From: Sangchul Lee <sc11.lee@samsung.com>
Date: Fri, 18 Sep 2020 09:38:02 +0000 (+0900)
Subject: Generate dot files to take snapshots of pipeline
X-Git-Tag: submit/tizen/20210729.023123~214
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=476f6eb60db3f968e94e23fda61f9da63668cd7a;p=platform%2Fcore%2Fapi%2Fwebrtc.git

Generate dot files to take snapshots of pipeline

Add code to create dot files when
 - after invoking state changed callback
 - a decodebin is added inside of pad-added callback of the webrtcbin
 - a rendering sink is added inside of pad-added callback of the decodebin

[Version] 0.1.28
[Issue Type] Debug

Change-Id: I3a752d5af5cb58cf21fbca3e9e45785b5e542c1d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 76050b03..a54bc367 100644
--- 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:    0.1.27
+Version:    0.1.28
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 98d0aa42..34225b10 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -51,6 +51,9 @@
 #define DEFAULT_AUDIO_CHANNELS            1
 #define DEFAULT_AUDIO_SAMPLERATE          8000
 
+#define DEFAULT_DOT_FILE_NAME_PREFIX      "webrtc"
+#define DEFAULT_DOT_DIRECTORY             "/tmp"
+
 typedef enum {
 	CODEC_TYPE_OPUS,
 	CODEC_TYPE_VORBIS,
@@ -97,12 +100,41 @@ do { \
 	} \
 } while (0)
 
+#define GENERATE_DOT(x_webrtc, x_fmt, x_arg...) \
+do { \
+	gchar *dot_name; \
+	if (!x_webrtc->ini.generate_dot) \
+		break; \
+	dot_name = g_strdup_printf(""x_fmt"", x_arg); \
+	__generate_dot(x_webrtc, dot_name); \
+	g_free(dot_name); \
+} while (0)
+
 static const char* __state_str[] = {
 	"IDLE",
 	"NEGOTIATING",
 	"PLAYING",
 };
 
+static void __generate_dot(webrtc_s *webrtc, const gchar *name)
+{
+	gchar *dot_name;
+
+	RET_IF(webrtc == NULL, "webrtc is NULL");
+	RET_IF(webrtc->gst.pipeline == NULL, "pipeline is NULL");
+
+	if (!name)
+		dot_name = g_strdup(DEFAULT_DOT_FILE_NAME_PREFIX);
+	else
+		dot_name = g_strconcat(DEFAULT_DOT_FILE_NAME_PREFIX, ".", name, NULL);
+
+	GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(webrtc->gst.pipeline), GST_DEBUG_GRAPH_SHOW_ALL, dot_name);
+
+	LOG_INFO("dot file[%s] is generated", dot_name);
+
+	g_free(dot_name);
+}
+
 /* Use g_free() to release the return value. */
 static gchar* __get_string_from_json_object(JsonObject *object)
 {
@@ -203,6 +235,8 @@ static void __invoke_state_changed_cb(webrtc_s *webrtc, webrtc_state_e old, webr
 		((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");
 	}
+
+	GENERATE_DOT(webrtc, "STATE_%s", __state_str[webrtc->state]);
 }
 
 static gboolean __bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_data)
@@ -256,8 +290,6 @@ static gboolean __bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_d
 		LOG_INFO("GST_MESSAGE_STATE_CHANGED: %s", state_transition_name);
 		g_free(state_transition_name);
 
-		/* TODO: generate dot */
-
 		g_mutex_lock(&webrtc->mutex);
 		if (webrtc->pend_state == webrtc->state) {
 			LOG_DEBUG("pend_state[%s] is same with current state", __state_str[webrtc->pend_state]);
@@ -405,8 +437,14 @@ int _ini_load(webrtc_s *webrtc)
 	memset(&webrtc->ini, 0, sizeof(webrtc_ini_s));
 
 	/* FIXME: load from ini file */
+	/* set it TRUE temporarily until ini ready */
+	webrtc->ini.generate_dot = TRUE;
 
-	webrtc->ini.generate_dot = FALSE;
+	if (webrtc->ini.generate_dot) {
+		/* FIXME: get path from ini */
+		LOG_INFO("dot file will be stored in [%s]", DEFAULT_DOT_DIRECTORY);
+		g_setenv("GST_DEBUG_DUMP_DOT_DIR", DEFAULT_DOT_DIRECTORY, FALSE);
+	}
 
 	return WEBRTC_ERROR_NONE;
 }
@@ -830,6 +868,8 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
 
 	if (ret != WEBRTC_ERROR_NONE)
 		LOG_ERROR("failed to build a rendering pipeline");
+
+	GENERATE_DOT(webrtc, "%s", GST_ELEMENT_NAME(decodebin));
 }
 
 static int __decodebin_autoplug_select_cb(GstElement *decodebin, GstPad *pad, GstCaps *caps, GstElementFactory *factory, gpointer user_data)
@@ -1029,6 +1069,8 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
 
 	ret = __add_rendering_sink_bin(webrtc, new_pad);
 	RET_IF(ret != WEBRTC_ERROR_NONE, "failed to __add_rendering_sink_bin()");
+
+	GENERATE_DOT(webrtc, "webrtcbin_%s", GST_PAD_NAME(new_pad));
 }
 
 static void __webrtcbin_no_more_pads_cb(GstElement *webrtcbin, gpointer user_data)