From: Sangchul Lee <sc11.lee@samsung.com>
Date: Mon, 1 Nov 2021 09:43:00 +0000 (+0900)
Subject: webrtc_internal: Add APIs to set/get RTP packet drop probability
X-Git-Tag: submit/tizen/20211109.062149~2
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F90%2F265890%2F7;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_internal: Add APIs to set/get RTP packet drop probability

Functions below are added.
 - webrtc_media_source_set_rtp_packet_drop_probability()
 - webrtc_media_source_get_rtp_packet_drop_probability()

[Version] 0.3.2
[Issue Type] API

Change-Id: I0dfbc2e39705c2d47485807bc0cc1c8ba1850d58
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_internal.h b/include/webrtc_internal.h
index 86098877..7fbdc98b 100644
--- a/include/webrtc_internal.h
+++ b/include/webrtc_internal.h
@@ -253,6 +253,41 @@ int webrtc_screen_source_set_crop(webrtc_h webrtc, unsigned int source_id, int x
  */
 int webrtc_screen_source_unset_crop(webrtc_h webrtc, unsigned int source_id);
 
+/**
+ * @internal
+ * @brief Sets the probability of RTP packet dropping.
+ * @since_tizen 7.0
+ * @param[in] webrtc       WebRTC handle
+ * @param[in] source_id    The file source id
+ * @param[in] probability  The probability to be dropped (from @c 0 to @c 1.0 = 100%)
+ * @return @c 0 on success,
+ *         otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE    Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @pre Add screen source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
+ * @see webrtc_media_source_get_rtp_packet_drop_probability()
+ */
+int webrtc_media_source_set_rtp_packet_drop_probability(webrtc_h webrtc, unsigned int source_id, float probability);
+
+/**
+ * @internal
+ * @brief Gets the probability of RTP packet dropping.
+ * @since_tizen 7.0
+ * @remarks The default value is 0.
+ * @param[in] webrtc       WebRTC handle
+ * @param[in] source_id    The file source id
+ * @param[out] probability  The probability to be dropped (from @c 0 to @c 1.0 = 100%)
+ * @return @c 0 on success,
+ *         otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE    Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @pre Add screen source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
+ * @see webrtc_media_source_set_rtp_packet_drop_probability()
+ */
+int webrtc_media_source_get_rtp_packet_drop_probability(webrtc_h webrtc, unsigned int source_id, float *probability);
+
 /**
  * @internal
  * @brief Creates a signaling server for private network.
diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 6d816374..3a4343b7 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -586,6 +586,8 @@ int _set_screen_source_crop(webrtc_s *webrtc, unsigned int source_id, int x, int
 int _unset_screen_source_crop(webrtc_s *webrtc, unsigned int source_id);
 bool _check_if_path_is_set_to_file_sources(webrtc_s *webrtc);
 int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h packet);
+int _set_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, float probability);
+int _get_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, float *probability);
 void _invoke_state_changed_cb(webrtc_s *webrtc, webrtc_state_e old, webrtc_state_e new);
 void _post_state_cb_in_idle(webrtc_s *webrtc, webrtc_state_e new_state);
 void _post_error_cb_in_idle(webrtc_s *webrtc, webrtc_error_e error);
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 70c71685..876bc2e3 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:    0.3.1
+Version:    0.3.2
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_internal.c b/src/webrtc_internal.c
index 69424587..c4625688 100644
--- a/src/webrtc_internal.c
+++ b/src/webrtc_internal.c
@@ -154,4 +154,33 @@ int webrtc_screen_source_unset_crop(webrtc_h webrtc, unsigned int source_id)
 
 	return _unset_screen_source_crop(_webrtc, source_id);
 }
+
+int webrtc_media_source_set_rtp_packet_drop_probability(webrtc_h webrtc, unsigned int source_id, float probability)
+{
+	webrtc_s *_webrtc = (webrtc_s*)webrtc;
+	g_autoptr(GMutexLocker) locker = NULL;
+
+	RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+	RET_VAL_IF(probability > 1.0, WEBRTC_ERROR_INVALID_PARAMETER, "probability > 1.0");
+	RET_VAL_IF(probability < 0, WEBRTC_ERROR_INVALID_PARAMETER, "probability < 0");
+
+	locker = g_mutex_locker_new(&_webrtc->mutex);
+
+	return _set_rtp_packet_drop_probability(webrtc, source_id, probability);
+}
+
+int webrtc_media_source_get_rtp_packet_drop_probability(webrtc_h webrtc, unsigned int source_id, float *probability)
+{
+	webrtc_s *_webrtc = (webrtc_s*)webrtc;
+	g_autoptr(GMutexLocker) locker = NULL;
+
+	RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+	RET_VAL_IF(probability == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "probability is NULL");
+
+	locker = g_mutex_locker_new(&_webrtc->mutex);
+
+	return _get_rtp_packet_drop_probability(webrtc, source_id, probability);
+}
 //LCOV_EXCL_STOP
\ No newline at end of file
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 32f4acfd..f9586369 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -4261,7 +4261,7 @@ int _set_screen_source_crop(webrtc_s *webrtc, unsigned int source_id, int x, int
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
-	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 	RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_SCREEN, WEBRTC_ERROR_INVALID_PARAMETER, "source type is not screen");
 	RET_VAL_IF(w == 0, WEBRTC_ERROR_INVALID_PARAMETER, "w is 0");
 	RET_VAL_IF(h == 0, WEBRTC_ERROR_INVALID_PARAMETER, "h is 0");
@@ -4320,7 +4320,7 @@ int _unset_screen_source_crop(webrtc_s *webrtc, unsigned int source_id)
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
-	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 	RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_SCREEN, WEBRTC_ERROR_INVALID_PARAMETER, "source type is not screen");
 
 	screen_source =  gst_bin_get_by_name(source->bin, DEFAULT_NAME_SCREENSRC);
@@ -4368,12 +4368,11 @@ int _gst_filesrc_pipeline_set_state(webrtc_s *webrtc, GstState state)
 
 int _set_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool looping)
 {
-	webrtc_gst_slot_s *source = NULL;
+	webrtc_gst_slot_s *source;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
-
-	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 	RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_FILE, WEBRTC_ERROR_INVALID_PARAMETER, "invalid source type [%d]", source->type);
 
 	source->filesrc_loop = looping;
@@ -4385,13 +4384,12 @@ int _set_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool looping)
 
 int _get_filesrc_looping(webrtc_s * webrtc, unsigned int source_id, bool *looping)
 {
-	webrtc_gst_slot_s *source = NULL;
+	webrtc_gst_slot_s *source;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
 	RET_VAL_IF(looping == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "looping is NULL");
-
-	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 	RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_FILE, WEBRTC_ERROR_INVALID_PARAMETER, "invalid source type [%d]", source->type);
 
 	*looping = source->filesrc_loop;
@@ -4400,4 +4398,79 @@ int _get_filesrc_looping(webrtc_s * webrtc, unsigned int source_id, bool *loopin
 
 	return WEBRTC_ERROR_NONE;
 }
+
+int _set_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, float probability)
+{
+	webrtc_gst_slot_s *source;
+	GstElement *netsim;
+	GstBin *bin;
+
+	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+	RET_VAL_IF(probability > 1.0, WEBRTC_ERROR_INVALID_PARAMETER, "probability > 1.0");
+	RET_VAL_IF(probability < 0, WEBRTC_ERROR_INVALID_PARAMETER, "probability < 0");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+	RET_VAL_IF(webrtc->ini.general.network_simulator == false, WEBRTC_ERROR_INVALID_OPERATION, "network simulator is disabled, please check the ini");
+	bin = (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) ? GST_BIN(source->filesrc_pipeline) : GST_BIN(source->bin);
+	RET_VAL_IF(bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
+
+	if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) {
+		int count = 0;
+		if ((netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR))) {
+			g_object_set(G_OBJECT(netsim), "drop-probability", probability, NULL);
+			LOG_INFO("webrtc[%p] source_id[%u] probability[%f] applies to [%s]", webrtc, source_id, probability, GST_ELEMENT_NAME(netsim));
+			count++;
+		}
+		if ((netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR))) {
+			g_object_set(G_OBJECT(netsim), "drop-probability", probability, NULL);
+			LOG_INFO("webrtc[%p] source_id[%u] probability[%f] applies to [%s]", webrtc, source_id, probability, GST_ELEMENT_NAME(netsim));
+			count++;
+		}
+		RET_VAL_IF(count == 0, WEBRTC_ERROR_INVALID_OPERATION, "could not find any element for network simulator");
+		return WEBRTC_ERROR_NONE;
+	}
+
+	if (!(netsim = __find_element_in_bin(bin, DEFAULT_ELEMENT_NETWORK_SIMULATOR))) {
+		LOG_ERROR("could not find any element for network simulator");
+		return WEBRTC_ERROR_INVALID_OPERATION;
+	}
+	g_object_set(G_OBJECT(netsim), "drop-probability", probability, NULL);
+
+	LOG_INFO("webrtc[%p] source_id[%u] probability[%f] applies to [%s]", webrtc, source_id, probability, GST_ELEMENT_NAME(netsim));
+
+	return WEBRTC_ERROR_NONE;
+}
+
+int _get_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, float *probability)
+{
+	webrtc_gst_slot_s *source;
+	GstElement *netsim;
+	GstBin *bin;
+
+	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+	RET_VAL_IF(probability == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "probability is NULL");
+	RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+	RET_VAL_IF(webrtc->ini.general.network_simulator == false, WEBRTC_ERROR_INVALID_OPERATION, "network simulator is disabled, please check the ini");
+	bin = (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) ? GST_BIN(source->filesrc_pipeline) : GST_BIN(source->bin);
+	RET_VAL_IF(bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
+
+	if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) {
+		if (!(netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR)) &&
+			!(netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR))) {
+			LOG_ERROR("could not find any element for network simulator");
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+	} else {
+		if (!(netsim = __find_element_in_bin(bin, DEFAULT_ELEMENT_NETWORK_SIMULATOR))) {
+			LOG_ERROR("could not find any element for network simulator");
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+	}
+
+	g_object_get(G_OBJECT(netsim), "drop-probability", (gfloat *)probability, NULL);
+	LOG_INFO("webrtc[%p] source_id[%u] probability[%f]", webrtc, source_id, *probability);
+
+	return WEBRTC_ERROR_NONE;
+}
 //LCOV_EXCL_STOP