Use GST_TIME_FORMAT for buffer timestamps

[Version] 0.3.288
[Issue type] Logs

Change-Id: Ibc961bf988aa9225b5f4f4d1d7df2a381fbc459c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 5fdeb87f3088c36029a92801c750706bdf6a3bea..1afdff7fb50606e4f736ab259158d1c41c2319d2 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:    0.3.287
+Version:    0.3.288
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index 00472cf2975fc6929ab88c0218dacc1bc2c8ccb4..97e8cbf6bbfb48f60c4e98936e052ad2ca4f023f 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -937,8 +937,9 @@ static media_packet_h __make_media_packet(webrtc_gst_slot_s *sink, GstBuffer *bu
        if (ret != WEBRTC_ERROR_NONE)
                goto error;
 
-       LOG_DEBUG("sink[%p], packet[%p], buffer[%p, pts:%"G_GUINT64_FORMAT", dts:%"G_GUINT64_FORMAT", duration:%"G_GUINT64_FORMAT"]",
-               sink, packet, buffer, GST_BUFFER_PTS(buffer), GST_BUFFER_DTS(buffer), GST_BUFFER_DURATION(buffer));
+       LOG_DEBUG("sink[%p], packet[%p], buffer[%p, pts:%"GST_TIME_FORMAT", dts:%"GST_TIME_FORMAT", duration:%"GST_TIME_FORMAT"]",
+               sink, packet, buffer, GST_TIME_ARGS(GST_BUFFER_PTS(buffer)), GST_TIME_ARGS(GST_BUFFER_DTS(buffer)),
+               GST_TIME_ARGS(GST_BUFFER_DURATION(buffer)));
 
        return packet;
 

diff --git a/src/webrtc_source_mediapacket.c b/src/webrtc_source_mediapacket.c
index 04d24474713165c83954d5531081dc94fc2ee478..467672b8c997fdcce31f9ee85f74c59c43b8135b 100644 (file)
--- a/src/webrtc_source_mediapacket.c
+++ b/src/webrtc_source_mediapacket.c
@@ -112,8 +112,8 @@ static GstBuffer* __make_buffer_from_zerocopy_video_packet(webrtc_gst_slot_s *so
        GST_BUFFER_DURATION(buffer) = duration;
        gst_buffer_append_memory(buffer, mem);
 
-       LOG_DEBUG("buffer[%p, pts:%"G_GUINT64_FORMAT", dts:%"G_GUINT64_FORMAT", duration:%"G_GUINT64_FORMAT", tizen memory:%p]",
-               buffer, pts, dts, duration, mem);
+       LOG_DEBUG("buffer[%p, pts:%"GST_TIME_FORMAT", dts:%"GST_TIME_FORMAT", duration:%"GST_TIME_FORMAT", tizen memory:%p]",
+               buffer, GST_TIME_ARGS(pts), GST_TIME_ARGS(dts), GST_TIME_ARGS(duration), mem);
 
        return buffer;
 }
@@ -553,8 +553,8 @@ int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h
        }
        GST_BUFFER_DURATION(buffer) = duration;
 
-       LOG_DEBUG("new gst buffer[%p, pts:%"G_GUINT64_FORMAT", dts:%"G_GUINT64_FORMAT", duration:%"G_GUINT64_FORMAT"]",
-               buffer, pts, dts, duration);
+       LOG_DEBUG("new gst buffer[%p, pts:%"GST_TIME_FORMAT", dts:%"GST_TIME_FORMAT", duration:%"GST_TIME_FORMAT"]",
+               buffer, GST_TIME_ARGS(pts), GST_TIME_ARGS(dts), GST_TIME_ARGS(duration));
 
        g_signal_emit_by_name(G_OBJECT(appsrc), "push-buffer", buffer, &gst_ret, NULL);
        if (gst_ret != GST_FLOW_OK) {