webrtc_doc: Add description for statistics module

[Version] 0.3.75
[Issue Type] Documentation

Change-Id: I4a616a34500b5227ba33016a04b39ef950b9ca87
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/doc/webrtc_doc.h b/doc/webrtc_doc.h
index 98ffd05522fe2fc3e0fd4bc59a01b07224bdc9c2..16a8daf13f6a4cd3c8dd77187adf73a621947ea1 100644 (file)
--- a/doc/webrtc_doc.h
+++ b/doc/webrtc_doc.h
@@ -35,7 +35,7 @@
  * @section CAPI_MEDIA_WEBRTC_OVERVIEW Overview
  * The WebRTC API provides functions to communicate with the remote peer using multimedia sources and generic data.
  * The multimedia sources include audio/video stream from microphone, camera or media file.
- * The generic data includes string or byte data.
+ * The generic data includes string or byte data.\n
  * This API set allows you to:
  *  - add/remove/mute/pause the media source
  *  - create/destroy data channel and send/receive data via the channel
@@ -46,7 +46,7 @@
  *  - get notified about various changes via callbacks
  *
  * @subsection CAPI_MEDIA_WEBRTC_LIFE_CYCLE_STATE_TRANSITIONS State Transitions
- * <div><table class="doxtable" >
+ * <div><table class="doxtable">
  * <tr>
  *    <th><b>FUNCTION</b></th>
  *    <th><b>PRE-STATE</b></th>
@@ -101,7 +101,7 @@
  *
  * @subsection CAPI_MEDIA_WEBRTC_LIFE_CYCLE_CALLBACK_OPERATIONS Callback(Event) Operations
  * The callback mechanism is used to notify the application about significant webrtc events.
- * <div><table class="doxtable" >
+ * <div><table class="doxtable">
  *     <tr>
  *        <th><b>REGISTER</b></th>
  *        <th><b>UNREGISTER</b></th>
@@ -150,21 +150,21 @@
  *        <td>webrtc_ice_connection_state_change_cb()</td>
  *        <td>This callback is used to notify that the WebRTC ICE connection state has changed</td>
  *    </tr>
- *</table></div>
+ * </table></div>
  *
  * @subsection CAPI_MEDIA_WEBRTC_FOREACH_OPERATIONS Foreach Operations
- * <div><table class="doxtable" >
+ * <div><table class="doxtable">
  *     <tr>
- *        <th><b>FOREACH</b></th>
- *        <th><b>CALLBACK</b></th>
- *        <th><b>DESCRIPTION</b></th>
+ *         <th><b>FOREACH</b></th>
+ *         <th><b>CALLBACK</b></th>
+ *         <th><b>DESCRIPTION</b></th>
  *     </tr>
  *     <tr>
- *        <td>webrtc_foreach_turn_server()</td>
- *        <td>webrtc_turn_server_cb()</td>
- *        <td>It retrieves the added TURN server one by one</td>
+ *         <td>webrtc_foreach_turn_server()</td>
+ *         <td>webrtc_turn_server_cb()</td>
+ *         <td>It retrieves the added TURN server one by one</td>
  *     </tr>
- *</table></div>
+ * </table></div>
  *
  * @section CAPI_MEDIA_WEBRTC_MODULE_FEATURE Related Features
  * This API is related with the following features:\n
@@ -237,4 +237,36 @@
  *
 */
 
+/**
+ * @ingroup CAPI_MEDIA_WEBRTC_MODULE
+ * @defgroup CAPI_MEDIA_WEBRTC_STATISTICS_MODULE Statistics
+ * @brief The @ref CAPI_MEDIA_WEBRTC_STATISTICS_MODULE API provides a function to retrieve all the statistics properties.
+ * @section CAPI_MEDIA_WEBRTC_STATISTICS_MODULE_HEADER Required Header
+ *    \#include <webrtc.h>
+ *
+ * @section CAPI_MEDIA_WEBRTC_STATISTICS_MODULE_OVERVIEW Overview
+ * The WebRTC Statistics API provides a function to get statistics information to monitor the status of the underlying network and pipeline.
+ * You can get various properties for each statistics type as below.
+ * - WEBRTC_STATS_TYPE_CODEC is for a codec currently being used by RTP streams
+ * - WEBRTC_STATS_TYPE_INBOUND_RTP is for RTP stream currently received
+ * - WEBRTC_STATS_TYPE_OUTBOUND_RTP is for RTP stream currently sent
+ * - WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP is for the remote peer's inbound RTP stream corresponding to an outbound RTP stream
+ * - WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP is for the remote peer's outbound RTP stream corresponding to an inbound RTP stream
+ *
+ * @subsection CAPI_MEDIA_WEBRTC_STATISTICS_FOREACH_OPERATIONS Foreach Operations
+ * <div><table class="doxtable">
+ *     <tr>
+ *         <th><b>FOREACH</b></th>
+ *         <th><b>CALLBACK</b></th>
+ *         <th><b>DESCRIPTION</b></th>
+ *     </tr>
+ *     <tr>
+ *         <td>webrtc_foreach_stats()</td>
+ *         <td>webrtc_stats_cb()</td>
+ *         <td>It retrieves all the statistics properties</td>
+ *     </tr>
+ * </table></div>
+ *
+*/
+
 #endif /* __TIZEN_MEDIA_WEBRTC_DOC_H__ */

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 74421c2d07b8dc79fd16d5ec436d41aa90905885..1826b4da47376be0f2363764f93635693f0e216f 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:    0.3.74
+Version:    0.3.75
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0