Blocking filesrc streaming until the connection with peer is completed

Common streaming sending is blocked while connecting with peer.
but until now, filesrc has not been blocked because filesrc constructs it's own pipeline.

[Version] 0.3.12
[Issue Type] Improvement

Change-Id: I18c45e075b5b1db6e78822d7af2ee6a0148a8d8b

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index e2e1e4b66d0b7ac5381b5f8c205df0a9641eb688..0b0ff4d2fc2e397fe9d84a5ddd6db7d5c666c3aa 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -494,6 +494,10 @@ typedef struct _webrtc_gst_slot_s {
        GstBus *filesrc_bus;
        guint filesrc_bus_watcher;
        bool filesrc_loop;
+       struct {
+               GstPad *sink_pad;
+               gulong sink_pad_probe_id;
+       } filesrc_av[AV_IDX_MAX];
 
        webrtc_display_s *display;
 } webrtc_gst_slot_s;
@@ -683,6 +687,7 @@ int _check_feature(const char *feature);
 int _gst_filesrc_pipeline_set_state(webrtc_s *webrtc, GstState state);
 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);
+int _remove_filesrc_pad_block_probe(webrtc_s *webrtc);
 
 #ifdef __cplusplus
 }

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 0aa12413c53d6ee2915fa5a3c5ccda85bbc82c1b..c3f6fd50202436268995b715e62188e2e4309537 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:    0.3.11
+Version:    0.3.12
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 619721f4196a643a7504b4b8c74ac257a1ac3a0b..ec20681a88bd69ac09420b9ed9e521b63fd8e19b 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -228,8 +228,10 @@ void _invoke_state_changed_cb(webrtc_s *webrtc, webrtc_state_e old, webrtc_state
                LOG_DEBUG("<<< end of the callback");
        }
 
-       if (new == WEBRTC_STATE_PLAYING)
+       if (new == WEBRTC_STATE_PLAYING) {
                _set_stats_timer(webrtc);
+               _remove_filesrc_pad_block_probe(webrtc);
+       }
 
        GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s.state_%s", GST_ELEMENT_NAME(webrtc->gst.pipeline), __state_str[webrtc->state]);
 }

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 16c86889fe6ddb41ef8d3d005a6ee6a41dd9edb7..84f4d8c5ba26d08d0bf1e4e8e929a80efbd2245e 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1850,20 +1850,36 @@ static void __filesrc_pipeline_video_stream_handoff_cb(GstElement *object, GstBu
                LOG_ERROR("failed to 'push-buffer', gst_ret[0x%x]", gst_ret);
 }
 
+static GstPadProbeReturn __fakesink_block_probe_cb(GstPad *pad,  GstPadProbeInfo *info, gpointer u_data)
+{
+       webrtc_gst_slot_s *source = u_data;
+       gchar *media = NULL;
+
+       gst_structure_get(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0), "media", G_TYPE_STRING, &media, NULL);
+
+       LOG_DEBUG("source[%p, id:%u] fakesink pad[%p] for [%s] is blocked", source, source->id, pad, media);
+
+       return GST_PAD_PROBE_OK;
+}
+
 static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info, gpointer u_data)
 {
        webrtc_gst_slot_s *source = u_data;
        GstCaps *new_cap = NULL;
        GstElement *appsrc = NULL;
        gchar *media = NULL;
+       int av_idx;
 
        gst_structure_get(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0), "media", G_TYPE_STRING, &media, NULL);
 
-       if (g_strrstr(media, "audio"))
-               appsrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_AUDIO_APPSRC);
-       else
-               appsrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_APPSRC);
+       if (!g_strrstr(media, "audio") && !g_strrstr(media, "video")) {
+               LOG_ERROR("not supported media type [%s]", media);
+               return GST_PAD_PROBE_OK;
+       }
 
+       av_idx = GET_AV_IDX(g_strrstr(media, "audio"));
+
+       appsrc = gst_bin_get_by_name(source->bin, _av_tbl[av_idx].appsrc_name);
        RET_VAL_IF(appsrc == NULL, GST_PAD_PROBE_OK, "There is no appsrc for [%s]", media);
 
        new_cap = gst_caps_copy(gst_pad_get_current_caps(pad));
@@ -1872,6 +1888,10 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info
        LOG_INFO("setting caps for [%s appsrc] successfully", media);
        PRINT_CAPS(new_cap, "appsrc");
 
+       source->filesrc_av[av_idx].sink_pad = pad;
+       source->filesrc_av[av_idx].sink_pad_probe_id = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BLOCK,
+                                       __fakesink_block_probe_cb, source, NULL);
+
        return GST_PAD_PROBE_REMOVE;
 }
 
@@ -2062,7 +2082,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        LOG_INFO("[%s] new_pad[%s] media_type[%s]", GST_ELEMENT_NAME(element), GST_PAD_NAME(pad), media_type);
 
        if (!g_strrstr(media_type, "audio") && !g_strrstr(media_type, "video")) {
-               LOG_ERROR("invalid media type [%s]", media_type);
+               LOG_ERROR("not supported media type [%s]", media_type);
                return;
        }
 
@@ -4554,4 +4574,33 @@ int _get_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, f
 
        return WEBRTC_ERROR_NONE;
 }
+
+static void __remove_filesrc_pad_block_foreach_cb(gpointer key, gpointer value, gpointer user_data)
+{
+       webrtc_gst_slot_s *source = (webrtc_gst_slot_s *)value;
+       int av_idx;
+
+       if (source->type != WEBRTC_MEDIA_SOURCE_TYPE_FILE)
+               return;
+
+       for (av_idx = 0; av_idx < AV_IDX_MAX; av_idx++) {
+               if (source->filesrc_av[av_idx].sink_pad_probe_id == 0)
+                       continue;
+
+               LOG_DEBUG("source[%p, id:%u] fakesink pad[%p] for [%s] is unblocked",
+                       source, source->id, source->filesrc_av[av_idx].sink_pad, GET_MEDIA_TYPE_NAME(av_idx == AV_IDX_AUDIO));
+               gst_pad_remove_probe(source->filesrc_av[av_idx].sink_pad, source->filesrc_av[av_idx].sink_pad_probe_id);
+               source->filesrc_av[av_idx].sink_pad = NULL;
+               source->filesrc_av[av_idx].sink_pad_probe_id = 0;
+       }
+}
+
+int _remove_filesrc_pad_block_probe(webrtc_s *webrtc)
+{
+       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+
+       g_hash_table_foreach(webrtc->gst.source_slots, __remove_filesrc_pad_block_foreach_cb, NULL);
+
+       return WEBRTC_ERROR_NONE;
+}
 //LCOV_EXCL_STOP