webrtc_source: add queue after the decodebin in the filesrc pipeline

[Version] 0.2.119
[Issue Type] Improvement

Previously, a probe for loopback support was attached to decodebin's pad,
but decodebin deletes all pads when state goes to NULL.
The loopback setting must be maintained until the user unset it.
So, a queue was added after the decodebin and support loopback.

Change-Id: Ia1aaba9fa3b5b995161216294dcb49b6930c8624

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 0ffebebe33a4abd90c3ce5d8ba4267cb2edbfd03..f3d572d664deedb9b7567403b69bfb35c5d7533f 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:    0.2.118
+Version:    0.2.119
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index f8fd51846ae1094767d508382557e820c1959d2d..a86355eb7d5597b8850804f4c8a7ffd99a53e8c1 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -38,6 +38,8 @@
 #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"
@@ -1208,6 +1210,19 @@ static bool __link_elements(GList *element_list)
        return true;
 }
 
+static bool __sync_elements_state_with_parent(GList *element_list)
+{
+       GList *list;
+
+       RET_VAL_IF(element_list == NULL, false, "element_list is NULL");
+
+       for (list = element_list; list; list = g_list_next(list))
+               RET_VAL_IF(!gst_element_sync_state_with_parent((GstElement *)list->data), false,
+                       "failed to gst_element_sync_state_with_parent(), [%s]", GST_ELEMENT_NAME((GstElement *)list->data));
+
+       return true;
+}
+
 static bool __link_switch_srcs(GstElement *switch_element, GList *switch_src_list)
 {
        GstElement *element;
@@ -1679,7 +1694,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
        GstElement *appsrc = NULL;
        GstElement *queue = NULL;
        GstElement *capsfilter = NULL;
-       int idx = GET_AV_IDX_BY_TYPE(media_type);
+       int av_idx = GET_AV_IDX_BY_TYPE(media_type);
 
        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");
@@ -1689,7 +1704,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
        ret = _add_no_target_ghostpad_to_slot(source, true, &src_pad);
        RET_VAL_IF(ret != WEBRTC_ERROR_NONE, ret, "failed to _add_no_target_ghostpad_to_slot()");
 
-       if (!(appsrc = _create_element("appsrc", _av_tbl[idx].appsrc_name)))
+       if (!(appsrc = _create_element(DEFAULT_ELEMENT_APPSRC, _av_tbl[av_idx].appsrc_name)))
                return WEBRTC_ERROR_INVALID_OPERATION;
 
        g_object_set(G_OBJECT(appsrc),
@@ -1697,12 +1712,12 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
                        "format", GST_FORMAT_TIME,
                        NULL);
 
-       if (!(queue = _create_element("queue", _av_tbl[idx].queue_name))) {
+       if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, _av_tbl[av_idx].queue_name))) {
                SAFE_GST_OBJECT_UNREF(appsrc);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!(capsfilter = _create_element("capsfilter", _av_tbl[idx].capsfilter_name))) {
+       if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, _av_tbl[av_idx].capsfilter_name))) {
                SAFE_GST_OBJECT_UNREF(appsrc);
                SAFE_GST_OBJECT_UNREF(queue);
                return WEBRTC_ERROR_INVALID_OPERATION;
@@ -1734,7 +1749,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
        if (ret != WEBRTC_ERROR_NONE)
                goto error;
 
-       __add_probe_to_pad_for_pause(source, idx, src_pad, __payloaded_data_probe_cb);
+       __add_probe_to_pad_for_pause(source, av_idx, src_pad, __payloaded_data_probe_cb);
 
        return WEBRTC_ERROR_NONE;
 
@@ -1746,25 +1761,29 @@ error:
 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;
-       int idx = GET_AV_IDX(is_audio);
+       int av_idx = GET_AV_IDX(is_audio);
 
        RET_IF(source == NULL, "pad is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
 
-       payload = gst_bin_get_by_name(bin, _av_tbl[idx].payload_name);
+       queue = gst_bin_get_by_name(bin, _av_tbl[av_idx].queue_name);
+       RET_IF(queue == NULL, "queue is NULL");
+
+       payload = gst_bin_get_by_name(bin, _av_tbl[av_idx].payload_name);
        RET_IF(payload == NULL, "payload is NULL");
 
-       capsfilter = gst_bin_get_by_name(bin, _av_tbl[idx].capsfilter_name);
+       capsfilter = gst_bin_get_by_name(bin, _av_tbl[av_idx].capsfilter_name);
        RET_IF(capsfilter == NULL, "capsfilter is NULL");
 
-       fakesink = gst_bin_get_by_name(bin, _av_tbl[idx].fakesink_name);
+       fakesink = gst_bin_get_by_name(bin, _av_tbl[av_idx].fakesink_name);
        RET_IF(fakesink == NULL, "fakesink is NULL");
 
-       gst_bin_remove_many(bin, payload, capsfilter, fakesink, NULL);
+       gst_bin_remove_many(bin, queue, payload, capsfilter, fakesink, NULL);
 }
 
 static void __filesrc_pipeline_audio_stream_handoff_cb(GstElement *object, GstBuffer *buffer, GstPad *pad, gpointer data)
@@ -1865,7 +1884,7 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
 
        RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
-       if (!(fakesink = _create_element("fakesink", _av_tbl[GET_AV_IDX(is_audio)].fakesink_name)))
+       if (!(fakesink = _create_element(DEFAULT_ELEMENT_FAKESINK, _av_tbl[GET_AV_IDX(is_audio)].fakesink_name)))
                return NULL;
 
        sink_pad = gst_element_get_static_pad(fakesink, "sink");
@@ -1888,74 +1907,74 @@ 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;
+       GList *element_list = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
        RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
 
-       if (!(payload = __create_payload_for_filesrc_pipeline(pad, is_audio)))
+       if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, _av_tbl[GET_AV_IDX(is_audio)].queue_name)))
                return WEBRTC_ERROR_INVALID_OPERATION;
+       APPEND_ELEMENT(element_list, queue);
 
-       if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio))) {
-               SAFE_GST_OBJECT_UNREF(payload);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
-       if (!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, is_audio))) {
-               SAFE_GST_OBJECT_UNREF(payload);
-               SAFE_GST_OBJECT_UNREF(capsfilter);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
+       if (!(payload = __create_payload_for_filesrc_pipeline(pad, is_audio)))
+               goto exit;
+       APPEND_ELEMENT(element_list, payload);
 
-       gst_bin_add_many(bin, payload, capsfilter, fakesink, NULL);
+       if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio)))
+               goto exit;
+       APPEND_ELEMENT(element_list, capsfilter);
 
-       if (!gst_element_link_many(payload, capsfilter, fakesink, NULL)) {
-               LOG_ERROR("failed to gst_element_link_many()");
-               goto error;
-       }
+       if (!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, is_audio)))
+               goto exit;
+       APPEND_ELEMENT(element_list, fakesink);
 
-       if (!gst_element_sync_state_with_parent(payload)) {
-               LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(payload));
-               goto error;
-       }
+       if (!__add_elements_to_bin(bin, element_list))
+               goto exit;
 
-       if (!gst_element_sync_state_with_parent(capsfilter)) {
-               LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(capsfilter));
-               goto error;
-       }
+       if (!__link_elements(element_list))
+               goto exit_with_remove_from_bin;
 
-       if (!gst_element_sync_state_with_parent(fakesink)) {
-               LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(fakesink));
-               goto error;
-       }
+       if (!__sync_elements_state_with_parent(element_list))
+               goto exit_with_remove_from_bin;
+
+       SAFE_G_LIST_FREE(element_list);
 
        return WEBRTC_ERROR_NONE;
 
-error:
-       gst_bin_remove_many(bin, payload, capsfilter, fakesink, NULL);
+exit_with_remove_from_bin:
+       /* elements will be dereferenced */
+       __remove_elements_from_bin(bin, element_list);
+       SAFE_G_LIST_FREE(element_list);
+       return WEBRTC_ERROR_INVALID_OPERATION;
+exit:
+       __unref_elements(element_list);
+       SAFE_G_LIST_FREE(element_list);
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
-static int __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source, bool is_audio)
+static int __link_decodebin_with_queue(GstPad *pad, webrtc_gst_slot_s *source, bool is_audio)
 {
-       GstElement *payload = NULL;
+       GstElement *queue = NULL;
        GstPad *sink_pad = NULL;
 
        RET_VAL_IF(pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pad is NULL");
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 
-       payload = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), _av_tbl[GET_AV_IDX(is_audio)].payload_name);
-       if (!payload) {
-               LOG_ERROR("failed to get element [%s]", _av_tbl[GET_AV_IDX(is_audio)].payload_name);
+       queue = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), _av_tbl[GET_AV_IDX(is_audio)].queue_name);
+       if (!queue) {
+               LOG_ERROR("failed to get element [%s]", _av_tbl[GET_AV_IDX(is_audio)].queue_name);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       sink_pad = gst_element_get_static_pad(payload, "sink");
+       sink_pad = gst_element_get_static_pad(queue, "sink");
        if (!sink_pad) {
-               LOG_ERROR("sink_pad is NULL for [%s]", GST_ELEMENT_NAME(payload));
+               LOG_ERROR("sink_pad is NULL for [%s]", GST_ELEMENT_NAME(queue));
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
@@ -1967,7 +1986,7 @@ static int __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source,
 
        g_object_unref(sink_pad);
 
-       LOG_INFO("decodebin is linked to [%s]", GST_ELEMENT_NAME(payload));
+       LOG_INFO("decodebin is linked to [%s]", GST_ELEMENT_NAME(queue));
 
        return WEBRTC_ERROR_NONE;
 }
@@ -1979,6 +1998,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        const gchar *media_type = NULL;
        gboolean is_audio;
        int av_idx;
+       GstElement *queue = NULL;
 
        RET_IF(source == NULL, "source is NULL");
        RET_IF(source->filesrc_pipeline == NULL, "filesrc_pipeline is NULL");
@@ -1998,7 +2018,9 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
 
        if (source->av[av_idx].src_pad_probe_id > 0) {
                LOG_INFO("Pipeline already built");
-               __link_decodebin_with_payload(pad, source, is_audio);
+               ret = __link_decodebin_with_queue(pad, source, is_audio);
+               if (ret != WEBRTC_ERROR_NONE)
+                       LOG_ERROR("failed to __link_decodebin_with_queue()");
                return;
        }
 
@@ -2008,9 +2030,12 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
                return;
        }
 
-       ret = __link_decodebin_with_payload(pad, source, is_audio);
+       queue = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), _av_tbl[GET_AV_IDX(is_audio)].queue_name);
+       RET_IF(queue == NULL, "queue is NULL");
+
+       ret = __link_decodebin_with_queue(pad, source, is_audio);
        if (ret != WEBRTC_ERROR_NONE) {
-               LOG_ERROR("failed to __link_decodebin_with_payload()");
+               LOG_ERROR("failed to __link_decodebin_with_queue()");
                __remove_rest_of_elements_for_filesrc_pipeline(source, is_audio);
                return;
        }
@@ -2032,7 +2057,7 @@ 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, pad, __source_data_probe_cb);
+       __add_probe_to_pad_for_render(source, av_idx, gst_element_get_static_pad(queue, "src"), __source_data_probe_cb);
 }
 
 static GstAutoplugSelectResult __filesrc_pipeline_decodebin_autoplug_select_cb(GstElement *bin, GstPad *pad, GstCaps *caps, GstElementFactory* factory, gpointer udata)
@@ -2047,29 +2072,6 @@ static GstAutoplugSelectResult __filesrc_pipeline_decodebin_autoplug_select_cb(G
        return GST_AUTOPLUG_SELECT_TRY;
 }
 
-static void __filesrc_pipeline_decodebin_pad_removed_cb(GstElement *element, GstPad *pad, gpointer data)
-{
-       webrtc_gst_slot_s *source = data;
-       int idx = 0;
-
-       RET_IF(source == NULL, "source is NULL");
-
-       if (GST_PAD_DIRECTION(pad) != GST_PAD_SRC)
-               return;
-
-       LOG_INFO("[%s] removed_pad[%s]", GST_ELEMENT_NAME(element), GST_PAD_NAME(pad));
-
-       for (idx = 0; idx < AV_IDX_MAX; idx++) {
-               if (source->av[idx].render.src_pad_probe_id > 0) {
-                       source->av[idx].render.src_pad_probe_id = 0;
-                       source->av[idx].render.src_pad = NULL;
-
-                       gst_caps_unref(source->av[idx].render.appsrc_caps);
-                       source->av[idx].render.appsrc_caps = NULL;
-               }
-       }
-}
-
 static gboolean __filesrc_pipeline_bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_data)
 {
        webrtc_gst_slot_s *source = (webrtc_gst_slot_s *)user_data;
@@ -2176,7 +2178,7 @@ static int __build_filesrc_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source)
                goto error;
        }
 
-       if (!(filesrc = _create_element("filesrc", DEFAULT_NAME_FILE_SRC)))
+       if (!(filesrc = _create_element(DEFAULT_ELEMENT_FILESRC, DEFAULT_NAME_FILE_SRC)))
                goto error;
 
        if (!(decodebin = _create_element("decodebin", NULL))) {
@@ -2194,7 +2196,6 @@ static int __build_filesrc_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 
        g_signal_connect(decodebin, "autoplug-select", G_CALLBACK(__filesrc_pipeline_decodebin_autoplug_select_cb), NULL);
        g_signal_connect(decodebin, "pad-added", G_CALLBACK(__filesrc_pipeline_decodebin_pad_added_cb), (gpointer)source);
-       g_signal_connect(decodebin, "pad-removed", G_CALLBACK(__filesrc_pipeline_decodebin_pad_removed_cb), (gpointer)source);
 
        return WEBRTC_ERROR_NONE;
 
@@ -3905,7 +3906,7 @@ static int __build_loopback_render_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s
        g_free(pipeline_name);
        RET_VAL_IF(source->av[idx].render.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "render_pipeline is NULL");
 
-       appsrc = _create_element("appsrc", NULL);
+       appsrc = _create_element(DEFAULT_ELEMENT_APPSRC, NULL);
        if (!appsrc)
                goto error;
 
@@ -4233,7 +4234,7 @@ static void __set_filesrc_pipline_state_foreach_cb(gpointer key, gpointer value,
                return;
        }
 
-       LOG_INFO("change pipeline state to [%s]", gst_element_state_get_name(state));
+       LOG_INFO("change filesrc pipeline state to [%s]", gst_element_state_get_name(state));
 }
 
 int _gst_filesrc_pipeline_set_state(webrtc_s *webrtc, GstState state)