[farsight2/master] Use the gst debug stuff everywhere else

[Olivier Crête]

---
 gst/fsrtpconference/fs-rtp-codec-cache.c |    4 ++--
 gst/fsrtpconference/fs-rtp-conference.c  |    2 +-
 gst/fsrtpconference/fs-rtp-conference.h  |    1 +
 gst/fsrtpconference/fs-rtp-session.c     |    6 ++++--
 gst/fsrtpconference/fs-rtp-substream.c   |    4 +++-
 5 files changed, 11 insertions(+), 6 deletions(-)

diff --git a/gst/fsrtpconference/fs-rtp-codec-cache.c b/gst/fsrtpconference/fs-rtp-codec-cache.c
index 5ea1816..1ec4056 100644
--- a/gst/fsrtpconference/fs-rtp-codec-cache.c
+++ b/gst/fsrtpconference/fs-rtp-codec-cache.c
@@ -38,8 +38,8 @@
 #ifdef STANDALONE
 # undef GST_WARNING
 # undef GST_DEBUG
-# undef GST_LOG
 # define GST_DEBUG(...) g_debug (__VA_ARGS__)
+# define GST_WARNING(...) g_warning (__VA_ARGS__)
 #else
 # include "fs-rtp-conference.h"
 # define GST_CAT_DEFAULT fsrtpconference_disco
@@ -522,7 +522,7 @@ save_codecs_cache(FsMediaType media_type, GList *blueprints)
        item = g_list_next (item)) {
     CodecBlueprint *codec_blueprint = item->data;
     if (!save_codec_blueprint (fd, codec_blueprint)) {
-      g_warning ("Unable to save codec cache");
+      GST_WARNING ("Unable to save codec cache");
       close (fd);
       g_free (tmp_path);
       g_free (cache_path);
diff --git a/gst/fsrtpconference/fs-rtp-conference.c b/gst/fsrtpconference/fs-rtp-conference.c
index 38f9991..6f5245c 100644
--- a/gst/fsrtpconference/fs-rtp-conference.c
+++ b/gst/fsrtpconference/fs-rtp-conference.c
@@ -40,7 +40,7 @@
 
 #include <string.h>
 
-GST_DEBUG_CATEGORY_STATIC (fsrtpconference_debug);
+GST_DEBUG_CATEGORY (fsrtpconference_debug);
 GST_DEBUG_CATEGORY (fsrtpconference_disco);
 GST_DEBUG_CATEGORY (fsrtpconference_nego);
 #define GST_CAT_DEFAULT fsrtpconference_debug
diff --git a/gst/fsrtpconference/fs-rtp-conference.h b/gst/fsrtpconference/fs-rtp-conference.h
index a02c49d..646a002 100644
--- a/gst/fsrtpconference/fs-rtp-conference.h
+++ b/gst/fsrtpconference/fs-rtp-conference.h
@@ -67,6 +67,7 @@ struct _FsRtpConferenceClass
 GType fs_rtp_conference_get_type(void);
 
 
+GST_DEBUG_CATEGORY_EXTERN (fsrtpconference_debug);
 GST_DEBUG_CATEGORY_EXTERN (fsrtpconference_disco);
 GST_DEBUG_CATEGORY_EXTERN (fsrtpconference_nego);
 
diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c
index 7c50a2b..c73b268 100644
--- a/gst/fsrtpconference/fs-rtp-session.c
+++ b/gst/fsrtpconference/fs-rtp-session.c
@@ -46,6 +46,8 @@
 #include "fs-rtp-codec-negotiation.h"
 #include "fs-rtp-substream.h"
 
+#define GST_CAT_DEFAULT fsrtpconference_debug
+
 /* Signals */
 enum
 {
@@ -544,7 +546,7 @@ fs_rtp_session_set_property (GObject *object,
             new_local_codecs_configuration;
 
         } else {
-          g_warning ("Invalid new codec configurations");
+          GST_WARNING ("Invalid new codec configurations");
         }
       } else {
         if (self->priv->local_codecs_configuration)
@@ -1592,7 +1594,7 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
   else
     pipeline_factory = blueprint->receive_pipeline_factory;
 
-  g_debug ("creating %s codec bin for id %d, pipeline_factory %p",
+  GST_DEBUG ("creating %s codec bin for id %d, pipeline_factory %p",
     direction_str, codec->id, pipeline_factory);
   codec_bin = gst_bin_new (name);
 
diff --git a/gst/fsrtpconference/fs-rtp-substream.c b/gst/fsrtpconference/fs-rtp-substream.c
index 39d465a..fb1ca37 100644
--- a/gst/fsrtpconference/fs-rtp-substream.c
+++ b/gst/fsrtpconference/fs-rtp-substream.c
@@ -33,6 +33,8 @@
 #include <gst/farsight/fs-stream.h>
 #include <gst/farsight/fs-session.h>
 
+#define GST_CAT_DEFAULT fsrtpconference_debug
+
 /**
  * SECTION:fs-rtp-sub-stream
  * @short_description: The receive codec bin for a ssrc and a pt
@@ -325,7 +327,7 @@ fs_rtp_sub_stream_set_property (GObject *object,
     case PROP_STREAM:
       FS_RTP_SESSION_LOCK (self->priv->session);
       if (self->priv->stream)
-        g_warning ("Stream already set, not re-setting");
+        GST_WARNING ("Stream already set, not re-setting");
       else
         self->priv->stream = g_value_get_object (value);
       FS_RTP_SESSION_UNLOCK (self->priv->session);
-- 
1.5.6.5