webrtc_source: Remove unnecessary capsfilter in screen source pipeline

Unnecessary logs are removed.

[Version] 0.1.156
[Issue Type] Improvement

Change-Id: I3dcc19f1bdb07c74cabd367bf4151c0138b4ebd6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 930a42b128f0b31d6f882baf81889bfab1ddf2b7..10237fbcecea59058fb2d99de9eb3f66780d86dd 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:    0.1.155
+Version:    0.1.156
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index c7fe2b0d7ff5da766ced35382dffa42ea94af3d2..66c19f842d259a69d60e09690e3029db88af5b8f 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -713,65 +713,16 @@ static const char *__get_source_element(webrtc_s *webrtc, webrtc_media_source_ty
        return source->source_element;
 }
 
-static int __create_elements_for_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source,
-       GstElement **screensrc, GstElement **capsfilter, GstElement **videoconvert)
-{
-       GstCaps *caps = NULL;
-       gchar *caps_str = NULL;
-       ini_item_media_source_s *ini_source  = NULL;
-
-       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
-       RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-       RET_VAL_IF(screensrc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "screensrc is NULL");
-       RET_VAL_IF(capsfilter == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "capsfilter is NULL");
-       RET_VAL_IF(videoconvert == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "videoconvert is NULL");
-
-       if (!(*screensrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_SCREEN), NULL))) {
-               LOG_ERROR("failed to create screensrc");
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
-
-       if (!(*videoconvert = _create_element(DEFAULT_ELEMENT_VIDEOCONVERT, NULL))) {
-               LOG_ERROR("failed to create videoconvert");
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
-
-       if (!(*capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, NULL))) {
-               LOG_ERROR("failed to create capsfilter");
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
-
-       ini_source = _ini_get_source_by_type(&webrtc->ini, source->type);
-       if (ini_source == NULL)
-               ini_source = &webrtc->ini.media_source;
-
-       caps = gst_caps_new_simple(MEDIA_TYPE_VIDEO_RAW,
-                                       "format", G_TYPE_STRING, "BGRA",
-                                       "framerate", GST_TYPE_FRACTION, ini_source->v_framerate, 1,
-                                       "width", G_TYPE_INT, ini_source->v_width,
-                                       "height", G_TYPE_INT, ini_source->v_height,
-                                       NULL);
-       caps_str = gst_caps_to_string(caps);
-       LOG_INFO("capsfilter caps[%s] for screensrc", caps_str);
-       g_free(caps_str);
-
-       g_object_set(G_OBJECT(*capsfilter), "caps", caps, NULL);
-       gst_caps_unref(caps);
-
-       return WEBRTC_ERROR_NONE;
-}
-
 static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
        int ret = WEBRTC_ERROR_NONE;
-       GstElement *screensrc = NULL;
+       GstElement *screensrc;
+       GstElement *videoconvert;
        GstElement *capsfilter1 = NULL;
-       GstElement *videoconvert = NULL;
-       GstElement *capsfilter2 = NULL;
        GstElement *videoenc = NULL;
        GstElement *videopay = NULL;
        GstElement *queue = NULL;
-       GstElement *capsfilter3 = NULL;
+       GstElement *capsfilter2 = NULL;
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -780,20 +731,23 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
        source->media_types = MEDIA_TYPE_VIDEO;
 
-       if ((ret = __create_elements_for_screensrc(webrtc, source, &screensrc, &capsfilter1, &videoconvert)) != WEBRTC_ERROR_NONE)
+       if (!(screensrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_SCREEN), NULL)))
+               return WEBRTC_ERROR_INVALID_OPERATION;
+
+       if (!(videoconvert = _create_element(DEFAULT_ELEMENT_VIDEOCONVERT, NULL)))
                goto exit;
 
-       if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter2, &videoenc, &videopay, &queue, &capsfilter3)) != WEBRTC_ERROR_NONE)
+       if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter1, &videoenc, &videopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
                goto exit;
 
-       gst_bin_add_many(source->bin, screensrc, capsfilter1, videoconvert, capsfilter2, videoenc, videopay, queue, capsfilter3, NULL);
-       if (!gst_element_link_many(screensrc, capsfilter1, videoconvert, capsfilter2, videoenc, videopay, queue, capsfilter3, NULL)) {
+       gst_bin_add_many(source->bin, screensrc, videoconvert, capsfilter1, videoenc, videopay, queue, capsfilter2, NULL);
+       if (!gst_element_link_many(screensrc, videoconvert, capsfilter1, videoenc, videopay, queue, capsfilter2, NULL)) {
                LOG_ERROR("failed to gst_element_link_many()");
                ret = WEBRTC_ERROR_INVALID_OPERATION;
                goto exit_with_remove_from_bin;
        }
 
-       ret = _set_ghost_pad_target(source->src_pad, capsfilter3, true);
+       ret = _set_ghost_pad_target(source->src_pad, capsfilter2, true);
        if (ret != WEBRTC_ERROR_NONE)
                goto exit_with_remove_from_bin;
 
@@ -801,17 +755,16 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
 exit_with_remove_from_bin:
        /* elements will be dereferenced */
-       gst_bin_remove_many(source->bin, screensrc, capsfilter1, videoconvert, capsfilter2, videoenc, videopay, queue, capsfilter3, NULL);
+       gst_bin_remove_many(source->bin, screensrc, videoconvert, capsfilter1, videoenc, videopay, queue, capsfilter2, NULL);
        return ret;
 exit:
        SAFE_GST_OBJECT_UNREF(screensrc);
-       SAFE_GST_OBJECT_UNREF(capsfilter1);
        SAFE_GST_OBJECT_UNREF(videoconvert);
-       SAFE_GST_OBJECT_UNREF(capsfilter2);
+       SAFE_GST_OBJECT_UNREF(capsfilter1);
        SAFE_GST_OBJECT_UNREF(videoenc);
        SAFE_GST_OBJECT_UNREF(videopay);
        SAFE_GST_OBJECT_UNREF(queue);
-       SAFE_GST_OBJECT_UNREF(capsfilter3);
+       SAFE_GST_OBJECT_UNREF(capsfilter2);
        return ret;
 }
 
@@ -834,10 +787,9 @@ static int __build_camerasrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 #ifdef TIZEN_FEATURE_RES_MGR
        webrtc->resource.need_to_acquire[MM_RESOURCE_MANAGER_RES_TYPE_CAMERA] = true;
 #endif
-       if (!(camerasrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_CAMERA), NULL))) {
-               LOG_ERROR("failed to create camerasrc");
+       if (!(camerasrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_CAMERA), NULL)))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+
        /* FIXME: set camera default setting from ini */
 
        /* NOTE: in case of an element that supports tizen zerocopy format, not to emit an error in GST_STATE_PLAYING
@@ -892,10 +844,8 @@ static int __build_audiosrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
        source->media_types = MEDIA_TYPE_AUDIO;
 
-       if (!(audiosrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_MIC), NULL))) {
-               LOG_ERROR("failed to create audiosrc");
+       if (!(audiosrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_MIC), NULL)))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       }
 
        if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &audioenc, &audiopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
                goto exit;
@@ -944,10 +894,9 @@ static int __build_videotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
        source->media_types = MEDIA_TYPE_VIDEO;
 
-       if (!(videotestsrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST), NULL))) {
-               LOG_ERROR("failed to create videotestsrc");
+       if (!(videotestsrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST), NULL)))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+
        g_object_set(G_OBJECT(videotestsrc), "is-live", TRUE, "pattern", 18, NULL); /* 18: ball */
 
        if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &videoenc, &videopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
@@ -997,10 +946,9 @@ static int __build_audiotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
        source->media_types = MEDIA_TYPE_AUDIO;
 
-       if (!(audiotestsrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST), NULL))) {
-               LOG_ERROR("failed to create audiotestsrc");
+       if (!(audiotestsrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST), NULL)))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+
        g_object_set(G_OBJECT(audiotestsrc), "is-live", TRUE, NULL);
 
        if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &audioenc, &audiopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)