From 0dcd5d3a6bc654444a466b6197f8ea715b946075 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Wed, 14 Mar 2007 14:09:21 +0000 Subject: add debugging and reformat docs Original commit message from CVS: add debugging and reformat docs --- gst/audioresample/gstaudioresample.c | 29 +++++++++++++++++++++-------- 1 file changed, 21 insertions(+), 8 deletions(-) (limited to 'gst/audioresample') 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); -- cgit v1.2.1