webrtc_source: rearrange codes to reduce code complexity

[Version] 0.2.117
[Issue Type] Refactoring

Change-Id: I31b43afb40ae1bd5a829dc1ccb99685bd4ead1a4

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index db161ffda9407bd9ebaaddc14b0e814e4b074dd3..12ae164673e890983dd45c3f5647e99324aea3e6 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:    0.2.116
+Version:    0.2.117
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 55a99ecdb43ec7bd1b47abbcd889326bb2b80ecb..7fb0d64a4107278ed61d446f7abe024bfad2925e 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1716,18 +1716,28 @@ error:
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
-static void __remove_rest_of_filesrc_element(webrtc_gst_slot_s *source, bool is_audio)
+static void __remove_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
 {
-       GstElement *queue = NULL;
+       GstBin *bin = NULL;
+
+       GstElement *payload = NULL;
        GstElement *capsfilter = NULL;
+       GstElement *fakesink = NULL;
+
+       RET_IF(source == NULL, "pad is NULL");
+
+       bin = GST_BIN(source->filesrc_pipeline);
 
-       queue = gst_bin_get_by_name(source->bin, is_audio ? DEFAULT_NAME_AUDIO_QUEUE : DEFAULT_NAME_VIDEO_QUEUE);
-       RET_IF(queue == NULL, "queue is NULL");
+       payload = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD: DEFAULT_NAME_VIDEO_PAYLOAD);
+       RET_IF(payload == NULL, "payload is NULL");
 
-       capsfilter = gst_bin_get_by_name(source->bin, is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER);
+       capsfilter = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER);
        RET_IF(capsfilter == NULL, "capsfilter is NULL");
 
-       gst_bin_remove_many(source->bin, queue, capsfilter, NULL);
+       fakesink = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_FAKESINK: DEFAULT_NAME_VIDEO_FAKESINK);
+       RET_IF(fakesink == NULL, "fakesink is NULL");
+
+       gst_bin_remove_many(bin, payload, capsfilter, fakesink, NULL);
 }
 
 static void __filesrc_pipeline_audio_stream_handoff_cb(GstElement *object, GstBuffer *buffer, GstPad *pad, gpointer data)
@@ -1775,6 +1785,25 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info
        return GST_PAD_PROBE_REMOVE;
 }
 
+static GstElement * __create_payload_for_filesrc_pipeline(GstPad *pad, bool is_audio)
+{
+       element_info_s elem_info;
+       GstElement *payload = NULL;
+
+       RET_VAL_IF(pad == NULL, NULL, "pad is NULL");
+
+       CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_PAYLOADER_RTP,
+                                                               gst_pad_get_current_caps(pad),
+                                                               NULL,
+                                                               payload);
+
+       RET_VAL_IF(payload == NULL, NULL, "payload is NULL");
+
+       gst_element_set_name(payload, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+
+       return payload;
+}
+
 static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
 {
        GstElement *capsfilter = NULL;
@@ -1826,33 +1855,43 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
        return fakesink;
 }
 
-static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstElement *payloader, bool is_audio)
+static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstPad *pad, bool is_audio)
 {
        GstBin *bin = NULL;
+       GstElement *payload = NULL;
        GstElement *capsfilter = NULL;
        GstElement *fakesink = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-       RET_VAL_IF(payloader == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "payloader is NULL");
        RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
 
-       if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio)))
+       if (!(payload = __create_payload_for_filesrc_pipeline(pad, is_audio)))
                return WEBRTC_ERROR_INVALID_OPERATION;
 
+       if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio))) {
+               SAFE_GST_OBJECT_UNREF(payload);
+               return WEBRTC_ERROR_INVALID_OPERATION;
+       }
        if (!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, is_audio))) {
+               SAFE_GST_OBJECT_UNREF(payload);
                SAFE_GST_OBJECT_UNREF(capsfilter);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       gst_bin_add_many(bin, capsfilter, fakesink, NULL);
+       gst_bin_add_many(bin, payload, capsfilter, fakesink, NULL);
 
-       if (!gst_element_link_many(payloader, capsfilter, fakesink, NULL)) {
+       if (!gst_element_link_many(payload, capsfilter, fakesink, NULL)) {
                LOG_ERROR("failed to gst_element_link_many()");
                goto error;
        }
 
+       if (!gst_element_sync_state_with_parent(payload)) {
+               LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(payload));
+               goto error;
+       }
+
        if (!gst_element_sync_state_with_parent(capsfilter)) {
                LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(capsfilter));
                goto error;
@@ -1866,81 +1905,54 @@ static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *sou
        return WEBRTC_ERROR_NONE;
 
 error:
-       gst_bin_remove_many(bin, capsfilter, fakesink, NULL);
+       gst_bin_remove_many(bin, payload, capsfilter, fakesink, NULL);
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
-static GstElement * __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source, bool is_audio, bool create)
+static int __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source, bool is_audio)
 {
-       element_info_s elem_info;
-       GstBin *bin = NULL;
        GstElement *payload = NULL;
-       GstPad *payload_sink_pad = NULL;
-
-       RET_VAL_IF(pad == NULL, NULL, "pad is NULL");
-       RET_VAL_IF(source == NULL, NULL, "source is NULL");
-       RET_VAL_IF(source->filesrc_pipeline == NULL, NULL, "filesrc_pipeline is NULL");
-
-       bin = GST_BIN(source->filesrc_pipeline);
-
-       if (create) {
-               CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_PAYLOADER_RTP,
-                                                       gst_pad_get_current_caps(pad),
-                                                       NULL,
-                                                       payload);
-
-               RET_VAL_IF(payload == NULL, NULL, "There is no payload that supports this content.");
-
-               gst_element_set_name(payload, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+       GstPad *sink_pad = NULL;
 
-               gst_bin_add(bin, payload);
+       RET_VAL_IF(pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pad is NULL");
+       RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 
-               if (!gst_element_sync_state_with_parent(payload)) {
-                       LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(payload));
-                       gst_bin_remove(bin, payload);
-                       return NULL;
-               }
-       } else {
-               payload = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
-               RET_VAL_IF(payload == NULL, NULL, "There is no payload to link");
+       payload = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+       if (!payload) {
+               LOG_ERROR("failed to get element [%s]", is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+               return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       payload_sink_pad = gst_element_get_static_pad(payload, "sink");
-       if (!payload_sink_pad) {
-               LOG_ERROR("payload_sink_pad is NULL");
-               gst_bin_remove(bin, payload);
-               return NULL;
+       sink_pad = gst_element_get_static_pad(payload, "sink");
+       if (!sink_pad) {
+               LOG_ERROR("sink_pad is NULL for [%s]", GST_ELEMENT_NAME(payload));
+               return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (gst_pad_link(pad, payload_sink_pad) != GST_PAD_LINK_OK) {
+       if (gst_pad_link(pad, sink_pad) != GST_PAD_LINK_OK) {
                LOG_ERROR("failed to gst_pad_link()");
-               gst_bin_remove(bin, payload);
-               g_object_unref(payload_sink_pad);
-               return NULL;
+               g_object_unref(sink_pad);
+               return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       g_object_unref(payload_sink_pad);
+       g_object_unref(sink_pad);
 
        LOG_INFO("decodebin is linked to [%s]", GST_ELEMENT_NAME(payload));
 
-       return payload;
+       return WEBRTC_ERROR_NONE;
 }
 
 static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPad *pad, gpointer data)
 {
        int ret = WEBRTC_ERROR_NONE;
        webrtc_gst_slot_s *source = data;
-       GstBin *bin = NULL;
        const gchar *media_type = NULL;
-       GstElement *payload = NULL;
        gboolean is_audio;
        int av_idx;
 
        RET_IF(source == NULL, "source is NULL");
        RET_IF(source->filesrc_pipeline == NULL, "filesrc_pipeline is NULL");
 
-       bin = GST_BIN(source->filesrc_pipeline);
-
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0));
        RET_IF(media_type == NULL, "media_type is NULL");
 
@@ -1956,32 +1968,34 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
 
        if (source->av[av_idx].src_pad_probe_id > 0) {
                LOG_INFO("Pipeline already built");
-               __link_decodebin_with_payload(pad, source, is_audio, false);
+               __link_decodebin_with_payload(pad, source, is_audio);
                return;
        }
 
-       payload = __link_decodebin_with_payload(pad, source, is_audio, true);
-       RET_IF(payload == NULL, "failed to __link_decodebin_with_payload()");
-
-       ret = __create_rest_of_elements_for_filesrc_pipeline(source, payload, is_audio);
+       ret = __create_rest_of_elements_for_filesrc_pipeline(source, pad, is_audio);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __create_rest_of_elements_for_filesrc_pipeline()");
-               gst_bin_remove(bin, payload);
+               return;
+       }
+
+       ret = __link_decodebin_with_payload(pad, source, is_audio);
+       if (ret != WEBRTC_ERROR_NONE) {
+               LOG_ERROR("failed to __link_decodebin_with_payload()");
+               __remove_rest_of_elements_for_filesrc_pipeline(source, is_audio);
                return;
        }
 
        ret = __build_filesrc_bin(source, is_audio ? MEDIA_TYPE_AUDIO : MEDIA_TYPE_VIDEO);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __build_filesrc_bin()");
-               gst_bin_remove(bin, payload);
+               __remove_rest_of_elements_for_filesrc_pipeline(source, is_audio);
                return;
        }
 
        ret = __link_source_with_webrtcbin(source, source->webrtc->gst.webrtcbin);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __link_source_with_webrtcbin()");
-               gst_bin_remove(bin, payload);
-               __remove_rest_of_filesrc_element(source, is_audio);
+               __remove_rest_of_elements_for_filesrc_pipeline(source, is_audio);
                g_hash_table_remove(source->webrtc->gst.source_slots, GST_ELEMENT_NAME(source->bin));
                return;
        }