WEBRTC_STATE_PLAYING, /**< Started all streams */
} webrtc_state_e;
+/**
+ * @brief Enumeration for WebRTC peer connection state.
+ * @since_tizen 6.5
+ */
+typedef enum {
+ WEBRTC_PEER_CONNECTION_STATE_NEW, /**< New */
+ WEBRTC_PEER_CONNECTION_STATE_CONNECTING, /**< Connecting */
+ WEBRTC_PEER_CONNECTION_STATE_CONNECTED, /**< Connected */
+ WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED, /**< Disconnected */
+ WEBRTC_PEER_CONNECTION_STATE_FAILED, /**< Failed */
+ WEBRTC_PEER_CONNECTION_STATE_CLOSED, /**< Closed */
+} webrtc_peer_connection_state_e;
+
/**
* @brief Enumeration for WebRTC media source type.
* @since_tizen 6.5
*/
typedef void (*webrtc_state_changed_cb)(webrtc_h webrtc, webrtc_state_e previous, webrtc_state_e current, void *user_data);
+/**
+ * @brief Called when the WebRTC peer connection state is changed.
+ * @since_tizen 6.5
+ * @remarks The @a webrtc is the same object for which the callback was set.\nThe @a webrtc should not be released.
+ * @param[in] webrtc WebRTC handle
+ * @param[in] state The peer connection state
+ * @param[in] user_data The user data passed from the callback registration function
+ * @see webrtc_set_peer_connection_state_change_cb()
+ * @see webrtc_unset_peer_connection_state_change_cb()
+ */
+typedef void (*webrtc_peer_connection_state_change_cb)(webrtc_h webrtc, webrtc_peer_connection_state_e state, void *user_data);
+
/**
* @brief Called when the buffer state of media packet source is changed.
* @since_tizen 6.5
*/
int webrtc_foreach_turn_server(webrtc_h webrtc, webrtc_turn_server_cb callback, void *user_data);
+/**
+ * @brief Sets a callback function to be invoked when the WebRTC peer connection state is changed.
+ * @since_tizen 6.5
+ * @param[in] webrtc WebRTC handle
+ * @param[in] callback Callback function pointer
+ * @param[in] user_data The user data to be passed to the callback function
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @post webrtc_peer_connection_state_change_cb() will be invoked.
+ * @see webrtc_unset_peer_connection_state_change_cb()
+ * @see webrtc_peer_connection_state_change_cb()
+ * @see webrtc_get_peer_connection_state()
+ */
+int webrtc_set_peer_connection_state_change_cb(webrtc_h webrtc, webrtc_peer_connection_state_change_cb callback, void *user_data);
+
+/**
+ * @brief Unsets the peer connection state change callback function.
+ * @since_tizen 6.5
+ * @param[in] webrtc WebRTC handle
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @see webrtc_set_peer_connection_state_change_cb()
+ * @see webrtc_get_peer_connection_state()
+ */
+int webrtc_unset_peer_connection_state_change_cb(webrtc_h webrtc);
+
+/**
+ * @brief Gets the peer connection state.
+ * @since_tizen 6.5
+ * @remarks The default value is #WEBRTC_PEER_CONNECTION_STATE_NEW.
+ * @param[in] webrtc WebRTC handle
+ * @param[out] state The peer connection state
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @see webrtc_set_peer_connection_state_change_cb()
+ * @see webrtc_unset_peer_connection_state_change_cb()
+ */
+int webrtc_get_peer_connection_state(webrtc_h webrtc, webrtc_peer_connection_state_e *state);
+
/**
* @brief Sets a negotiation needed callback function to be invoked when a change has occurred which requires session negotiation.
* @since_tizen 6.5
webrtc_callbacks_s error_cb;
webrtc_callbacks_s state_changed_cb;
+ webrtc_callbacks_s peer_connection_state_change_cb;
webrtc_callbacks_s negotiation_needed_cb;
webrtc_callbacks_s ice_candidate_cb;
webrtc_callbacks_s track_added_cb;
webrtc_callbacks_s data_channel_cb;
struct {
+ webrtc_peer_connection_state_e peer_connection_state;
const gchar *signaling_state;
- const gchar *peer_connection_state;
const gchar *ice_connection_state;
const gchar *ice_gathering_state;
} internal_states;
Name: capi-media-webrtc
Summary: A WebRTC library in Tizen Native API
-Version: 0.1.160
+Version: 0.1.161
Release: 0
Group: Multimedia/API
License: Apache-2.0
return WEBRTC_ERROR_NONE;
}
+int webrtc_set_peer_connection_state_change_cb(webrtc_h webrtc, webrtc_peer_connection_state_change_cb callback, void *user_data)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(callback == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "callback is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+
+ LOG_WARNING_IF_CALLBACK_EXISTS(_webrtc->peer_connection_state_change_cb);
+
+ _webrtc->peer_connection_state_change_cb.callback = callback;
+ _webrtc->peer_connection_state_change_cb.user_data = user_data;
+
+ LOG_INFO("callback[%p] user_data[%p]", callback, user_data);
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_unset_peer_connection_state_change_cb(webrtc_h webrtc)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->state != WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, &_webrtc->mutex, "the state should be IDLE");
+ RET_VAL_WITH_UNLOCK_IF(_webrtc->peer_connection_state_change_cb.callback == NULL, WEBRTC_ERROR_INVALID_OPERATION, &_webrtc->mutex, "callback was not set");
+
+ LOG_INFO("callback[%p] user_data[%p] is reset to NULL",
+ _webrtc->peer_connection_state_change_cb.callback, _webrtc->peer_connection_state_change_cb.user_data);
+
+ _webrtc->peer_connection_state_change_cb.callback = NULL;
+ _webrtc->peer_connection_state_change_cb.user_data = NULL;
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ return WEBRTC_ERROR_NONE;
+}
+
+int webrtc_get_peer_connection_state(webrtc_h webrtc, webrtc_peer_connection_state_e *state)
+{
+ webrtc_s *_webrtc = (webrtc_s*)webrtc;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(state == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "state is NULL");
+
+ g_mutex_lock(&_webrtc->mutex);
+
+ *state = _webrtc->internal_states.peer_connection_state;
+
+ LOG_INFO("state[%d]", *state);
+
+ g_mutex_unlock(&_webrtc->mutex);
+
+ return WEBRTC_ERROR_NONE;
+}
+
int webrtc_set_negotiation_needed_cb(webrtc_h webrtc, webrtc_negotiation_needed_cb callback, void *user_data)
{
webrtc_s *_webrtc = (webrtc_s*)webrtc;
GstWebRTCRTPTransceiverDirection gst;
} direction_info_s;
+typedef struct {
+ const char *str;
+ webrtc_peer_connection_state_e state;
+} peer_connection_state_info_s;
+
static direction_info_s __direction_info[] = {
[WEBRTC_TRANSCEIVER_DIRECTION_SENDONLY] = { "SENDONLY", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY },
[WEBRTC_TRANSCEIVER_DIRECTION_RECVONLY] = { "RECVONLY", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY },
[WEBRTC_TRANSCEIVER_DIRECTION_SENDRECV] = { "SENDRECV", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV }
};
+static peer_connection_state_info_s __peer_connection_state_info[] = {
+ [GST_WEBRTC_PEER_CONNECTION_STATE_NEW] = { "NEW", WEBRTC_PEER_CONNECTION_STATE_NEW },
+ [GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING] = { "CONNECTING", WEBRTC_PEER_CONNECTION_STATE_CONNECTING },
+ [GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED] = { "CONNECTED", WEBRTC_PEER_CONNECTION_STATE_CONNECTED },
+ [GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED] = { "DISCONNECTED", WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED },
+ [GST_WEBRTC_PEER_CONNECTION_STATE_FAILED] = { "FAILED", WEBRTC_PEER_CONNECTION_STATE_FAILED },
+ [GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED] = { "CLOSED", WEBRTC_PEER_CONNECTION_STATE_CLOSED },
+};
+
static const char *__get_error_string(webrtc_error_e error)
{
switch (error) {
}
}
+static void __invoke_peer_connection_state_change_cb(webrtc_s *webrtc, GstWebRTCPeerConnectionState state)
+{
+ webrtc_callbacks_s *cb;
+
+ RET_IF(webrtc == NULL, "webrtc is NULL");
+
+ cb = &webrtc->peer_connection_state_change_cb;
+
+ if (cb->callback) {
+ LOG_DEBUG(">>> callback[%p], user_data[%p]", cb->callback, cb->user_data);
+ ((webrtc_peer_connection_state_change_cb)(cb->callback))((webrtc_h)webrtc, __peer_connection_state_info[state].state, cb->user_data);
+ LOG_DEBUG("<<< end of the callback");
+ }
+
+ GENERATE_DOT(webrtc, "PEER_CONNECTION_STATE_%s", __peer_connection_state_info[state].str);
+}
+
static gboolean __bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_data)
{
webrtc_s *webrtc = (webrtc_s *)user_data;
static void __webrtcbin_peer_connection_state_cb(GstElement *webrtcbin, GParamSpec * pspec, gpointer user_data)
{
webrtc_s *webrtc = (webrtc_s *)user_data;
- GstWebRTCPeerConnectionState peer_connection_state;
- const gchar *new_state = "UNKNOWN";
+ GstWebRTCPeerConnectionState state;
RET_IF(webrtcbin == NULL, "webrtcbin is NULL");
RET_IF(webrtc == NULL, "webrtc is NULL");
- g_object_get(webrtcbin, "connection-state", &peer_connection_state, NULL);
+ g_object_get(webrtcbin, "connection-state", &state, NULL);
- switch (peer_connection_state) {
- case GST_WEBRTC_PEER_CONNECTION_STATE_NEW:
- new_state = "NEW";
+ LOG_DEBUG("[PeerConnectionState] is changed to [%s]", __peer_connection_state_info[state].str);
+ webrtc->internal_states.peer_connection_state = __peer_connection_state_info[state].state;
+
+ __invoke_peer_connection_state_change_cb(webrtc, state);
+ switch (state) {
+ case GST_WEBRTC_PEER_CONNECTION_STATE_NEW:
g_mutex_lock(&webrtc->mutex);
if (webrtc->state == WEBRTC_STATE_NEGOTIATING)
_post_state_cb_in_idle(webrtc, WEBRTC_STATE_PLAYING);
g_mutex_unlock(&webrtc->mutex);
break;
- case GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING:
- new_state = "CONNECTING";
- break;
-
- case GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED:
- new_state = "CONNECTED";
- break;
-
- /* FIXME: error forwarding in case of disconnection/failed via error callback */
- case GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED:
- new_state = "DISCONNECTED";
- break;
-
case GST_WEBRTC_PEER_CONNECTION_STATE_FAILED:
- new_state = "FAILED";
-
__invoke_error_cb(webrtc, WEBRTC_ERROR_CONNECTION_FAILED);
break;
- case GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED:
- new_state = "CLOSED";
+ default:
break;
}
-
- LOG_DEBUG("[PeerConnectionState] is changed to [%s]", new_state);
-
- webrtc->internal_states.peer_connection_state = new_state;
}
static void __webrtcbin_signaling_state_cb(GstElement *webrtcbin, GParamSpec * pspec, gpointer user_data)
projects / platform / core / api / webrtc.git / commitdiff