webrtc_source: fix crash when change resolution while loopback

For sources whose resolution is dynamically changed, caps of appsrc for loopback should be changed as well.
Plus, initialize app source related information when destroy loopback pipeline.

[Version] 0.2.206
[Issue Type] Bug fix

Change-Id: I3dc9bd35a8e8f6075885eebd6f50af818054c4d4

diff --git a/include/webrtc_source_private.h b/include/webrtc_source_private.h
index b81b0d8b3d2bf914bf149b408bac4def9063993f..1104466486658ca746188758bad834b3f304a7ef 100644 (file)
--- a/include/webrtc_source_private.h
+++ b/include/webrtc_source_private.h
@@ -110,5 +110,7 @@ int _link_source_with_webrtcbin(webrtc_gst_slot_s *source, GstElement *webrtcbin
 int _create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool need_capsfilter, GList **element_list, bool is_audio);
 int _set_encoder_element_bitrate(GstElement *encoder, int target_bitrate);
 int _get_encoder_element_bitrate(GstElement *encoder, int *target_bitrate);
+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);
 
 #endif /* __TIZEN_MEDIA_WEBRTC_SOURCE_COMMON_H__ */

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index eea71f0429d52e418a507f34764b7b15a348acac..0275d99221a1da8491a2f72b31226f8ee8d3122e 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:    0.3.205
+Version:    0.3.206
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 20573baab1577adf16e9524db3c62db08277aaf5..e8417976515f54a591fa666a52a05b362865aac9 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1403,6 +1403,17 @@ int _set_video_resolution(webrtc_s *webrtc, unsigned int source_id, int width, i
                gst_caps_unref(new_caps);
        }
 
+       if (webrtc->state != WEBRTC_STATE_IDLE) {
+               GstCaps *new_caps;
+
+               if (!(new_caps = __make_video_raw_caps_with_resolution(source, &webrtc->ini, width, height)))
+                       return WEBRTC_ERROR_INVALID_OPERATION;
+               PRINT_CAPS(new_caps, "appsrc");
+
+               _unset_caps_for_render(source, AV_IDX_VIDEO);
+               _set_caps_for_render(source, new_caps, AV_IDX_VIDEO);
+       }
+
        source->video_info.width = width;
        source->video_info.height = height;
 
@@ -2463,6 +2474,8 @@ int _unset_audio_loopback(webrtc_s *webrtc, unsigned int source_id)
        gst_element_set_state(source->av[AV_IDX_AUDIO].render.pipeline, GST_STATE_NULL);
        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_AUDIO].render.pipeline);
 
+       source->av[AV_IDX_AUDIO].render.appsrc = NULL;
+
        return WEBRTC_ERROR_NONE;
 }
 
@@ -2532,6 +2545,8 @@ int _unset_video_loopback(webrtc_s *webrtc, unsigned int source_id)
        gst_element_set_state(source->av[AV_IDX_VIDEO].render.pipeline, GST_STATE_NULL);
        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_VIDEO].render.pipeline);
 
+       source->av[AV_IDX_VIDEO].render.appsrc = NULL;
+
        return WEBRTC_ERROR_NONE;
 }
 

diff --git a/src/webrtc_source_private.c b/src/webrtc_source_private.c
index 8bb77b40e4fc83eb8e3d3b9e346b2975fac70784..fc72187ac954f63819048009547020fff5fc9904 100644 (file)
--- a/src/webrtc_source_private.c
+++ b/src/webrtc_source_private.c
@@ -572,10 +572,7 @@ void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int i
        gst_object_unref(source->av[idx].render.src_pad);
        source->av[idx].render.src_pad = NULL;
 
-       if (source->av[idx].render.appsrc_caps) {
-               gst_caps_unref(source->av[idx].render.appsrc_caps);
-               source->av[idx].render.appsrc_caps = NULL;
-       }
+       _unset_caps_for_render(source, idx);
 }
 
 rtp_payload_info_s * _get_payload_info(webrtc_transceiver_codec_e codec)
@@ -1058,9 +1055,8 @@ int _create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool n
                        if ((sink_caps = __make_default_encoded_caps(source, &webrtc->ini, NULL))) {
                                PRINT_CAPS(sink_caps, "capsfilter");
                                g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
-                               source->av[idx].render.appsrc_caps = sink_caps;
-                               if (source->av[idx].render.appsrc)
-                                       g_object_set(G_OBJECT(source->av[idx].render.appsrc), "caps", sink_caps, NULL);
+
+                               _set_caps_for_render(source, sink_caps, idx);
                        }
 
                        source->av[idx].render.need_decoding = true;
@@ -1072,9 +1068,8 @@ int _create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool n
                if ((sink_caps = __make_default_raw_caps(source, &webrtc->ini))) {
                        PRINT_CAPS(sink_caps, "capsfilter");
                        g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
-                       source->av[idx].render.appsrc_caps = sink_caps;
-                       if (source->av[idx].render.appsrc)
-                               g_object_set(G_OBJECT(source->av[idx].render.appsrc), "caps", sink_caps, NULL);
+
+                       _set_caps_for_render(source, sink_caps, idx);
                }
 
                _add_probe_to_pad_for_render(source, idx, gst_element_get_static_pad(capsfilter, "src"), _source_data_probe_cb);
@@ -1150,3 +1145,25 @@ error:
 
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
+
+void _set_caps_for_render(webrtc_gst_slot_s *source, GstCaps *caps, int av_idx)
+{
+       RET_IF(source == NULL, "source is NULL");
+       RET_IF(caps == NULL, "caps is NULL");
+
+       source->av[av_idx].render.appsrc_caps = caps;
+
+       if (source->av[av_idx].render.appsrc)
+               g_object_set(G_OBJECT(source->av[av_idx].render.appsrc), "caps", caps, NULL);
+}
+
+void _unset_caps_for_render(webrtc_gst_slot_s *source, int av_idx)
+{
+       RET_IF(source == NULL, "source is NULL");
+
+       if (!(source->av[av_idx].render.appsrc_caps))
+               return;
+
+       gst_caps_unref(source->av[av_idx].render.appsrc_caps);
+       source->av[av_idx].render.appsrc_caps = NULL;
+}