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gst/audioresample/gstaudioresample.c: Implement latency query.

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Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
This commit is contained in:
Sebastian Dröge 2007-11-23 10:21:11 +00:00
parent d96e11806a
commit c7cc386eef
1 changed files with 78 additions and 0 deletions
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78
gst/audioresample/gstaudioresample.c
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@ -127,6 +127,9 @@ static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
static gboolean audioresample_start (GstBaseTransform * base);
static gboolean audioresample_stop (GstBaseTransform * base);
static gboolean audioresample_query (GstPad * pad, GstQuery * query);
static const GstQueryType *audioresample_query_type (GstPad * pad);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
@ -196,6 +199,9 @@ gst_audioresample_init (GstAudioresample * audioresample,
audioresample->filter_length = DEFAULT_FILTERLEN;
audioresample->need_discont = FALSE;
gst_pad_set_query_function (trans->srcpad, audioresample_query);
gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
}
/* vmethods */
@ -701,6 +707,76 @@ done:
return res;
}
static gboolean
audioresample_query (GstPad * pad, GstQuery * query)
{
GstAudioresample *audioresample =
GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = audioresample->i_rate;
gint resampler_latency = audioresample->filter_length / 2;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
if ((peer = gst_pad_get_peer (trans->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG ("Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
if (rate != 0 && resampler_latency != 0)
latency =
gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
else
latency = 0;
GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG ("Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (audioresample);
return res;
}
static const GstQueryType *
audioresample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
@ -718,6 +794,8 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
if (audioresample->resample) {
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
gst_element_post_message (GST_ELEMENT (audioresample),
gst_message_new_latency (GST_OBJECT (audioresample)));
}
break;
default: