webrtc_private: Add probe callback to the newly added source pad of webrtcbin

_add_probe_to_pad_for_render() and _remove_probe_to_pad_for_render() are moved
from webrtc_source_loopback.c.

It has been fixed to use gst_object_ref() before assigning it to slot's variable.

This probe callback will be used to decide to render the RTP data or not.

[Version] 0.4.53
[Issue Type] Improvement

Change-Id: I9b539c842e754e4d6d93cc9f95f28f1b79184f00
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index f8f0a0779b273d62d2aff1e0a3f89aab9d7deabf..2afdf1feafae804fc484cd4966fb7657e0b67b63 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -793,8 +793,6 @@ int _get_display_mode_from_loopback(webrtc_s *webrtc, unsigned int track_id, web
 int _set_display_visible_to_loopback(webrtc_s *webrtc, unsigned int track_id, bool visible);
 int _get_display_visible_from_loopback(webrtc_s *webrtc, unsigned int track_id, bool *visible);
 GstPadProbeReturn _source_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpointer user_data);
-void _add_probe_to_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx, GstPad *pad, void *probe_cb);
-void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx);
 void _set_caps_for_render(webrtc_gst_slot_s *source, GstCaps *caps, int av_idx);
 void _unset_caps_for_render(webrtc_gst_slot_s *source, int av_idx);
 int _update_caps_for_render_with_resolution(webrtc_gst_slot_s *source, int width, int height);
@@ -841,8 +839,9 @@ bool _is_owner_of_track_build_context(webrtc_s *webrtc, unsigned int track_id);
 int _decodebin_autoplug_select_cb(GstElement *decodebin, GstPad *pad, GstCaps *caps, GstElementFactory *factory, gpointer user_data);
 void _track_build_context_destroy_cb(gpointer data);
 void _sink_slot_destroy_cb(gpointer data);
-int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio);
-int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio);
+GstPadProbeReturn _webrtcbin_payloaded_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpointer user_data);
+int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio, webrtc_gst_slot_s **sink_slot);
+int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio, webrtc_gst_slot_s **sink_slot);
 int _set_stream_info_to_sink(webrtc_s *webrtc, unsigned int track_id, sound_stream_info_h stream_info);
 int _set_display_to_sink(webrtc_s *webrtc, unsigned int track_id, unsigned int type, void *display);
 int _set_display_surface_id_to_sink(webrtc_s *webrtc, unsigned int track_id, int surface_id, int x, int y, int width, int height);
@@ -932,6 +931,8 @@ int _check_feature(const char *feature);
 /* private */
 int _stop(webrtc_s *webrtc);
 int _gst_init(webrtc_s *webrtc);
+void _add_probe_to_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx, GstPad *pad, GstPadProbeCallback probe_cb);
+void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx);
 int _gst_build_pipeline(webrtc_s *webrtc);
 void _gst_destroy_pipeline(webrtc_s *webrtc);
 int _gst_pipeline_set_state(webrtc_s *webrtc, GstState state);

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 1cc92c086c02e1ee7c2daabc38052bdef63c69b4..9a260a06965fa0e6cb48c746731bf980606890bd 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:    0.4.52
+Version:    0.4.53
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 1d1859b95017bf41bca1a03c78bb49ad99814104..c6df867ccb65c9b729d8075528ff9003cb09ef30 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1457,15 +1457,56 @@ bool _is_audio_media_type(const char *media_type)
        return (bool)g_strrstr(media_type, "audio");
 }
 
+void _add_probe_to_pad_for_render(webrtc_gst_slot_s *slot, unsigned int idx, GstPad *pad, GstPadProbeCallback probe_cb)
+{
+       probe_userdata_s *probe_userdata;
+
+       ASSERT(slot);
+       ASSERT(idx < AV_IDX_MAX);
+       ASSERT(pad);
+       ASSERT(probe_cb);
+
+       probe_userdata = g_new0(probe_userdata_s, 1);
+       probe_userdata->slot = slot;
+       probe_userdata->av_idx = idx;
+       slot->av[idx].render.src_pad = gst_object_ref(pad);
+       slot->av[idx].render.src_pad_probe_id = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER,
+                       probe_cb, probe_userdata, g_free);
+
+       LOG_DEBUG("slot[id:%u, av_idx:%u] pad[%p] probe[id:%lu, callback:%p]",
+               slot->id, idx, pad, slot->av[idx].render.src_pad_probe_id, probe_cb);
+}
+
+void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *slot, unsigned int idx)
+{
+       ASSERT(slot);
+       ASSERT(idx < AV_IDX_MAX);
+
+       if (slot->av[idx].render.src_pad_probe_id == 0)
+               return;
+
+       LOG_DEBUG("slot[id:%u, av_idx:%s] pad[%p] probe_id[%lu]",
+               slot->id, GET_MEDIA_TYPE_NAME(idx == AV_IDX_AUDIO), slot->av[idx].render.src_pad, slot->av[idx].render.src_pad_probe_id);
+
+       gst_pad_remove_probe(slot->av[idx].render.src_pad, slot->av[idx].render.src_pad_probe_id);
+       slot->av[idx].render.src_pad_probe_id = 0;
+
+       gst_object_unref(slot->av[idx].render.src_pad);
+       slot->av[idx].render.src_pad = NULL;
+
+       _unset_caps_for_render(slot, idx);
+}
+
 static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpointer user_data)
 {
        int ret = WEBRTC_ERROR_NONE;
        webrtc_s *webrtc = (webrtc_s *)user_data;
        gchar *media_type = NULL;
        bool is_audio;
+       webrtc_gst_slot_s *sink;
        g_autoptr(GMutexLocker) locker = NULL;
 
-       RET_IF(webrtc == NULL, "webrtc is NULL");
+       ASSERT(webrtc);
 
        if (GST_PAD_DIRECTION(new_pad) != GST_PAD_SRC)
                return;
@@ -1492,13 +1533,15 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
 
        if ((!is_audio && webrtc->encoded_video_frame_cb.callback) ||
                (is_audio && webrtc->encoded_audio_frame_cb.callback)) {
-               ret = _add_forwarding_sink_bin(webrtc, new_pad, is_audio);
+               ret = _add_forwarding_sink_bin(webrtc, new_pad, is_audio, &sink);
                RET_IF(ret != WEBRTC_ERROR_NONE, "failed to _add_forwarding_sink_bin()");
        } else {
-               ret = _add_rendering_sink_bin(webrtc, new_pad, is_audio);
+               ret = _add_rendering_sink_bin(webrtc, new_pad, is_audio, &sink);
                RET_IF(ret != WEBRTC_ERROR_NONE, "failed to _add_rendering_sink_bin()");
        }
 
+       _add_probe_to_pad_for_render(sink, is_audio ? AV_IDX_AUDIO : AV_IDX_VIDEO, new_pad, _webrtcbin_payloaded_data_probe_cb);
+
        GENERATE_DOT(webrtc, webrtc->gst.pipeline, "%s.webrtcbin-%s", GST_ELEMENT_NAME(webrtc->gst.pipeline), GST_PAD_NAME(new_pad));
 }
 

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index 4e164e9084f366a044001e5d0909ee5c263b6ec0..64c892e7a32698f7adbcd6a4c0a2a75ad53f072d 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -632,6 +632,8 @@ void _sink_slot_destroy_cb(gpointer data)
        if (sink->sound_stream_info.type)
                free(sink->sound_stream_info.type);
 
+       _remove_probe_from_pad_for_render(sink, AV_IDX_AUDIO);
+       _remove_probe_from_pad_for_render(sink, AV_IDX_VIDEO);
        _remove_probe_from_pad_for_dump(sink);
 
 #ifdef TIZEN_FEATURE_SNAPSHOT
@@ -704,15 +706,16 @@ error:
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
-int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
+int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio, webrtc_gst_slot_s **sink_slot)
 {
        gchar *bin_name;
        gchar *track_name;
        webrtc_gst_slot_s *sink;
        GstElement *decodebin;
 
-       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
-       RET_VAL_IF(src_pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "src_pad is NULL");
+       ASSERT(webrtc);
+       ASSERT(src_pad);
+       ASSERT(sink_slot);
 
        if (__alloc_sink_slot(webrtc, GST_PAD_NAME(src_pad), false, &sink, &bin_name) != WEBRTC_ERROR_NONE)
                return WEBRTC_ERROR_INVALID_OPERATION;
@@ -745,6 +748,8 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
 
        LOG_INFO("added a sink slot[%p, id:%u]", sink, sink->id);
 
+       *sink_slot = sink;
+
        return WEBRTC_ERROR_NONE;
 
 error_before_insert:
@@ -1034,7 +1039,22 @@ static GstCaps* __make_caps_if_h264_or_h265(GstPad *pad)
        return new_caps;
 }
 
-int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
+GstPadProbeReturn _webrtcbin_payloaded_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpointer user_data)
+{
+       probe_userdata_s *probe_data = (probe_userdata_s *)user_data;
+       g_autofree gchar *media_type = _get_mime_type_from_pad(pad);
+       webrtc_gst_slot_s *sink;
+
+       ASSERT(probe_data);
+
+       sink = probe_data->slot;
+       LOG_VERBOSE("pad[%p] media_type[%s] av_idx[%d] sink[%p, id:%u]",
+               pad, media_type, probe_data->av_idx, sink, sink->id);
+
+       return GST_PAD_PROBE_OK;
+}
+
+int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio, webrtc_gst_slot_s **sink_slot)
 {
        gchar *bin_name;
        gchar *track_name;
@@ -1046,8 +1066,9 @@ int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
        GstCaps *sink_caps;
        webrtc_gst_slot_s *sink;
 
-       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
-       RET_VAL_IF(src_pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "src_pad is NULL");
+       ASSERT(webrtc);
+       ASSERT(src_pad);
+       ASSERT(sink_slot);
 
        if (__alloc_sink_slot(webrtc, GST_PAD_NAME(src_pad), true, &sink, &bin_name) != WEBRTC_ERROR_NONE)
                return WEBRTC_ERROR_INVALID_OPERATION;
@@ -1101,6 +1122,8 @@ int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
 
        LOG_INFO("added a sink slot[%p, id:%u] to webrtc[%p]", sink, sink->id, webrtc);
 
+       *sink_slot = sink;
+
        return WEBRTC_ERROR_NONE;
 
 error_before_insert:

diff --git a/src/webrtc_source_loopback.c b/src/webrtc_source_loopback.c
index 7455f6e6a0c1dd44669685b0c2d932fababac5d4..70999a111f30eb337e1fdbdc95bab9f2506cbfda 100644 (file)
--- a/src/webrtc_source_loopback.c
+++ b/src/webrtc_source_loopback.c
@@ -588,46 +588,6 @@ GstPadProbeReturn _source_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpoi
        return GST_PAD_PROBE_OK;
 }
 
-void _add_probe_to_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx, GstPad *pad, void *probe_cb)
-{
-       probe_userdata_s *probe_userdata;
-
-       RET_IF(source == NULL, "source is NULL");
-       RET_IF(idx >= AV_IDX_MAX, "invalid idx(%u)", idx);
-       RET_IF(pad == NULL, "pad is NULL");
-       RET_IF(probe_cb == NULL, "probe_cb is NULL");
-
-       probe_userdata = g_new0(probe_userdata_s, 1);
-       probe_userdata->slot = source;
-       probe_userdata->av_idx = idx;
-       source->av[idx].render.src_pad = pad;
-       source->av[idx].render.src_pad_probe_id = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER,
-                       probe_cb, probe_userdata, g_free);
-
-       LOG_DEBUG("source[id:%u, av_idx:%u] pad[%p] probe[id:%lu, callback:%p]",
-               source->id, idx, pad, source->av[idx].render.src_pad_probe_id, probe_cb);
-}
-
-void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx)
-{
-       RET_IF(source == NULL, "source is NULL");
-       RET_IF(idx >= AV_IDX_MAX, "invalid idx(%u)", idx);
-
-       if (source->av[idx].render.src_pad_probe_id == 0)
-               return;
-
-       LOG_DEBUG("source[id:%u, av_idx:%s] pad[%p] probe_id[%lu]",
-               source->id, GET_MEDIA_TYPE_NAME(idx == AV_IDX_AUDIO), source->av[idx].render.src_pad, source->av[idx].render.src_pad_probe_id);
-
-       gst_pad_remove_probe(source->av[idx].render.src_pad, source->av[idx].render.src_pad_probe_id);
-       source->av[idx].render.src_pad_probe_id = 0;
-
-       gst_object_unref(source->av[idx].render.src_pad);
-       source->av[idx].render.src_pad = NULL;
-
-       _unset_caps_for_render(source, idx);
-}
-
 void _set_caps_for_render(webrtc_gst_slot_s *source, GstCaps *caps, int av_idx)
 {
        RET_IF(source == NULL, "source is NULL");