Check if handle is destroying before posting idle-callbacks

Sometimes, deadlock happens in C# TCT because of a sequence below.
 - webrtc handle seems to be destroyed before negotiation
  callbacks have not been processed in the main thread posted
  by g_idle_add_full().

It is checked that state and error callbacks have no chance
to get this situation.

[Version] 0.3.263
[Issue Type] Improvement

Change-Id: Ia7359bb45aaa1095dd77ee673c5b3b72056da62b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 082b75802071ad3fcecbf9e3fcf6a414ba31a652..115c4a76029fe45f215ebe60b193ee2677dfd587 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -482,6 +482,7 @@ typedef struct _webrtc_s {
        GCond desc_cond;
        GMutex stats_mutex;
        GCond stats_cond;
+       GMutex destroy_mutex;
 
        webrtc_gst_s gst;
        unsigned int payload_types;
@@ -499,6 +500,7 @@ typedef struct _webrtc_s {
 
        webrtc_state_e state;
        webrtc_state_e pend_state;
+       bool is_destroying;
 
        GList *signals;
 

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index f59c3028ee10e8f7e12eda3336e8985d77acb4e4..9ada9ba33b18eb5df07906a9daea7e2592cec7f2 100644 (file)
--- 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:    0.3.262
+Version:    0.3.263
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc.c b/src/webrtc.c
index e5b50cf498b688c963b65a9fa10181f902564a18..e971627aa578f10d2ed8164626090ed24f24dfd4 100644 (file)
--- a/src/webrtc.c
+++ b/src/webrtc.c
@@ -98,6 +98,7 @@ int webrtc_create(webrtc_h *webrtc)
        g_cond_init(&_webrtc->desc_cond);
        g_mutex_init(&_webrtc->stats_mutex);
        g_cond_init(&_webrtc->stats_cond);
+       g_mutex_init(&_webrtc->destroy_mutex);
 
        _load_ini(_webrtc);
        if (_is_resource_required(&_webrtc->ini)) {
@@ -137,16 +138,19 @@ int webrtc_destroy(webrtc_h webrtc)
 {
        int ret = WEBRTC_ERROR_NONE;
        g_autoptr(GMutexLocker) locker = NULL;
+       g_autoptr(GMutexLocker) d_locker = NULL;
        webrtc_s *_webrtc = (webrtc_s *)webrtc;
 
        RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 
        locker = g_mutex_locker_new(&_webrtc->mutex);
+       d_locker = g_mutex_locker_new(&_webrtc->destroy_mutex);
 
-       _remove_remained_event_sources(_webrtc);
-
+       _webrtc->is_destroying = true;
        _webrtc->pend_state = WEBRTC_STATE_IDLE;
 
+       _remove_remained_event_sources(_webrtc);
+
        _unset_stats_timer(_webrtc);
 
        ret = _gst_pipeline_set_state(_webrtc, GST_STATE_NULL);
@@ -159,7 +163,10 @@ int webrtc_destroy(webrtc_h webrtc)
        _webrtc->state = _webrtc->pend_state;
 
        _destroy_data_channels(_webrtc);
+
+       g_clear_pointer(&d_locker, g_mutex_locker_free);
        _gst_destroy_pipeline(_webrtc);
+
        _unload_ini(_webrtc);
 
 #if !defined(TIZEN_TV) && defined(TIZEN_FEATURE_UI)
@@ -175,6 +182,7 @@ int webrtc_destroy(webrtc_h webrtc)
 
        LOG_INFO("webrtc[%p] is destroyed", webrtc);
 
+       g_mutex_clear(&_webrtc->destroy_mutex);
        g_clear_pointer(&locker, g_mutex_locker_free);
        g_mutex_clear(&_webrtc->mutex);
 

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 87bf64df521a3880be00fe8f92414e62b035db85..ac88c2aa0e8d9736208566b775292b89521fedc4 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -773,6 +773,7 @@ static void __post_peer_connection_state_change_cb_in_idle(webrtc_s *webrtc, web
        g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
+       RET_IF(webrtc->is_destroying, "webrtc is destroying, skip it");
 
        data = g_new0(idle_userdata_s, 1);
        data->webrtc = webrtc;
@@ -794,6 +795,7 @@ static void __post_signaling_state_change_cb_in_idle(webrtc_s *webrtc, webrtc_si
        g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
+       RET_IF(webrtc->is_destroying, "webrtc is destroying, skip it");
 
        data = g_new0(idle_userdata_s, 1);
        data->webrtc = webrtc;
@@ -814,6 +816,7 @@ static void __post_ice_gathering_state_change_cb_in_idle(webrtc_s *webrtc, webrt
        g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
+       RET_IF(webrtc->is_destroying, "webrtc is destroying, skip it");
 
        data = g_new0(idle_userdata_s, 1);
        data->webrtc = webrtc;
@@ -835,6 +838,7 @@ static void __post_ice_connection_state_change_cb_in_idle(webrtc_s *webrtc, webr
        g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
+       RET_IF(webrtc->is_destroying, "webrtc is destroying, skip it");
 
        data = g_new0(idle_userdata_s, 1);
        data->webrtc = webrtc;
@@ -1148,6 +1152,7 @@ static void __webrtcbin_peer_connection_state_cb(GstElement *webrtcbin, GParamSp
 {
        webrtc_s *webrtc = (webrtc_s *)user_data;
        GstWebRTCPeerConnectionState state;
+       g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtcbin == NULL, "webrtcbin is NULL");
        RET_IF(webrtc == NULL, "webrtc is NULL");
@@ -1156,17 +1161,18 @@ static void __webrtcbin_peer_connection_state_cb(GstElement *webrtcbin, GParamSp
 
        LOG_DEBUG("webrtc[%p] [PeerConnectionState] is changed to [%s]", webrtc, __peer_connection_state_info[state].str);
 
+       locker = g_mutex_locker_new(&webrtc->destroy_mutex);
+
        __post_peer_connection_state_change_cb_in_idle(webrtc, __peer_connection_state_info[state].state);
 
        switch (state) {
        case GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED:
-               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_FAILED:
+               g_clear_pointer(&locker, g_mutex_locker_free);
                __invoke_error_cb(webrtc, WEBRTC_ERROR_CONNECTION_FAILED);
                break;
 
@@ -1180,6 +1186,7 @@ static void __webrtcbin_signaling_state_cb(GstElement *webrtcbin, GParamSpec * p
 {
        webrtc_s *webrtc = (webrtc_s *)user_data;
        GstWebRTCSignalingState state;
+       g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtcbin == NULL, "webrtcbin is NULL");
        RET_IF(webrtc == NULL, "webrtc is NULL");
@@ -1192,6 +1199,8 @@ static void __webrtcbin_signaling_state_cb(GstElement *webrtcbin, GParamSpec * p
                state == GST_WEBRTC_SIGNALING_STATE_STABLE)
                __update_data_recovery_type_from_remote_description(webrtc);
 
+       locker = g_mutex_locker_new(&webrtc->destroy_mutex);
+
        __post_signaling_state_change_cb_in_idle(webrtc, __signaling_state_info[state].state);
 }
 
@@ -1199,6 +1208,7 @@ static void __webrtcbin_ice_gathering_state_cb(GstElement *webrtcbin, GParamSpec
 {
        webrtc_s *webrtc = (webrtc_s *)user_data;
        GstWebRTCICEGatheringState state;
+       g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtcbin == NULL, "webrtcbin is NULL");
        RET_IF(webrtc == NULL, "webrtc is NULL");
@@ -1207,6 +1217,8 @@ static void __webrtcbin_ice_gathering_state_cb(GstElement *webrtcbin, GParamSpec
 
        LOG_DEBUG("webrtc[%p] [IceGatheringState] is changed to [%s]", webrtc, __ice_gathering_state_info[state].str);
 
+       locker = g_mutex_locker_new(&webrtc->destroy_mutex);
+
        __post_ice_gathering_state_change_cb_in_idle(webrtc, __ice_gathering_state_info[state].state);
 }
 
@@ -1215,6 +1227,7 @@ static void __webrtcbin_ice_connection_state_cb(GstElement *webrtcbin, GParamSpe
 {
        webrtc_s *webrtc = (webrtc_s *)user_data;
        GstWebRTCICEConnectionState state;
+       g_autoptr(GMutexLocker) locker = NULL;
 
        RET_IF(webrtcbin == NULL, "webrtcbin is NULL");
        RET_IF(webrtc == NULL, "webrtc is NULL");
@@ -1223,6 +1236,8 @@ static void __webrtcbin_ice_connection_state_cb(GstElement *webrtcbin, GParamSpe
 
        LOG_DEBUG("webrtc[%p] [IceConnectionState] is changed to [%s]", webrtc, __ice_connection_state_info[state].str);
 
+       locker = g_mutex_locker_new(&webrtc->destroy_mutex);
+
        __post_ice_connection_state_change_cb_in_idle(webrtc, __ice_connection_state_info[state].state);
 
        if (state == GST_WEBRTC_ICE_CONNECTION_STATE_FAILED)