount low callback\n");
+ g_print("gbt. Get buffered amount low threshold\n");
+ g_print("---------------------------------------- Stats ------------------------------------------\n");
+ g_print("sts. Get stats\n");
+ g_print("------------------------------- App. Setting & Signaling --------------------------------\n");
+ display_setting_status();
+ g_print("px. Set proxy URL\n");
+ g_print("ss. Set signaling server URL\n");
+ g_print("cs. Connect to the signaling server\n");
+ g_print("rs. Request session of remote peer id\n");
+ g_print("rj. Request join room\n");
+ g_print("sd. Send local description\n");
+ g_print("ssc. *Create signaling server\t");
+ g_print("ssd. *Destroy signaling server\n");
+ g_print("sss. *Start signaling server\t");
+ g_print("sst. *Stop signaling server\n");
+ g_print("scc. *Connect to signaling server\t");
+ g_print("scd. *Disconnect from signaling server\n");
+ g_print("-----------------------------------------------------------------------------------------\n");
+ g_print("=========================================================================================\n");
+ g_print(" >>> ");
+}
+
+void display_menu_webrtc_common(void)
+{
+ switch (g_menu_status) {
+ case CURRENT_STATUS_SET_STUN_SERVER:
+ g_print("*** input STUN server address.\n");
+ break;
+ case CURRENT_STATUS_ADD_TURN_SERVER:
+ g_print("*** input TURN server address.\n");
+ break;
+ case CURRENT_STATUS_SET_BUNDLE_POLICY:
+ g_print("*** input bundle policy.(0:none, 1:max-bundle)\n");
+ break;
+ case CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY:
+ g_print("*** input ICE transport policy.(0:all, 1:relay)\n");
+ break;
+ case CURRENT_STATUS_SET_LOCAL_DESCRIPTION:
+ 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)
+ g_print("*** input side.(1:sender, 2:receiver)\n");
+ else if (g_cnt == 1)
+ g_print("*** input drop probability.(0 ~ 1.0)\n");
+ break;
+ case CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY:
+ if (g_cnt == 0)
+ g_print("*** input side.(1:sender, 2:receiver)\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void display_menu_webrtc_media_source(void)
+{
+ switch (g_menu_status) {
+ case CURRENT_STATUS_ADD_MEDIA_SOURCE:
+ 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");
+ 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)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ else if (g_cnt == 2)
+ g_print("*** input pause or play.(1:pause, 0:play)\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE :
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ else if (g_cnt == 2)
+ g_print("*** input mute mode.(1:mute 0:unmute)\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input width.\n");
+ else if (g_cnt == 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)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 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)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ else if (g_cnt == 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)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media type.(1:audio 2:video)\n");
+ break;
+ case CURRENT_STATUS_FILE_SOURCE_SET_PATH:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input media path.\n");
+ break;
+ case CURRENT_STATUS_FILE_SOURCE_SET_LOOPING:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input looping state.(1:true 0:false)\n");
+ break;
+ case CURRENT_STATUS_FILE_SOURCE_GET_LOOPING:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB:
+ g_print("*** input media packet source id to set buffer state changed callback.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB:
+ 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)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 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:
+ g_print("*** input media packet source id to start pushing packet.\n");
+ break;
+ case CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE:
+ g_print("*** input media packet source id to stop pushing packet.\n");
+ break;
+ case CURRENT_STATUS_SET_CROP_SCREEN_SOURCE:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input x.\n");
+ else if (g_cnt == 2)
+ g_print("*** input y.\n");
+ else if (g_cnt == 3)
+ g_print("*** input width.\n");
+ else if (g_cnt == 4)
+ g_print("*** input height.\n");
+ else if (g_cnt == 5)
+ g_print("*** input whether screen rotates (0: horizontal, 1: vertical).\n");
+ break;
+ case CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE:
+ if (g_cnt == 0)
+ g_print("*** input source id.\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void display_menu_webrtc_media_render(void)
+{
+ switch (g_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)
+ g_print("*** input track id.\n");
+ else if (g_cnt == 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)
+ g_print("*** input track id.\n");
+ else if (g_cnt == 1)
+ g_print("*** input display visible.(1:true 0:false)\n");
+ break;
+ case CURRENT_STATUS_GET_DISPLAY_VISIBLE:
+ g_print("*** input track id.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK:
+ g_print("*** input source id.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK:
+ g_print("*** input source id.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK:
+ g_print("*** input source id.\n");
+ break;
+ case CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK:
+ g_print("*** input source id.\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void display_menu_webrtc_data_channel(void)
+{
+ switch (g_menu_status) {
+ case CURRENT_STATUS_DATA_CHANNEL_SEND_STRING:
+ g_print("*** input string to send.\n");
+ break;
+ case CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES:
+ g_print("*** input string to send.(it will be converted to bytes data)\n");
+ break;
+ case CURRENT_STATUS_DATA_CHANNEL_SEND_FILE:
+ g_print("*** input file path to send.\n");
+ break;
+ case CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB:
+ g_print("*** input data channel buffered amount low threshold.\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void display_menu_webrtc_stats(void)
+{
+ switch (g_menu_status) {
+ case CURRENT_STATUS_FOREACH_STATS:
+ if (g_cnt == 0)
+ g_print("*** input stats type.(1:all, 2:codec, 3:inbound-rtp/remote-outbound-rtp, 4:outbound-rtp/remote-inbound-rtp)\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void display_menu_app_signaling(void)
+{
+ switch (g_menu_status) {
+ case CURRENT_STATUS_SETTING_SIGNALING_SERVER:
+ g_print("*** input signaling server URL.\n");
+ break;
+ case CURRENT_STATUS_SETTING_PROXY:
+ g_print("*** input proxy URL.\n");
+ break;
+ case CURRENT_STATUS_REQUEST_SESSION:
+ g_print("*** input remote peer id.\n");
+ break;
+ case CURRENT_STATUS_REQUEST_JOIN_ROOM:
+ if (g_cnt == 0)
+ g_print("*** input source type.(1:audiotest/videotest 2:mic/camera 3:mic only)\n");
+ else if (g_cnt == 1)
+ g_print("*** input room name to join.\n");
+ break;
+ case CURRENT_STATUS_SEND_LOCAL_DESCRIPTION:
+ g_print("*** input type of local description to send to the server.(1:offer, 2:answer)\n");
+ break;
+ case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE:
+ g_print("*** input port.\n");
+ break;
+ case CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT:
+ if (g_cnt == 0)
+ g_print("*** input server ip.\n");
+ else if (g_cnt == 1)
+ g_print("*** input port.\n");
+ break;
+ }
+ g_print(" >>> ");
+}
+
+void displaymenu(void)
+{
+ if (g_menu_status == CURRENT_STATUS_MAINMENU) {
+ display_menu_main();
+
+ } else {
+ if (g_menu_status & TEST_MENU_WEBRTC_COMMON) {
+ display_menu_webrtc_common();
+
+ } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_SOURCE) {
+ display_menu_webrtc_media_source();
+
+ } else if (g_menu_status & TEST_MENU_WEBRTC_MEDIA_RENDER) {
+ display_menu_webrtc_media_render();
+
+ } else if (g_menu_status & TEST_MENU_WEBRTC_DATA_CHANNEL) {
+ display_menu_webrtc_data_channel();
+
+ } else if (g_menu_status & TEST_MENU_WEBRTC_STATS) {
+ display_menu_webrtc_stats();
+
+ } else if (g_menu_status & TEST_MENU_APP_SIGNALING) {
+ display_menu_app_signaling();
+
+ } else {
+ g_print("%s() > unknown menu status[0x%x]\n", __FUNCTION__, g_menu_status);
+ quit_program();
+ }
+ }
+}
+
+void reset_menu_state(void)
+{
+ g_menu_status = CURRENT_STATUS_MAINMENU;
+}
\ No newline at end of file
--- /dev/null
+/*
+ * Copyright (c) 2022 Samsung Electronics Co., Ltd All Rights Reserved
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __TIZEN_MEDIA_WEBRTC_TEST_PRIVATE_H__
+#define __TIZEN_MEDIA_WEBRTC_TEST_PRIVATE_H__
+
+#include <webrtc_internal.h>
+#include <media_format.h>
+#include <media_packet_internal.h>
+#include <sound_manager.h>
+#ifndef TIZEN_TV
+#include <esplusplayer_capi.h>
+#endif
+#include <appcore-efl.h>
+#include <Elementary.h>
+#include <glib.h>
+#include <gst/gst.h>
+#include <libsoup/soup.h>
+
+//#define __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__
+//#define __DEBUG_VALIDATE_ENCODED_FRAME_CB__
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define RET_IF(expr, fmt, arg...) \
+do { \
+ if ((expr)) { \
+ g_printerr("failed to %s(), "fmt"\n", __func__, ##arg); \
+ return; \
+ } \
+} while (0)
+
+#define TEST_MENU_WEBRTC_COMMON 0x00001000
+#define TEST_MENU_WEBRTC_MEDIA_SOURCE 0x00002000
+#define TEST_MENU_WEBRTC_MEDIA_RENDER 0x00004000
+#define TEST_MENU_WEBRTC_DATA_CHANNEL 0x00008000
+#define TEST_MENU_WEBRTC_STATS 0x00010000
+#define TEST_MENU_APP_SIGNALING 0x00020000
+
+enum {
+ CURRENT_STATUS_MAINMENU,
+ CURRENT_STATUS_TERMINATE,
+ CURRENT_STATUS_QUIT,
+ /* webrtc common */
+ CURRENT_STATUS_CREATE = TEST_MENU_WEBRTC_COMMON | 0x01,
+ CURRENT_STATUS_START = TEST_MENU_WEBRTC_COMMON | 0x02,
+ CURRENT_STATUS_STOP = TEST_MENU_WEBRTC_COMMON | 0x03,
+ CURRENT_STATUS_DESTROY = TEST_MENU_WEBRTC_COMMON | 0x04,
+ CURRENT_STATUS_GET_STATE = TEST_MENU_WEBRTC_COMMON | 0x05,
+ CURRENT_STATUS_SET_STUN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x06,
+ CURRENT_STATUS_GET_STUN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x07,
+ CURRENT_STATUS_ADD_TURN_SERVER = TEST_MENU_WEBRTC_COMMON | 0x08,
+ CURRENT_STATUS_GET_TURN_SERVERS = TEST_MENU_WEBRTC_COMMON | 0x09,
+ CURRENT_STATUS_SET_BUNDLE_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0A,
+ CURRENT_STATUS_GET_BUNDLE_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0B,
+ CURRENT_STATUS_SET_ICE_TRANSPORT_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0C,
+ CURRENT_STATUS_GET_ICE_TRANSPORT_POLICY = TEST_MENU_WEBRTC_COMMON | 0x0D,
+ CURRENT_STATUS_CREATE_OFFER = TEST_MENU_WEBRTC_COMMON | 0x0E,
+ CURRENT_STATUS_CREATE_ANSWER = TEST_MENU_WEBRTC_COMMON | 0x0F,
+ CURRENT_STATUS_CREATE_OFFER_ASYNC = TEST_MENU_WEBRTC_COMMON | 0x10,
+ CURRENT_STATUS_CREATE_ANSWER_ASYNC = TEST_MENU_WEBRTC_COMMON | 0x11,
+ CURRENT_STATUS_SET_LOCAL_DESCRIPTION = TEST_MENU_WEBRTC_COMMON | 0x12,
+ CURRENT_STATUS_SET_REMOTE_DESCRIPTION = TEST_MENU_WEBRTC_COMMON | 0x13,
+ CURRENT_STATUS_ADD_ICE_CANDIDATE = TEST_MENU_WEBRTC_COMMON | 0x14,
+ CURRENT_STATUS_SET_ALL_CALLBACKS = TEST_MENU_WEBRTC_COMMON | 0x15,
+ CURRENT_STATUS_UNSET_ALL_CALLBACKS = TEST_MENU_WEBRTC_COMMON | 0x16,
+ CURRENT_STATUS_GET_ALL_NEGOTIATION_STATES = TEST_MENU_WEBRTC_COMMON | 0x17,
+ CURRENT_STATUS_SET_RTP_PACKET_DROP_PROBABILITY = TEST_MENU_WEBRTC_COMMON | 0x18,
+ CURRENT_STATUS_GET_RTP_PACKET_DROP_PROBABILITY = TEST_MENU_WEBRTC_COMMON | 0x19,
+ /* webrtc media source */
+ CURRENT_STATUS_ADD_MEDIA_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x01,
+ CURRENT_STATUS_REMOVE_MEDIA_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x02,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_TRANSCEIVER_DIRECTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x03,
+ CURRENT_STATUS_MEDIA_SOURCE_GET_TRANSCEIVER_DIRECTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x04,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_PAUSE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x05,
+ CURRENT_STATUS_MEDIA_SOURCE_GET_PAUSE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x06,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_MUTE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x07,
+ CURRENT_STATUS_MEDIA_SOURCE_GET_MUTE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x08,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_RESOLUTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x09,
+ CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_RESOLUTION = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0A,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_FRAMERATE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0B,
+ CURRENT_STATUS_MEDIA_SOURCE_GET_VIDEO_FRAMERATE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0C,
+ CURRENT_STATUS_FILE_SOURCE_SET_PATH = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0D,
+ CURRENT_STATUS_FILE_SOURCE_SET_LOOPING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0E,
+ CURRENT_STATUS_FILE_SOURCE_GET_LOOPING = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x0F,
+ CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_FORMAT = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x10,
+ CURRENT_STATUS_MEDIA_PACKET_SOURCE_SET_BUFFER_STATE_CHANGED_CB = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x11,
+ CURRENT_STATUS_MEDIA_PACKET_SOURCE_UNSET_BUFFER_STATE_CHANGED_CB = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x12,
+ CURRENT_STATUS_START_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x13,
+ CURRENT_STATUS_STOP_PUSHING_PACKET_TO_MEDIA_PACKET_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x14,
+ CURRENT_STATUS_SET_CROP_SCREEN_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x15,
+ CURRENT_STATUS_UNSET_CROP_SCREEN_SOURCE = TEST_MENU_WEBRTC_MEDIA_SOURCE | 0x16,
+ /* webrtc media render */
+ CURRENT_STATUS_SET_DISPLAY_TYPE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x01,
+ CURRENT_STATUS_SET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x02,
+ CURRENT_STATUS_GET_DISPLAY_MODE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x03,
+ CURRENT_STATUS_SET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x04,
+ CURRENT_STATUS_GET_DISPLAY_VISIBLE = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x05,
+ CURRENT_STATUS_SET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x06,
+ CURRENT_STATUS_UNSET_ENCODED_AUDIO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x07,
+ CURRENT_STATUS_SET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x08,
+ CURRENT_STATUS_UNSET_ENCODED_VIDEO_FRAME_CB = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x09,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0A,
+ CURRENT_STATUS_MEDIA_SOURCE_UNSET_AUDIO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0B,
+ CURRENT_STATUS_MEDIA_SOURCE_SET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0C,
+ CURRENT_STATUS_MEDIA_SOURCE_UNSET_VIDEO_LOOPBACK = TEST_MENU_WEBRTC_MEDIA_RENDER | 0x0D,
+ /* webrtc data channel */
+ CURRENT_STATUS_DATA_CHANNEL_CREATE = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x01,
+ CURRENT_STATUS_DATA_CHANNEL_DESTROY = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x02,
+ CURRENT_STATUS_DATA_CHANNEL_GET_LABEL = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x03,
+ CURRENT_STATUS_DATA_CHANNEL_SEND_STRING = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x04,
+ CURRENT_STATUS_DATA_CHANNEL_SEND_STRING_AS_BYTES = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x05,
+ CURRENT_STATUS_DATA_CHANNEL_SEND_FILE = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x06,
+ CURRENT_STATUS_DATA_CHANNEL_SET_BUFFERED_AMOUNT_LOW_CB = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x07,
+ CURRENT_STATUS_DATA_CHANNEL_UNSET_BUFFERED_AMOUNT_LOW_CB = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x08,
+ CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT_LOW_THRESHOLD = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x09,
+ CURRENT_STATUS_DATA_CHANNEL_GET_BUFFERED_AMOUNT = TEST_MENU_WEBRTC_DATA_CHANNEL | 0x0A,
+ /* webrtc stats */
+ CURRENT_STATUS_FOREACH_STATS = TEST_MENU_WEBRTC_STATS | 0x01,
+ /* app. setting & signaling */
+ CURRENT_STATUS_SETTING_SIGNALING_SERVER = TEST_MENU_APP_SIGNALING | 0x01,
+ CURRENT_STATUS_CONNECT_SIGNALING_SERVER = TEST_MENU_APP_SIGNALING | 0x02,
+ CURRENT_STATUS_SETTING_PROXY = TEST_MENU_APP_SIGNALING | 0x03,
+ CURRENT_STATUS_REQUEST_SESSION = TEST_MENU_APP_SIGNALING | 0x04,
+ CURRENT_STATUS_REQUEST_JOIN_ROOM = TEST_MENU_APP_SIGNALING | 0x05,
+ CURRENT_STATUS_SEND_LOCAL_DESCRIPTION = TEST_MENU_APP_SIGNALING | 0x06,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CREATE = TEST_MENU_APP_SIGNALING | 0x07,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DESTROY = TEST_MENU_APP_SIGNALING | 0x08,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_START = TEST_MENU_APP_SIGNALING | 0x09,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_STOP = TEST_MENU_APP_SIGNALING | 0x0A,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_CONNECT = TEST_MENU_APP_SIGNALING | 0x0B,
+ CURRENT_STATUS_PRIVATE_SIGNALING_SERVER_DISCONNECT = TEST_MENU_APP_SIGNALING | 0x0C,
+};
+
+enum {
+ SERVER_STATUS_DISCONNECTED,
+ SERVER_STATUS_CONNECTED,
+ SERVER_STATUS_SESSION_ESTABLISHED,
+ SERVER_STATUS_SESSION_CLOSED,
+ SERVER_STATUS_ROOM_ESTABLISHED,
+ SERVER_STATUS_ERROR_FOUND
+};
+
+#define MAX_STRING_LEN 512
+#define MAX_CONNECTION_LEN 3
+#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;
+
+ webrtc_h webrtc;
+ GstElement *src_pipeline;
+#ifdef __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__
+ GstElement *render_pipeline;
+ GstElement *appsrc;
+#endif
+ GstElement *src;
+ GstElement *sink;
+ GstElement *demux;
+ GstBus *bus;
+ guint bus_watcher;
+ gulong handoff_signal_id;
+ gulong pad_added_signal_id;
+ bool is_overflowed;
+ bool is_stop_requested;
+ GCond cond;
+ GMutex mutex;
+ bool got_eos;
+} media_packet_source_s;
+
+typedef struct _connection_s {
+ int index;
+ int remote_peer_id;
+
+ bool is_for_room;
+ bool is_offer;
+ int room_source_type;
+
+ webrtc_h webrtc;
+ webrtc_data_channel_h channels[MAX_CHANNEL_LEN];
+ int channel_index;
+ webrtc_data_channel_h recv_channels[MAX_CHANNEL_LEN];
+ char *offer;
+ char *answer;
+ char *remote_desc;
+ GList *ice_candidates;
+
+ /* receive data & dump file */
+ gint64 sum_size;
+ gchar *expected_name;
+ gint64 expected_size;
+ char* receive_buffer;
+
+ struct {
+ sound_stream_info_h stream_info;
+ } source;
+
+ struct {
+ sound_stream_info_h stream_info;
+ webrtc_display_type_e display_type;
+ Evas_Object *eo;
+ Evas_Object *text_eo;
+ unsigned int loopback_track_id;
+#ifndef TIZEN_TV
+ esplusplayer_handle espp;
+#endif
+ } render;
+
+#ifndef TIZEN_TV
+ bool encoded_video_frame_cb_is_set;
+ bool encoded_audio_frame_cb_is_set;
+#endif
+#ifdef __DEBUG_VALIDATE_ENCODED_FRAME_CB__
+ GstElement *audio_render_pipeline;
+ GstElement *video_render_pipeline;
+ GstElement *appsrc_for_audio;
+ GstElement *appsrc_for_video;
+#endif
+ media_packet_source_s packet_sources[MAX_MEDIA_PACKET_SOURCE_LEN];
+} connection_s;
+
+typedef struct _signaling_server_s {
+ gchar url[MAX_STRING_LEN];
+ SoupWebsocketConnection *ws_conn;
+ int server_status;
+ gint32 local_peer_id;
+
+ /* for private network - internal API */
+ webrtc_signaling_client_h signaling_client;
+ char *private_ip;
+ int port;
+ bool is_connected;
+} signaling_server_s;
+
+typedef struct {
+ const char *cmd;
+ int status;
+ bool key_input_needed;
+} menu_info_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;
+
+void display_handle_status(int index);
+void display_setting_status(void);
+void display_menu_main(void);
+void display_menu_webrtc_common(void);
+void display_menu_webrtc_media_source(void);
+void display_menu_webrtc_media_render(void);
+void display_menu_webrtc_data_channel(void);
+void display_menu_webrtc_stats(void);
+void display_menu_app_signaling(void);
+void displaymenu(void);
+void reset_menu_state(void);
+void quit_program(void);
+bool foreach_turn_server(const char *turn_server, gpointer user_data);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif
\ No newline at end of file
projects / platform / core / api / webrtc.git / commitdiff