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audiorate: be more accurate on offset math

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Replace gst_util_uint64_scale_int for its rounding version
to improve accuracy and avoid inserting samples where
they aren't needed.

Fixes #499181
This commit is contained in:
Thiago Santos 2009-10-22 08:40:01 -03:00
parent eb9acba08f
commit 44d6ebc48f
1 changed files with 4 additions and 2 deletions
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6
gst/audiorate/gstaudiorate.c
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@ -175,6 +175,7 @@ gst_audio_rate_base_init (gpointer g_class)
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_rate_src_template));
}
static void
gst_audio_rate_class_init (GstAudioRateClass * klass)
{
@ -520,14 +521,15 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * 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 (in_samples, GST_SECOND, audiorate->rate);
gst_util_uint64_scale_int_round (in_samples, GST_SECOND, audiorate->rate);
in_stop = in_time + in_duration;
/* Figure out the total accumulated segment time. */
run_time = in_time + audiorate->src_segment.accum;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int (run_time, audiorate->rate, GST_SECOND);
in_offset = gst_util_uint64_scale_int_round (run_time, audiorate->rate,
GST_SECOND);
in_offset_end = in_offset + in_samples;
GST_LOG_OBJECT (audiorate,