webrtc_source: Fix invalid return value

Some cases returned ERROR_NONE despite error situations.
These are fixed.

[Version] 0.3.111
[Issue Type] Bug fix

Change-Id: I1be17af6f644754a8181f5fe8c349b99edf75c14
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index e55903605766389d1c90f75caeb0e8ad478697b3..ea3c57e2bc9017f40bfaef4227c69e635dd65a03 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:    0.3.110
+Version:    0.3.111
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 00f23495e61d19f554c0a54115c692f0313d1ffb..c96352a76b60485b9703af0f4ca94d216fdc990a 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1312,7 +1312,7 @@ static GstElement *__find_element_in_bin(GstBin *bin, const gchar *name)
 
 static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
-       int ret = WEBRTC_ERROR_NONE;
+       int ret;
        GstElement *screensrc;
        GstElement *videoconvert;
        GstElement *videotestsrc;
@@ -1362,7 +1362,7 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
                APPEND_ELEMENT(element_list, videoconvert);
        }
 
-       if ((ret = __create_rest_of_elements(webrtc, source, true, &element_list, false)) != WEBRTC_ERROR_NONE)
+       if (__create_rest_of_elements(webrtc, source, true, &element_list, false) != WEBRTC_ERROR_NONE)
                goto exit;
 
        if (!__add_elements_to_bin(source->bin, switch_src_list)) {
@@ -1377,22 +1377,16 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!__link_switch_srcs(videoswitch, switch_src_list)) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!__link_switch_srcs(videoswitch, switch_src_list))
                goto exit_with_remove_from_bin;
-       }
 
-       if (!__link_elements(element_list)) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!__link_elements(element_list))
                goto exit_with_remove_from_bin;
-       }
 
-       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER))) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER)))
                goto exit_with_remove_from_bin;
-       }
-       ret = _set_ghost_pad_target(src_pad, capsfilter, true);
-       if (ret != WEBRTC_ERROR_NONE)
+
+       if (_set_ghost_pad_target(src_pad, capsfilter, true) != WEBRTC_ERROR_NONE)
                goto exit_with_remove_from_bin;
 
        __add_probe_to_pad_for_pause(source, AV_IDX_VIDEO, src_pad, __payloaded_data_probe_cb);
@@ -1407,16 +1401,15 @@ exit_with_remove_from_bin:
        __remove_elements_from_bin(source->bin, element_list);
        SAFE_G_LIST_FREE(switch_src_list);
        SAFE_G_LIST_FREE(element_list);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 exit:
        SAFE_G_LIST_FREE_FULL(switch_src_list, gst_object_unref);
        SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
 static int __build_rest_of_videosrc(webrtc_s *webrtc, GstPad *src_pad, GstElement *videosrc_element, webrtc_gst_slot_s *source)
 {
-       int ret = WEBRTC_ERROR_NONE;
        GList *element_list = NULL;
        GstElement *capsfilter;
        const ini_item_media_source_s *ini_source;
@@ -1440,7 +1433,7 @@ static int __build_rest_of_videosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
        if (g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(videosrc_element)), "empty-buffer-timeout"))
                g_object_set(G_OBJECT(videosrc_element), "empty-buffer-timeout", 0, NULL);
 
-       if ((ret = __create_rest_of_elements(webrtc, source, true, &element_list, false)) != WEBRTC_ERROR_NONE)
+       if (__create_rest_of_elements(webrtc, source, true, &element_list, false) != WEBRTC_ERROR_NONE)
                goto exit;
 
        if (!__add_elements_to_bin(source->bin, element_list)) {
@@ -1448,17 +1441,13 @@ static int __build_rest_of_videosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!__link_elements(element_list)) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!__link_elements(element_list))
                goto exit_with_remove_from_bin;
-       }
 
-       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER))) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER)))
                goto exit_with_remove_from_bin;
-       }
-       ret = _set_ghost_pad_target(src_pad, capsfilter, true);
-       if (ret != WEBRTC_ERROR_NONE)
+
+       if (_set_ghost_pad_target(src_pad, capsfilter, true) != WEBRTC_ERROR_NONE)
                goto exit_with_remove_from_bin;
 
        __add_probe_to_pad_for_pause(source, AV_IDX_VIDEO, src_pad, __payloaded_data_probe_cb);
@@ -1470,10 +1459,10 @@ static int __build_rest_of_videosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
 exit_with_remove_from_bin:
        __remove_elements_from_bin(source->bin, element_list);
        SAFE_G_LIST_FREE(element_list);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 exit:
        SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
 static int __build_camerasrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
@@ -1503,7 +1492,6 @@ static int __build_camerasrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
 static int __build_rest_of_audiosrc(webrtc_s *webrtc, GstPad *src_pad, GstElement *audiosrc_element, webrtc_gst_slot_s *source)
 {
-       int ret = WEBRTC_ERROR_NONE;
        GList *element_list = NULL;
        GstElement *volume;
        GstElement *capsfilter;
@@ -1530,7 +1518,7 @@ static int __build_rest_of_audiosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
        source->av[AV_IDX_AUDIO].inbandfec = ini_source->use_inbandfec;
        source->av[AV_IDX_AUDIO].packet_loss_percentage = ini_source->packet_loss_percentage;
 
-       if ((ret = __create_rest_of_elements(webrtc, source, true, &element_list, true)) != WEBRTC_ERROR_NONE)
+       if (__create_rest_of_elements(webrtc, source, true, &element_list, true) != WEBRTC_ERROR_NONE)
                goto exit;
 
        if (!__add_elements_to_bin(source->bin, element_list)) {
@@ -1538,17 +1526,13 @@ static int __build_rest_of_audiosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!__link_elements(element_list)) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!__link_elements(element_list))
                goto exit_with_remove_from_bin;
-       }
 
-       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER))) {
-               ret = WEBRTC_ERROR_INVALID_OPERATION;
+       if (!(capsfilter = gst_bin_get_by_name(source->bin, ELEMENT_NAME_RTP_CAPSFILTER)))
                goto exit_with_remove_from_bin;
-       }
-       ret = _set_ghost_pad_target(src_pad, capsfilter, true);
-       if (ret != WEBRTC_ERROR_NONE)
+
+       if (_set_ghost_pad_target(src_pad, capsfilter, true) != WEBRTC_ERROR_NONE)
                goto exit_with_remove_from_bin;
 
        __add_probe_to_pad_for_pause(source, AV_IDX_AUDIO, src_pad, __payloaded_data_probe_cb);
@@ -1560,10 +1544,10 @@ static int __build_rest_of_audiosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
 exit_with_remove_from_bin:
        __remove_elements_from_bin(source->bin, element_list);
        SAFE_G_LIST_FREE(element_list);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 exit:
        SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
-       return ret;
+       return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
 static int __build_audiosrc(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool use_mic)
@@ -1705,8 +1689,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
        if (!__sync_elements_state_with_parent(element_list))
                goto exit_with_remove_from_bin;
 
-       ret = _set_ghost_pad_target(src_pad, capsfilter, true);
-       if (ret != WEBRTC_ERROR_NONE)
+       if (_set_ghost_pad_target(src_pad, capsfilter, true) != WEBRTC_ERROR_NONE)
                goto exit_with_remove_from_bin;
 
        __add_probe_to_pad_for_pause(source, av_idx, src_pad, __payloaded_data_probe_cb);