webrtc_private/sink: Print handle pointer address

Some logs for webrtc handle and decodebin are added.

[Version] 0.2.122
[Issue Type] Log

Change-Id: Ib07f9216c482d3e73a38cf51069e8cabc0669c94
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 4e25790bc527c41fffa1ae629ca98b76ce232697..0d139a200ff5aa32793001876fdfed013b0b2f94 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:    0.2.121
+Version:    0.2.122
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index bd06ae86696fbdcf875d80976ddb8a826b5e3112..2fd65ce2e92d08d6b0c4935982c9a05b0a988545 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -221,10 +221,10 @@ void _invoke_state_changed_cb(webrtc_s *webrtc, webrtc_state_e old, webrtc_state
        if (old == new)
                return;
 
-       LOG_INFO("state is changed [%s] -> [%s]", __state_str[old], __state_str[new]);
+       LOG_INFO("webrtc[%p] state is changed [%s] -> [%s]", webrtc, __state_str[old], __state_str[new]);
 
        if (webrtc->state_changed_cb.callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", webrtc->state_changed_cb.callback, webrtc->state_changed_cb.user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", webrtc->state_changed_cb.callback, webrtc->state_changed_cb.user_data);
                ((webrtc_state_changed_cb)(webrtc->state_changed_cb.callback))((webrtc_h)webrtc, old, new, webrtc->state_changed_cb.user_data);
                LOG_DEBUG("<<< end of the callback");
        }
@@ -239,10 +239,10 @@ static void __invoke_error_cb(webrtc_s *webrtc, webrtc_error_e error)
 {
        RET_IF(webrtc == NULL, "webrtc is NULL");
 
-       LOG_ERROR("error[0x%x, %s]", error, __get_error_string(error));
+       LOG_ERROR("webrtc[%p] error[0x%x, %s]", webrtc, error, __get_error_string(error));
 
        if (webrtc->error_cb.callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", webrtc->error_cb.callback, webrtc->error_cb.user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", webrtc->error_cb.callback, webrtc->error_cb.user_data);
                ((webrtc_error_cb)(webrtc->error_cb.callback))((webrtc_h)webrtc, error, webrtc->state, webrtc->error_cb.user_data);
                LOG_DEBUG("<<< end of the callback");
        }
@@ -256,8 +256,10 @@ static void __invoke_peer_connection_state_change_cb(webrtc_s *webrtc, GstWebRTC
 
        cb = &webrtc->peer_connection_state_change_cb;
 
+       LOG_INFO("webrtc[%p] peer connection state is changed to [%s]", webrtc, __peer_connection_state_info[state].str);
+
        if (cb->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", cb->callback, cb->user_data);
+               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");
        }
@@ -271,8 +273,10 @@ static void __invoke_signaling_state_change_cb(webrtc_s *webrtc, GstWebRTCPeerCo
 
        cb = &webrtc->signaling_state_change_cb;
 
+       LOG_INFO("webrtc[%p] signaling state is changed to [%s]", webrtc, __signaling_state_info[state].str);
+
        if (cb->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", cb->callback, cb->user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", cb->callback, cb->user_data);
                ((webrtc_signaling_state_change_cb)(cb->callback))((webrtc_h)webrtc, __signaling_state_info[state].state, cb->user_data);
                LOG_DEBUG("<<< end of the callback");
        }
@@ -286,8 +290,10 @@ static void __invoke_ice_gathering_state_change_cb(webrtc_s *webrtc, GstWebRTCIC
 
        cb = &webrtc->ice_gathering_state_change_cb;
 
+       LOG_INFO("webrtc[%p] ICE gathering state is changed to [%s]", webrtc, __ice_gathering_state_info[state].str);
+
        if (cb->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", cb->callback, cb->user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", cb->callback, cb->user_data);
                ((webrtc_ice_gathering_state_change_cb)(cb->callback))((webrtc_h)webrtc, __ice_gathering_state_info[state].state, cb->user_data);
                LOG_DEBUG("<<< end of the callback");
        }
@@ -301,8 +307,10 @@ static void __invoke_ice_connection_state_change_cb(webrtc_s *webrtc, GstWebRTCI
 
        cb = &webrtc->ice_connection_state_change_cb;
 
+       LOG_INFO("webrtc[%p] ICE connection state is changed to [%s]", webrtc, __ice_connection_state_info[state].str);
+
        if (cb->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", cb->callback, cb->user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", cb->callback, cb->user_data);
                ((webrtc_ice_connection_state_change_cb)(cb->callback))((webrtc_h)webrtc, __ice_connection_state_info[state].state, cb->user_data);
                LOG_DEBUG("<<< end of the callback");
        }
@@ -577,7 +585,7 @@ GstElement *_create_element(const char *factory_name, const char *name)
        element = gst_element_factory_make(factory_name, name);
        RET_VAL_IF(!element, NULL, "element is NULL [%s]", factory_name);
 
-       LOG_INFO("created element [%s, %s]", factory_name, SAFE_STR(name));
+       LOG_INFO("created element[%p, %s, %s]", element, factory_name, SAFE_STR(name));
 
        return element;
 }
@@ -804,8 +812,8 @@ static void __webrtcbin_on_negotiation_needed_cb(GstElement *webrtcbin, gpointer
                return;
        }
 
-       LOG_DEBUG(">>> invoke negotiation_needed_cb[%p], user_data[%p]",
-               webrtc->negotiation_needed_cb.callback, webrtc->negotiation_needed_cb.user_data);
+       LOG_DEBUG(">>> webrtc[%p] invoke negotiation_needed_cb[%p] user_data[%p]",
+               webrtc, webrtc->negotiation_needed_cb.callback, webrtc->negotiation_needed_cb.user_data);
        ((webrtc_negotiation_needed_cb)(webrtc->negotiation_needed_cb.callback))((webrtc_h)webrtc, webrtc->negotiation_needed_cb.user_data);
        LOG_DEBUG("<<< end of the callback");
 }
@@ -820,7 +828,7 @@ static void __webrtcbin_on_ice_candidate_cb(GstElement *webrtcbin, guint mlinein
        RET_IF(webrtc == NULL, "webrtc is NULL");
        RET_IF(webrtc->ice_candidate_cb.callback == NULL, "ice_candidate_cb is NULL");
 
-       LOG_DEBUG("mlineindex[%u], candidate[%s]", mlineindex, candidate);
+       LOG_DEBUG("webrtc[%p] mlineindex[%u] candidate[%s]", webrtc, mlineindex, candidate);
 
        _candidate = __make_ice_candidate_message(mlineindex, candidate);
        if (!_candidate)
@@ -829,8 +837,8 @@ static void __webrtcbin_on_ice_candidate_cb(GstElement *webrtcbin, guint mlinein
        _param_candidate = strdup(_candidate);
        g_free(_candidate);
 
-       LOG_DEBUG(">>> invoke ice_candidate_cb[%p], user_data[%p]",
-               webrtc->ice_candidate_cb.callback, webrtc->ice_candidate_cb.user_data);
+       LOG_DEBUG(">>> webrtc[%p] invoke ice_candidate_cb[%p] user_data[%p]",
+               webrtc, webrtc->ice_candidate_cb.callback, webrtc->ice_candidate_cb.user_data);
        ((webrtc_ice_candidate_cb)(webrtc->ice_candidate_cb.callback))((webrtc_h)webrtc, (const char *)_param_candidate, webrtc->ice_candidate_cb.user_data);
        LOG_DEBUG("<<< end of the callback");
 
@@ -847,7 +855,7 @@ static void __webrtcbin_peer_connection_state_cb(GstElement *webrtcbin, GParamSp
 
        g_object_get(webrtcbin, "connection-state", &state, NULL);
 
-       LOG_DEBUG("[PeerConnectionState] is changed to [%s]", __peer_connection_state_info[state].str);
+       LOG_DEBUG("webrtc[%p] [PeerConnectionState] is changed to [%s]", webrtc, __peer_connection_state_info[state].str);
        webrtc->negotiation_states.peer_connection_state = __peer_connection_state_info[state].state;
 
        __post_peer_connection_state_change_cb_in_idle(webrtc, state);
@@ -879,7 +887,7 @@ static void __webrtcbin_signaling_state_cb(GstElement *webrtcbin, GParamSpec * p
 
        g_object_get(webrtcbin, "signaling-state", &state, NULL);
 
-       LOG_DEBUG("[SignalingState] is changed to [%s]", __signaling_state_info[state].str);
+       LOG_DEBUG("webrtc[%p] [SignalingState] is changed to [%s]", webrtc, __signaling_state_info[state].str);
        webrtc->negotiation_states.signaling_state = __signaling_state_info[state].state;
 
        __post_signaling_state_change_cb_in_idle(webrtc, state);
@@ -895,7 +903,7 @@ static void __webrtcbin_ice_gathering_state_cb(GstElement *webrtcbin, GParamSpec
 
        g_object_get(webrtcbin, "ice-gathering-state", &state, NULL);
 
-       LOG_DEBUG("[IceGatheringState] is changed to [%s]", __ice_gathering_state_info[state].str);
+       LOG_DEBUG("webrtc[%p] [IceGatheringState] is changed to [%s]", webrtc, __ice_gathering_state_info[state].str);
        webrtc->negotiation_states.ice_gathering_state = __ice_gathering_state_info[state].state;
 
        __post_ice_gathering_state_change_cb_in_idle(webrtc, state);
@@ -911,7 +919,7 @@ static void __webrtcbin_ice_connection_state_cb(GstElement *webrtcbin, GParamSpe
 
        g_object_get(webrtcbin, "ice-connection-state", &state, NULL);
 
-       LOG_DEBUG("[IceConnectionState] is changed to [%s]", __ice_connection_state_info[state].str);
+       LOG_DEBUG("webrtc[%p] [IceConnectionState] is changed to [%s]", webrtc, __ice_connection_state_info[state].str);
        webrtc->negotiation_states.ice_connection_state = __ice_connection_state_info[state].state;
 
        __post_ice_connection_state_change_cb_in_idle(webrtc, state);
@@ -1135,8 +1143,8 @@ static void __webrtcbin_on_new_transceiver_cb(GstElement *webrtcbin, GstWebRTCRT
        RET_IF(transceiver == NULL, "transceiver is NULL");
        RET_IF(webrtc == NULL, "webrtc is NULL");
 
-       LOG_INFO("new transceiver[%p, mline:%u, mid:%s, direction:%d], user_data[%p]",
-               transceiver, transceiver->mline, transceiver->mid, transceiver->direction, user_data);
+       LOG_INFO("webrtc[%p] new transceiver[%p, mline:%u, mid:%s, direction:%d] user_data[%p]",
+               webrtc, transceiver, transceiver->mline, transceiver->mid, transceiver->direction, user_data);
 
        if (g_hash_table_size(webrtc->gst.source_slots) == 0) {
                /* In this case, it might be an answerer without setting any media source.
@@ -1371,6 +1379,8 @@ static void __update_session_description(GstPromise *promise, bool is_offer, gpo
                g_free(webrtc->desc_answer);
                webrtc->desc_answer = sdp_msg;
        }
+       LOG_INFO("webrtc[%p, %s] %s", webrtc, is_offer ? "offer" : "answer",
+               is_offer ? webrtc->desc_offer : webrtc->desc_answer);
 }
 
 static void __offer_created_cb(GstPromise *promise, gpointer user_data)
@@ -1389,8 +1399,9 @@ static void __offer_created_cb(GstPromise *promise, gpointer user_data)
        LOG_DEBUG_ENTER();
 
        __update_session_description(promise, true, webrtc);
+
        if (data->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", data->callback, data->user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", data->callback, data->user_data);
                ((webrtc_session_description_created_cb)(data->callback))((webrtc_h)webrtc, webrtc->desc_offer, data->user_data);
                LOG_DEBUG("<<< end of the callback");
                g_mutex_unlock(&webrtc->desc_mutex);
@@ -1422,7 +1433,7 @@ static void __answer_created_cb(GstPromise *promise, gpointer user_data)
        __update_session_description(promise, false, webrtc);
 
        if (data->callback) {
-               LOG_DEBUG(">>> callback[%p], user_data[%p]", data->callback, data->user_data);
+               LOG_DEBUG(">>> callback[%p] user_data[%p]", data->callback, data->user_data);
                ((webrtc_session_description_created_cb)(data->callback))((webrtc_h)webrtc, webrtc->desc_answer, data->user_data);
                LOG_DEBUG("<<< end of the callback");
                g_mutex_unlock(&webrtc->desc_mutex);
@@ -1465,8 +1476,6 @@ int _webrtcbin_create_session_description(webrtc_s *webrtc, bool is_offer, char
 
        g_mutex_unlock(&webrtc->desc_mutex);
 
-       LOG_INFO("%s", *desc);
-
        return WEBRTC_ERROR_NONE;
 }
 

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index afe121fdccac8749edee73eb13fba39436663e97..ac7db7aa94bf71b5dac9931845f779f906abde0b 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -317,7 +317,7 @@ static void __invoke_track_added_cb(webrtc_s *webrtc, const gchar *name, bool is
        RET_IF(webrtc == NULL, "webrtc is NULL");
        RET_IF(name == NULL, "name is NULL");
 
-       LOG_INFO("[%s] track[%s] is added", is_video ? "video" : "audio", name);
+       LOG_INFO("webrtc[%p] [%s] track[%s] is added", webrtc, is_video ? "video" : "audio", name);
 
        if (webrtc->track_added_cb.callback) {
                unsigned int id = __get_id_from_name(name);
@@ -369,7 +369,7 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
                return;
 
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(new_pad), 0));
-       LOG_INFO("[%s], new_pad[%s], media_type[%s]", GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad), media_type);
+       LOG_INFO("decodebin[%p, name:%s] new_pad[%s] media_type[%s]", decodebin, GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad), media_type);
 
        sink = __find_sink_slot(webrtc, GST_ELEMENT_NAME(decodebin));
        RET_IF(sink == NULL, "could not find an item by [%s] in sink slots", GST_ELEMENT_NAME(decodebin));
@@ -423,7 +423,7 @@ int _decodebin_autoplug_select_cb(GstElement *decodebin, GstPad *pad, GstCaps *c
        factory_name = GST_OBJECT_NAME(factory);
        klass = gst_element_factory_get_metadata(factory, GST_ELEMENT_METADATA_KLASS);
 
-       LOG_INFO("factory [name:%s, klass:%s]", factory_name, klass);
+       LOG_INFO("decodebin[%p] factory[name:%s, klass:%s]", decodebin, factory_name, klass);
 
        str_arr = webrtc->ini.general.gst_excluded_elements;
        while (str_arr && *str_arr) {
@@ -914,7 +914,7 @@ int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_video)
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       LOG_INFO("added a sink slot[%p, id:%u]", sink, sink->id);
+       LOG_INFO("added a sink slot[%p, id:%u] to webrtc[%p]", sink, sink->id, webrtc);
 
        return WEBRTC_ERROR_NONE;
 
@@ -968,7 +968,7 @@ int _set_stream_info_to_sink(webrtc_s *webrtc, unsigned int track_id, sound_stre
        sink->sound_stream_info.type = strdup(stream_type);
        sink->sound_stream_info.index = stream_index;
 
-       LOG_INFO("track_id[%u] stream_info[%p, type:%s, index:%d]", track_id, stream_info, stream_type, stream_index);
+       LOG_INFO("webrtc[%p] track_id[%u] stream_info[%p, type:%s, index:%d]", webrtc, track_id, stream_info, stream_type, stream_index);
 
        return WEBRTC_ERROR_NONE;
 }
@@ -992,7 +992,7 @@ int _set_display_to_sink(webrtc_s *webrtc, unsigned int track_id, unsigned int t
                RET_VAL_IF(sink->display == NULL, WEBRTC_ERROR_INVALID_OPERATION, "sink->display is NULL");
        }
 
-       LOG_INFO("track_id[%u]", track_id);
+       LOG_INFO("webrtc[%p] track_id[%u]", webrtc, track_id);
 
        _set_display_type_and_surface(sink->display, type, display);