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authorThomas Vander Stichele <thomas@apestaart.org>2007-03-14 14:09:21 +0000
committerThomas Vander Stichele <thomas@apestaart.org>2007-03-14 14:09:21 +0000
commit0dcd5d3a6bc654444a466b6197f8ea715b946075 (patch)
tree1aea0129b6c99b1f2f4c6515345180f26d5154bf
parent3c899d4a12d704caa491a942c0d3648b9844a6ce (diff)
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add debugging and reformat docs
Original commit message from CVS: add debugging and reformat docs
-rw-r--r--gst/audioresample/gstaudioresample.c29
1 files changed, 21 insertions, 8 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
index e5206361..d670858e 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/audioresample/gstaudioresample.c
@@ -540,8 +540,8 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
/* this is an error that when it happens, would need fixing in the
- * resample library; we told
- * it we wanted only GST_BUFFER_SIZE (outbuf), and it gave us more ! */
+ * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
+ * and it gave us more ! */
GST_WARNING_OBJECT (audioresample,
"audioresample, you memory corrupting bastard. "
"you gave me outsize %d while my buffer was size %d",
@@ -556,6 +556,14 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
}
GST_BUFFER_SIZE (outbuf) = outsize;
+ GST_LOG_OBJECT (audioresample, "transformed to buffer of %ld bytes, ts %"
+ GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
+ G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
+ outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+ GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
+
+
return GST_FLOW_OK;
}
@@ -576,7 +584,12 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
- GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
+ GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
+ GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
+ G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
+ size, GST_TIME_ARGS (timestamp),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
+ GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
if (audioresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
@@ -584,14 +597,14 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstClockTime stime;
- /* offset used to calculate the timestamps. We use the sample offset for this
- * to make it more accurate. We want the first buffer to have the same timestamp
- * as the incomming timestamp. */
+ /* offset used to calculate the timestamps. We use the sample offset for
+ * this to make it more accurate. We want the first buffer to have the
+ * same timestamp as the incoming timestamp. */
audioresample->next_ts = timestamp;
audioresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
- /* offset used to set as the buffer offset, this offset is always relative
- * to the stream time, note that timestamp is not... */
+ /* offset used to set as the buffer offset, this offset is always
+ * relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
audioresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);