webrtc_ini: Add new item to enable network simulator

The network simulator element will be imported when the item
is set to 'yes' in the ini file.

This element is added right after payload of a source bin.
Dropping packets can be simulated by calling new internal API
coming up next patch.

Missing g_value_unset() is added.

[Version] 0.3.1
[Issue Type] New feature

Change-Id: Ia9231c68c19f6bec6bafb404b4d269c5ae3ab5d8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 37da9ce4140c7d32ed5126e8ac78ee8e0cd48905..6d816374c3c34b320fb53462da02ccefbae4686e 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -288,6 +288,7 @@ typedef     struct _ini_item_general_s {
        int stats_log_period;
        bool verbose_log;
        bool nice_verbose;
+       bool network_simulator;
        gchar **gst_args;
        gchar **gst_excluded_elements;
        const char *stun_server;

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 1ba56a99cdfd13f96a7b5135e77417023b455eee..70c7168546f29d05e5e951783f2c9a6d7e4d1583 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:    0.2.147
+Version:    0.3.1
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_ini.c b/src/webrtc_ini.c
index efc778f19687cd6cc2852a6ef6ff266e5e239651..3045eee0f0eea573777037aae14db0dbc734a0b7 100644 (file)
--- a/src/webrtc_ini.c
+++ b/src/webrtc_ini.c
@@ -25,6 +25,7 @@ bool g_verbose = false;
 #define DEFAULT_STATS_LOG_PERIOD          0  /* sec */
 #define DEFAULT_VERBOSE_LOG               false
 #define DEFAULT_NICE_VERBOSE              false
+#define DEFAULT_NETWORK_SIMULATOR         false
 #define DEFAULT_JITTERBUFFER_LATENCY      200  /* ms */
 
 /* categories */
@@ -49,6 +50,7 @@ bool g_verbose = false;
 #define INI_ITEM_STATS_LOG_PERIOD         "stats log period"
 #define INI_ITEM_VERBOSE_LOG              "verbose log"
 #define INI_ITEM_NICE_VERBOSE             "nice verbose"
+#define INI_ITEM_NETWORK_SIMULATOR        "network simulator"
 #define INI_ITEM_GST_ARGS                 "gstreamer arguments"
 #define INI_ITEM_GST_EXCLUDED_ELEMENTS    "gstreamer excluded elements"
 #define INI_ITEM_STUN_SERVER              "stun server"
@@ -215,6 +217,7 @@ static void __dump_ini(webrtc_ini_s *ini)
        __dump_item(INI_ITEM_STATS_LOG_PERIOD, INI_ITEM_TYPE_INT, &ini->general.stats_log_period);
        __dump_item(INI_ITEM_VERBOSE_LOG, INI_ITEM_TYPE_BOOL, &ini->general.verbose_log);
        __dump_item(INI_ITEM_NICE_VERBOSE, INI_ITEM_TYPE_BOOL, &ini->general.nice_verbose);
+       __dump_item(INI_ITEM_NETWORK_SIMULATOR, INI_ITEM_TYPE_BOOL, &ini->general.network_simulator);
        __dump_item(INI_ITEM_GST_ARGS, INI_ITEM_TYPE_STRINGS, ini->general.gst_args);
        __dump_item(INI_ITEM_GST_EXCLUDED_ELEMENTS, INI_ITEM_TYPE_STRINGS, ini->general.gst_excluded_elements);
        __dump_item(INI_ITEM_STUN_SERVER, INI_ITEM_TYPE_STRING, (void *)ini->general.stun_server);
@@ -450,6 +453,7 @@ int _load_ini(webrtc_s *webrtc)
        ini->general.nice_verbose = __ini_get_boolean(ini->dict, INI_CATEGORY_GENERAL, INI_ITEM_NICE_VERBOSE, DEFAULT_NICE_VERBOSE);
        if (ini->general.nice_verbose)
                g_setenv("NICE_DEBUG", "nice-verbose", TRUE);
+       ini->general.network_simulator = __ini_get_boolean(ini->dict, INI_CATEGORY_GENERAL, INI_ITEM_NETWORK_SIMULATOR, DEFAULT_NETWORK_SIMULATOR);
        __ini_read_list(ini->dict, INI_CATEGORY_GENERAL, INI_ITEM_GST_ARGS, &ini->general.gst_args);
        __ini_read_list(ini->dict, INI_CATEGORY_GENERAL, INI_ITEM_GST_EXCLUDED_ELEMENTS, &ini->general.gst_excluded_elements);
        ini->general.stun_server = __ini_get_string(ini->dict, INI_CATEGORY_GENERAL, INI_ITEM_STUN_SERVER, NULL);

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 63f6312d5d27681abe4449089b8a0ab29581b325..32f4acfd1569043d30b31739ec264e13312fff44 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -28,39 +28,41 @@
 #define GST_KLASS_NAME_CONVERTER_AUDIO "Filter/Converter/Audio"
 #define GST_KLASS_NAME_CONVERTER_VIDEO "Filter/Converter/Video"
 
-#define DEFAULT_ELEMENT_CAMERASRC      "v4l2src"
-#define DEFAULT_ELEMENT_AUDIOSRC       "pulsesrc"
-#define DEFAULT_ELEMENT_VIDEOTESTSRC   "videotestsrc"
-#define DEFAULT_ELEMENT_AUDIOTESTSRC   "audiotestsrc"
-#define DEFAULT_ELEMENT_APPSRC         "appsrc"
-#define DEFAULT_ELEMENT_SCREENSRC      "waylandsrc"
-#define DEFAULT_ELEMENT_QUEUE          "queue"
-#define DEFAULT_ELEMENT_VOLUME         "volume"
-#define DEFAULT_ELEMENT_INPUT_SELECTOR "input-selector"
-#define DEFAULT_ELEMENT_VIDEOCROP      "videocrop"
-#define DEFAULT_ELEMENT_FILESRC        "filesrc"
-
-
-#define ELEMENT_NAME_FIRST_CAPSFILTER "firstCapsfilter"
-#define ELEMENT_NAME_RTP_CAPSFILTER   "rtpCapsfilter"
-#define ELEMENT_NAME_VIDEO_SRC        "videoSrc"
-#define ELEMENT_NAME_VIDEO_SWITCH     "videoSwitch"
-#define ELEMENT_NAME_VIDEO_MUTE_SRC   "videoMuteSrc"
-#define ELEMENT_NAME_VOLUME           "volume"
-#define ELEMENT_NAME_MIC_SRC          "micSrc"
-#define DEFAULT_NAME_FILE_SRC         "fileSrc"
-#define DEFAULT_NAME_AUDIO_QUEUE      "audioQueue"
-#define DEFAULT_NAME_VIDEO_QUEUE      "videoQueue"
-#define DEFAULT_NAME_AUDIO_CAPSFILTER "audioCapsfilter"
-#define DEFAULT_NAME_VIDEO_CAPSFILTER "videoCapsfilter"
-#define DEFAULT_NAME_AUDIO_PAYLOAD    "audioPayload"
-#define DEFAULT_NAME_VIDEO_PAYLOAD    "videoPayload"
-#define DEFAULT_NAME_VIDEOCROP        "videoCrop"
-#define DEFAULT_NAME_SCREENSRC        "waylandSrc"
-#define DEFAULT_NAME_AUDIO_FAKESINK   "audioFakeSink"
-#define DEFAULT_NAME_VIDEO_FAKESINK   "videoFakeSink"
-#define DEFAULT_NAME_AUDIO_APPSRC     "audioAppsrc"
-#define DEFAULT_NAME_VIDEO_APPSRC     "videoAppsrc"
+#define DEFAULT_ELEMENT_CAMERASRC         "v4l2src"
+#define DEFAULT_ELEMENT_AUDIOSRC          "pulsesrc"
+#define DEFAULT_ELEMENT_VIDEOTESTSRC      "videotestsrc"
+#define DEFAULT_ELEMENT_AUDIOTESTSRC      "audiotestsrc"
+#define DEFAULT_ELEMENT_APPSRC            "appsrc"
+#define DEFAULT_ELEMENT_SCREENSRC         "waylandsrc"
+#define DEFAULT_ELEMENT_QUEUE             "queue"
+#define DEFAULT_ELEMENT_VOLUME            "volume"
+#define DEFAULT_ELEMENT_INPUT_SELECTOR    "input-selector"
+#define DEFAULT_ELEMENT_VIDEOCROP         "videocrop"
+#define DEFAULT_ELEMENT_FILESRC           "filesrc"
+#define DEFAULT_ELEMENT_NETWORK_SIMULATOR "netsim"
+
+#define ELEMENT_NAME_FIRST_CAPSFILTER        "firstCapsfilter"
+#define ELEMENT_NAME_RTP_CAPSFILTER          "rtpCapsfilter"
+#define ELEMENT_NAME_VIDEO_SRC               "videoSrc"
+#define ELEMENT_NAME_VIDEO_SWITCH            "videoSwitch"
+#define ELEMENT_NAME_VIDEO_MUTE_SRC          "videoMuteSrc"
+#define ELEMENT_NAME_VOLUME                  "volume"
+#define ELEMENT_NAME_MIC_SRC                 "micSrc"
+#define DEFAULT_NAME_FILE_SRC                "fileSrc"
+#define DEFAULT_NAME_AUDIO_QUEUE             "audioQueue"
+#define DEFAULT_NAME_VIDEO_QUEUE             "videoQueue"
+#define DEFAULT_NAME_AUDIO_CAPSFILTER        "audioCapsfilter"
+#define DEFAULT_NAME_VIDEO_CAPSFILTER        "videoCapsfilter"
+#define DEFAULT_NAME_AUDIO_PAYLOAD           "audioPayload"
+#define DEFAULT_NAME_VIDEO_PAYLOAD           "videoPayload"
+#define DEFAULT_NAME_VIDEOCROP               "videoCrop"
+#define DEFAULT_NAME_SCREENSRC               "waylandSrc"
+#define DEFAULT_NAME_AUDIO_FAKESINK          "audioFakeSink"
+#define DEFAULT_NAME_VIDEO_FAKESINK          "videoFakeSink"
+#define DEFAULT_NAME_AUDIO_APPSRC            "audioAppsrc"
+#define DEFAULT_NAME_VIDEO_APPSRC            "videoAppsrc"
+#define DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR "audioNetSim"
+#define DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR "videoNetSim"
 
 #define APPEND_ELEMENT(x_list, x_element) \
 do { \
@@ -109,6 +111,7 @@ typedef struct {
        const char *payload_name;
        const char *capsfilter_name;
        const char *fakesink_name;
+       const char *network_simulator_name;
 } av_mapping_table_s;
 
 static av_mapping_table_s _av_tbl[AV_IDX_MAX] = {
@@ -118,6 +121,7 @@ static av_mapping_table_s _av_tbl[AV_IDX_MAX] = {
                DEFAULT_NAME_AUDIO_PAYLOAD,
                DEFAULT_NAME_AUDIO_CAPSFILTER,
                DEFAULT_NAME_AUDIO_FAKESINK,
+               DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR
        },
        {
                DEFAULT_NAME_VIDEO_APPSRC,
@@ -125,6 +129,7 @@ static av_mapping_table_s _av_tbl[AV_IDX_MAX] = {
                DEFAULT_NAME_VIDEO_PAYLOAD,
                DEFAULT_NAME_VIDEO_CAPSFILTER,
                DEFAULT_NAME_VIDEO_FAKESINK,
+               DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR
        }
 };
 
@@ -850,7 +855,6 @@ static void __remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsign
 
 static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool need_capsfilter, GList **element_list, bool is_audio)
 {
-       GstElement *capsfilter = NULL;
        GstElement *encoder = NULL;
        GstElement *payloader;
        GstElement *queue;
@@ -873,7 +877,8 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
        idx = GET_AV_IDX(is_audio);
 
        if (need_capsfilter) {
-               if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_FIRST_CAPSFILTER)))
+               GstElement *capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_FIRST_CAPSFILTER);
+               if (!capsfilter)
                        return WEBRTC_ERROR_INVALID_OPERATION;
                APPEND_ELEMENT(*element_list, capsfilter);
 
@@ -942,6 +947,13 @@ skip_encoder:
                goto error;
        APPEND_ELEMENT(*element_list, payloader);
 
+       if (webrtc->ini.general.network_simulator) {
+               GstElement *netsim = _create_element(DEFAULT_ELEMENT_NETWORK_SIMULATOR, NULL);
+               if (!netsim)
+                       goto error;
+               APPEND_ELEMENT(*element_list, netsim);
+       }
+
        if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, NULL)))
                goto error;
        APPEND_ELEMENT(*element_list, queue);
@@ -992,29 +1004,29 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
                                                __make_encoded_caps_from_media_format(source, &media_type),
                                                NULL,
                                                payloader);
-       if (payloader == NULL) {
-               g_free(media_type);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+       if (!payloader)
+               goto error;
        APPEND_ELEMENT(*element_list, payloader);
 
-       if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, NULL))) {
-               g_free(media_type);
-               return WEBRTC_ERROR_INVALID_OPERATION;
+       if (webrtc->ini.general.network_simulator) {
+               GstElement *netsim = _create_element(DEFAULT_ELEMENT_NETWORK_SIMULATOR, NULL);
+               if (!netsim)
+                       goto error;
+               APPEND_ELEMENT(*element_list, netsim);
        }
+
+       if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, NULL)))
+               goto error;
        APPEND_ELEMENT(*element_list, queue);
 
-       if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_RTP_CAPSFILTER))) {
-               g_free(media_type);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+       if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_RTP_CAPSFILTER)))
+               goto error;
        APPEND_ELEMENT(*element_list, capsfilter);
 
        payload_id = __get_available_payload_id(webrtc);
-       if (payload_id == 0) {
-               g_free(media_type);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+       if (payload_id == 0)
+               goto error;
+
        source->av[GET_AV_IDX_BY_TYPE(source->media_types)].payload_id = payload_id;
 
        if ((sink_caps = __make_rtp_caps(media_type, payload_id))) {
@@ -1023,8 +1035,11 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
        }
 
        g_free(media_type);
-
        return WEBRTC_ERROR_NONE;
+
+error:
+       g_free(media_type);
+       return WEBRTC_ERROR_INVALID_OPERATION;
 }
 //LCOV_EXCL_STOP
 
@@ -1263,13 +1278,14 @@ static GstElement *__find_element_in_bin(GstBin *bin, const gchar *name)
 
                if (g_strrstr(GST_ELEMENT_NAME(element), name)) {
                        LOG_DEBUG("found element by name [%s]", GST_ELEMENT_NAME(element));
+                       g_value_unset(&value);
                        gst_iterator_free(bin_iterator);
                        return element;
                }
 
                g_value_reset(&value);
        }
-
+       g_value_unset(&value);
        gst_iterator_free(bin_iterator);
 
        return NULL;
@@ -1762,14 +1778,17 @@ exit:
 
 static void __remove_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
 {
-       GstBin *bin = NULL;
-       GstElement *queue = NULL;
-       GstElement *payload = NULL;
-       GstElement *capsfilter = NULL;
-       GstElement *fakesink = NULL;
+       GstBin *bin;
+       GstElement *queue;
+       GstElement *payload;
+       GstElement *capsfilter;
+       GstElement *fakesink;
+       GstElement *netsim;
        int av_idx = GET_AV_IDX(is_audio);
 
        RET_IF(source == NULL, "pad is NULL");
+       RET_IF(source->webrtc == NULL, "webrtc is NULL");
+       RET_IF(source->filesrc_pipeline == NULL, "filesrc_pipeline is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
 
@@ -1786,6 +1805,12 @@ static void __remove_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *so
        RET_IF(fakesink == NULL, "fakesink is NULL");
 
        gst_bin_remove_many(bin, queue, payload, capsfilter, fakesink, NULL);
+
+       if (source->webrtc->ini.general.network_simulator) {
+               netsim = gst_bin_get_by_name(bin, _av_tbl[av_idx].network_simulator_name);
+               RET_IF(netsim == NULL, "netsim is NULL");
+               gst_bin_remove(bin, netsim);
+       }
 }
 
 static void __filesrc_pipeline_audio_stream_handoff_cb(GstElement *object, GstBuffer *buffer, GstPad *pad, gpointer data)
@@ -1908,14 +1933,16 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
 
 static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstPad *pad, bool is_audio)
 {
-       GstBin *bin = NULL;
-       GstElement *queue = NULL;
-       GstElement *payload = NULL;
-       GstElement *capsfilter = NULL;
-       GstElement *fakesink = NULL;
+       GstBin *bin;
+       GstElement *queue;
+       GstElement *payload;
+       GstElement *netsim;
+       GstElement *capsfilter;
+       GstElement *fakesink;
        GList *element_list = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+       RET_VAL_IF(source->webrtc == NULL, WEBRTC_ERROR_INVALID_OPERATION, "webrtc is NULL");
        RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
@@ -1928,6 +1955,12 @@ static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *sou
                goto exit;
        APPEND_ELEMENT(element_list, payload);
 
+       if (source->webrtc->ini.general.network_simulator) {
+               if (!(netsim = _create_element(DEFAULT_ELEMENT_NETWORK_SIMULATOR, _av_tbl[GET_AV_IDX(is_audio)].network_simulator_name)))
+                       goto exit;
+               APPEND_ELEMENT(element_list, netsim);
+       }
+
        if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio)))
                goto exit;
        APPEND_ELEMENT(element_list, capsfilter);