diff options
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 78 |
1 files changed, 78 insertions, 0 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c index a475baf8b..ebcd1ea6a 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/audioresample/gstaudioresample.c @@ -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: |