[gst-cvs] gst-plugins-base: audiorate: move debug calculation into debug macro

[<thi...@ke...>]

Module: gst-plugins-base
Branch: master
Commit: e55bf9bdd82299b8498b267b67997872df7e18fa
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=e55bf9bdd82299b8498b267b67997872df7e18fa

Author: Thiago Santos <thi...@em...>
Date:   Thu Oct 22 09:12:03 2009 -0300

audiorate: move debug calculation into debug macro

Remove in_duration and move its calculation to
GST_LOG_OBJECT macro. This way it will only be calculated
if we have debug enabled.

---

 gst/audiorate/gstaudiorate.c |    9 +++------
 1 files changed, 3 insertions(+), 6 deletions(-)

diff --git a/gst/audiorate/gstaudiorate.c b/gst/audiorate/gstaudiorate.c
index cf28dad..c87c234 100644
--- a/gst/audiorate/gstaudiorate.c
+++ b/gst/audiorate/gstaudiorate.c
@@ -475,7 +475,7 @@ static GstFlowReturn
 gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
 {
   GstAudioRate *audiorate;
-  GstClockTime in_time, in_duration, run_time;
+  GstClockTime in_time, run_time;
   guint64 in_offset, in_offset_end, in_samples;
   guint in_size;
   GstFlowReturn ret = GST_FLOW_OK;
@@ -519,9 +519,6 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
 
   in_size = GST_BUFFER_SIZE (buf);
   in_samples = in_size / audiorate->bytes_per_sample;
-  /* get duration from the size because we can and it's more accurate */
-  in_duration =
-      gst_util_uint64_scale_int_round (in_samples, GST_SECOND, audiorate->rate);
 
   /* Figure out the total accumulated segment time. */
   run_time = in_time + audiorate->src_segment.accum;
@@ -537,8 +534,8 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
       ", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%"
       G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT,
       GST_TIME_ARGS (in_time), GST_TIME_ARGS (run_time),
-      GST_TIME_ARGS (in_duration), in_size, in_offset, in_offset_end,
-      audiorate->next_offset);
+      GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, audiorate->rate)),
+      in_size, in_offset, in_offset_end, audiorate->next_offset);
 
   /* do we need to insert samples */
   if (in_offset > audiorate->next_offset) {