webrtc_source: Apply GENERATE_DOT() macro to loopback and filesrc

The filesrc pipeline name and loopback render pipeline name are
changed to be identified easily that which source belongs to it
by its name.

Dot file names are also changed.

[Version] 0.2.135
[Issue Type] Debug feature

Change-Id: I843eb7b4e8f42df20c5cd04a89c42a773e79af8c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 8e9b29afae278edd9c36541a565934f9648a9c12..ac77e418e84c6736b2e6859cea1b20d100c294cf 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:    0.2.134
+Version:    0.2.135
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 362b83f50bec8955c0a085909524ea33c0727f3a..f53c714c79958663f226df7a7b2c3e1bd70da8ca 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -229,7 +229,7 @@ void _invoke_state_changed_cb(webrtc_s *webrtc, webrtc_state_e old, webrtc_state
        if (new == WEBRTC_STATE_PLAYING)
                _set_stats_timer(webrtc);
 
-       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "STATE_%s", __state_str[webrtc->state]);
+       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s.state_%s", GST_ELEMENT_NAME(webrtc->gst.pipeline), __state_str[webrtc->state]);
 }
 
 static void __invoke_error_cb(webrtc_s *webrtc, webrtc_error_e error)
@@ -1116,7 +1116,7 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
                RET_IF(ret != WEBRTC_ERROR_NONE, "failed to _add_rendering_sink_bin()");
        }
 
-       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "webrtcbin_%s", GST_PAD_NAME(new_pad));
+       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s.webrtcbin-%s", GST_ELEMENT_NAME(webrtc->gst.pipeline), GST_PAD_NAME(new_pad));
 }
 
 static void __webrtcbin_no_more_pads_cb(GstElement *webrtcbin, gpointer user_data)
@@ -1185,7 +1185,7 @@ int _gst_build_pipeline(webrtc_s *webrtc)
 {
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 
-       webrtc->gst.pipeline = gst_pipeline_new("webrtc-pipeline");
+       webrtc->gst.pipeline = gst_pipeline_new("main-pipeline");
        RET_VAL_IF(webrtc->gst.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "pipeline is NULL");
 
        if (!(webrtc->gst.bus = gst_pipeline_get_bus(GST_PIPELINE(webrtc->gst.pipeline)))) {

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index 11b988d8aed334c08424d9587b8f63ed4d775a3e..828ade24c74852027cd32aef88df80fd2169e59d 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -394,7 +394,7 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
                _post_error_cb_in_idle(webrtc, ret);
        }
 
-       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s", GST_ELEMENT_NAME(decodebin));
+       GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s.%s", GST_ELEMENT_NAME(webrtc->gst.pipeline), GST_ELEMENT_NAME(decodebin));
 }
 
 static bool __is_factory_name_for_hw(gchar *factory_name)

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index e2dbcca6d8a56e0aff8fc5a4a293910ef99bab61..a85583474806a1f97adde1e976b650e177542d5e 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -2026,6 +2026,9 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        source->av[av_idx].render.need_decoding = true;
        source->av[av_idx].render.appsrc_caps = gst_pad_get_current_caps(pad);
        __add_probe_to_pad_for_render(source, av_idx, gst_element_get_static_pad(queue, "src"), __source_data_probe_cb);
+
+       GENERATE_DOT(source->webrtc, source->filesrc_pipeline, "%s.%s-%s",
+               GST_ELEMENT_NAME(source->filesrc_pipeline), GST_ELEMENT_NAME(element), GST_PAD_NAME(pad));
 }
 
 static GstAutoplugSelectResult __filesrc_pipeline_decodebin_autoplug_select_cb(GstElement *bin, GstPad *pad, GstCaps *caps, GstElementFactory* factory, gpointer udata)
@@ -2128,12 +2131,15 @@ static int __build_filesrc_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
        GstElement *filesrc = NULL;
        GstElement *decodebin = NULL;
+       gchar *pipeline_name;
 
        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(source->bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
 
-       source->filesrc_pipeline = gst_pipeline_new("filesrc-pipeline");
+       pipeline_name = g_strdup_printf("filesrc-pipeline-for-source_%u", source->id);
+       source->filesrc_pipeline = gst_pipeline_new(pipeline_name);
+       g_free(pipeline_name);
        RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "pipeline is NULL");
 
        if (!(source->filesrc_bus = gst_pipeline_get_bus(GST_PIPELINE(source->filesrc_pipeline)))) {
@@ -3858,11 +3864,17 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
                ret = __build_loopback_audiosink(source, decodebin);
                if (ret != WEBRTC_ERROR_NONE)
                        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_AUDIO].render.pipeline);
+               else
+                       GENERATE_DOT(source->webrtc, source->av[AV_IDX_AUDIO].render.pipeline, "%s.%s-%s",
+                               GST_ELEMENT_NAME(source->av[AV_IDX_AUDIO].render.pipeline), GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad));
 
        } else if (g_strrstr(media_type, "video")) {
                ret = __build_loopback_videosink(source, decodebin);
                if (ret != WEBRTC_ERROR_NONE)
                        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_VIDEO].render.pipeline);
+               else
+                       GENERATE_DOT(source->webrtc, source->av[AV_IDX_VIDEO].render.pipeline, "%s.%s-%s",
+                               GST_ELEMENT_NAME(source->av[AV_IDX_VIDEO].render.pipeline), GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad));
 
        } else {
                LOG_ERROR("not supported media type[%s]", media_type);
@@ -3888,7 +3900,7 @@ static int __build_loopback_render_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s
                RET_VAL_IF(source->display->surface == NULL, WEBRTC_ERROR_INVALID_OPERATION, "display->surface is NULL");
        }
 
-       pipeline_name = g_strdup_printf("webrtc-source_%u-%s-render-pipeline", source->id, GET_MEDIA_TYPE_NAME(type == MEDIA_TYPE_AUDIO));
+       pipeline_name = g_strdup_printf("loopback-pipeline-for-source_%u", source->id);
        source->av[idx].render.pipeline = gst_pipeline_new(pipeline_name);
        g_free(pipeline_name);
        RET_VAL_IF(source->av[idx].render.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "render_pipeline is NULL");
@@ -3950,6 +3962,9 @@ static int __build_loopback_render_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s
        if (track_id)
                *track_id = source->av[idx].render.track_id;
 
+       if (!source->av[idx].render.need_decoding)
+               GENERATE_DOT(webrtc, source->av[idx].render.pipeline, "%s", GST_ELEMENT_NAME(source->av[idx].render.pipeline));
+
        return WEBRTC_ERROR_NONE;
 
 error: