<div dir="ltr"><div>Ah! That was it ! Thanks! This property seems not documented either in audiomixer/aggregator. Any reason for this ? </div>Ajit. <div><div> <div class="gmail_quote"><div dir="ltr">On Tue, 29 Nov 2016 at 17:37 Nicolas Dufresne <<a href="mailto:nicolas@ndufresne.ca">nicolas@ndufresne.ca</a>> wrote: </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi, Le mardi 29 novembre 2016 à 22:56 +0000, Ajit Warrier a écrit : > I am using the following pipeline on an x86 PC, using v1.10.1: > > gst-launch-1.0 -v udpsrc caps="application/x-rtp" port=1234 > multicast-group=239.0.0.190 ! rtpg722depay ! avdec_g722 ! audiomixer > ! alsasink > > It receives g722 coded rtp audio and plays it out. When using the > audiomixer in this pipeline, if I perform the following operations: > > - start pipeline > - wait for couple of minutes > - THEN send rtp audio > > I experience a delay of 5-6 seconds before playback starts. This only > happens if I wait for couple of minutes, in fact the playback delay > gets worse the longer I wait. And it only happens with the audiomixer > in the pipeline. Without the audiomixer, there is no playback delay. You should be able to solve this issue using the "start-time-selection" property. on the audio mixer. Using the method first (1). ... ! audiomixer start-time-selection=first ! ... Otherwise the mixer will produce silence from 0 to the first buffer running time received, which in most cases lead to this pause. > > I ran GST_DEBUG=6 and got the following output that looks suspicious: > > Multiple repeated instances of: > > 0:02:52.285149489 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3360:gst_base_sink_chain_unlocked:<alsasink0> got times > start: 0:02:47.560000000, end: 0:02:47.570000000 > 0:02:52.285156228 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:1946:gst_base_sink_get_sync_times:<alsasink0> got times > start: 0:02:47.560000000, stop: 0:02:47.570000000, do_sync 0 > 0:02:52.285163242 5934 0x55dc774f8d40 LOG basesink > gstbasesink.c:2483:gst_base_sink_do_sync:<alsasink0> avg frame diff > 0:00:00.010000000 > 0:02:52.285168838 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3468:gst_base_sink_chain_unlocked:<alsasink0> rendering > object 0x7f3934017af0 > 0:02:52.285173812 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:1899:gst_audio_base_sink_render:<alsasink0> > Received GAP or ignoring audio for trickplay. Dropping contents > 0:02:52.285178641 5934 0x55dc774f8d40 DEBUG GST_EVENT > gstevent.c:305:gst_event_new_custom: creating new event > 0x7f393c004a60 gap 40966 > 0:02:52.285184292 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:912:gst_audio_ring_buffer_start:<audiosinkringbu > ffer0> starting ringbuffer > 0:02:52.285188437 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:929:gst_audio_ring_buffer_start:<audiosinkringbu > ffer0> was not stopped, try paused > 0:02:52.285192343 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:937:gst_audio_ring_buffer_start:<audiosinkringbu > ffer0> was not paused, must have been started > 0:02:52.285199470 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:1911:gst_base_sink_get_sync_times:<alsasink0> Got Gap > time 0:02:47.560000000 duration 0:00:00.010000000 > 0:02:52.285206383 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:1946:gst_base_sink_get_sync_times:<alsasink0> got times > start: 0:02:47.560000000, stop: 0:02:47.570000000, do_sync 1 > 0:02:52.285213779 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:2523:gst_base_sink_do_sync:<alsasink0> reset rc_time to > time 0:02:47.780000000 > 0:02:52.285219206 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:2535:gst_base_sink_do_sync:<alsasink0> possibly waiting > for clock to reach 0:02:47.560000000, adjusted 0:02:47.780000000 > 0:02:52.285225617 5934 0x55dc774f8d40 LOG basesink > gstbasesink.c:2117:gst_base_sink_wait_clock:<alsasink0> time > 0:02:47.780000000, base_time 1349:31:49.960166366 > 0:02:52.285232112 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:535:gst_clock_id_wait:<GstSystemClock> waiting on clock > entry 0x7f393c0048a0 > 0:02:52.285236787 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:1045:gst_clock_get_internal_time:<GstSystemClock> internal > time 1349:34:41.935386462 > 0:02:52.285242365 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:1090:gst_clock_get_time:<GstSystemClock> adjusted time > 1349:34:41.935386462 > 0:02:52.285247488 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstsystemclock.c:717:gst_system_clock_id_wait_jitter_unlocked: entry > 0x7f393c0048a0 time 1349:34:37.740166366 now 1349:34:41.935386462 > diff (time-now) -<a href="tel:(419)%20522-0096" value="+14195220096" class="gmail_msg" target="_blank">4195220096</a> > 0:02:52.285254637 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:545:gst_clock_id_wait:<GstSystemClock> done waiting entry > 0x7f393c0048a0, res: 1 (early) > 0:02:52.285259717 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:2542:gst_base_sink_do_sync:<alsasink0> clock returned > 1, jitter 0:00:04.195220096 > 0:02:52.285265829 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:2858:gst_base_sink_is_too_late:<alsasink0> frame > dropping disabled > 0:02:52.285269802 5934 0x55dc774f8d40 LOG GST_EVENT > gstevent.c:222:_gst_event_free: freeing event 0x7f393c004a60 type gap > 0:02:52.285275293 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3507:gst_base_sink_chain_unlocked:<alsasink0> object > unref after render 0x7f3934017af0 > 0:02:52.285279814 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:724:_gst_buffer_free: finalize 0x7f3934017af0 > 0:02:52.285284056 5934 0x55dc774f8d40 DEBUG GST_MEMORY > gstmemory.c:87:_gst_memory_free: free memory 0x7f393c012620 > 0:02:52.285289468 5934 0x55dc774f8d40 DEBUG GST_SCHEDULING > gstpad.c:4209:gst_pad_chain_data_unchecked:<alsasink0:sink> called > chainfunction &gst_base_sink_chain with buffer 0x7f3934017af0, > returned ok > 0:02:52.285295407 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1372:gst_audio_aggregator_aggregate:<audiomixer0 > > pushed outbuf, result = ok > 0:02:52.285300098 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:831:gst_aggregator_aggregate_func:<audiomixer0> flow > return is ok > 0:02:52.285305893 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function 0x7f39559040f0 on pad > 0:02:52.285311540 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:428:gst_aggregator_check_pads_ready:<audiomixer0> > checking pads > 0:02:52.285315809 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:460:gst_aggregator_check_pads_ready:<audiomixer0> > pads are ready > 0:02:52.285319732 5934 0x55dc774f8d40 DEBUG aggregator > gstaggregator.c:630:gst_aggregator_wait_and_check:<audiomixer0> all > pads have data > 0:02:52.285324475 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function 0x7f39559040f0 on pad > 0:02:52.285330484 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function sync_pad_values on pad > 0:02:52.285338421 5934 0x55dc774f8d40 LOG default > gstobject.c:1120:gst_object_sync_values:<audiomixer0:sink_0> > sync_values > 0:02:52.285343028 5934 0x55dc774f8d40 DEBUG GST_MEMORY > gstmemory.c:138:gst_memory_init: new memory 0x7f393c012440, > maxsize:327 offset:0 size:320 > 0:02:52.285348790 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:798:gst_buffer_new: new 0x7f3934017c00 > 0:02:52.285352988 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:413:_memory_add: buffer 0x7f3934017c00, idx -1, mem > 0x7f393c012440 > 0:02:52.285357932 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:854:gst_buffer_new_allocate: new buffer 0x7f3934017c00 of > size 320 from allocator (nil) > 0:02:52.285362993 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:1720:gst_buffer_map_range: buffer 0x7f3934017c00, idx 0, > length -1, flags 0002 > 0:02:52.285368270 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:212:_get_merged_memory: buffer 0x7f3934017c00, idx 0, > length 1 > 0:02:52.285373686 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1194:gst_audio_aggregator_aggregate:<audiomixer0 > > Starting to mix 160 samples for offset 2681120 with timestamp > 0:02:47.570000000 > 0:02:52.285380278 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1365:gst_audio_aggregator_aggregate:<audiomixer0 > > pushing outbuf 0x7f3934017c00, timestamp 0:02:47.570000000 offset > 2681120 > 0:02:52.285386988 5934 0x55dc774f8d40 DEBUG GST_SCHEDULING > gstpad.c:4203:gst_pad_chain_data_unchecked:<alsasink0:sink> calling > chainfunction &gst_base_sink_chain with buffer buffer: > 0x7f3934017c00, pts 0:02:47.570000000, dts 99: > 99:99.999999999, dur 0:00:00.010000000, size 320, offset 2681120, > offset_end 2681280, flags 0x800 > > > Finally followed by the playback: > > 0:02:52.285535775 5934 0x55dc774f8d40 DEBUG GST_SCHEDULING > gstpad.c:4209:gst_pad_chain_data_unchecked:<alsasink0:sink> called > chainfunction &gst_base_sink_chain with buffer 0x7f3934017c00, > returned ok > 0:02:52.285541688 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1372:gst_audio_aggregator_aggregate:<audiomixer0 > > pushed outbuf, result = ok > 0:02:52.285546617 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:831:gst_aggregator_aggregate_func:<audiomixer0> flow > return is ok > 0:02:52.285552504 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function 0x7f39559040f0 on pad > 0:02:52.285558288 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:428:gst_aggregator_check_pads_ready:<audiomixer0> > checking pads > 0:02:52.285562564 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:460:gst_aggregator_check_pads_ready:<audiomixer0> > pads are ready > 0:02:52.285566505 5934 0x55dc774f8d40 DEBUG aggregator > gstaggregator.c:630:gst_aggregator_wait_and_check:<audiomixer0> all > pads have data > 0:02:52.285571214 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function 0x7f39559040f0 on pad > 0:02:52.285577286 5934 0x55dc774f8d40 LOG aggregator > gstaggregator.c:378:gst_aggregator_iterate_sinkpads:<audiomixer0:sink > _0> calling function sync_pad_values on pad > 0:02:52.285582555 5934 0x55dc774f8d40 LOG default > gstobject.c:1120:gst_object_sync_values:<audiomixer0:sink_0> > sync_values > 0:02:52.285586893 5934 0x55dc774f8d40 DEBUG GST_MEMORY > gstmemory.c:138:gst_memory_init: new memory 0x7f393c012260, > maxsize:327 offset:0 size:320 > 0:02:52.285592416 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:798:gst_buffer_new: new 0x7f3934017c00 > 0:02:52.285596623 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:413:_memory_add: buffer 0x7f3934017c00, idx -1, mem > 0x7f393c012260 > 0:02:52.285604025 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:854:gst_buffer_new_allocate: new buffer 0x7f3934017c00 of > size 320 from allocator (nil) > 0:02:52.285609397 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:1720:gst_buffer_map_range: buffer 0x7f3934017c00, idx 0, > length -1, flags 0002 > 0:02:52.285614824 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:212:_get_merged_memory: buffer 0x7f3934017c00, idx 0, > length 1 > 0:02:52.285620362 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1194:gst_audio_aggregator_aggregate:<audiomixer0 > > Starting to mix 160 samples for offset 2681280 with timestamp > 0:02:47.580000000 > 0:02:52.285627060 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1277:gst_audio_aggregator_aggregate:<audiomixer0 > :sink_0> Mixing buffer for current offset > 0:02:52.285631472 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:1720:gst_buffer_map_range: buffer 0x7f3934017c00, idx 0, > length -1, flags 0003 > 0:02:52.285636893 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:212:_get_merged_memory: buffer 0x7f3934017c00, idx 0, > length 1 > 0:02:52.285641867 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:1720:gst_buffer_map_range: buffer 0x7f3934017270, idx 0, > length -1, flags 0001 > 0:02:52.285647069 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:212:_get_merged_memory: buffer 0x7f3934017270, idx 0, > length 1 > 0:02:52.285652574 5934 0x55dc774f8d40 LOG audiomixer > gstaudiomixer.c:640:gst_audiomixer_aggregate_one_buffer:<audiomixer0: > sink_0> mixing 16 bytes at offset 304 from offset 0 > 0:02:52.285790505 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1283:gst_audio_aggregator_aggregate:<audiomixer0 > :sink_0> Pad is at or after current offset: 2681440 >= 2681440 > 0:02:52.285798603 5934 0x55dc774f8d40 LOG audioaggregator > gstaudioaggregator.c:1365:gst_audio_aggregator_aggregate:<audiomixer0 > > pushing outbuf 0x7f3934017c00, timestamp 0:02:47.580000000 offset > 2681280 > 0:02:52.285806631 5934 0x55dc774f8d40 DEBUG GST_SCHEDULING > gstpad.c:4203:gst_pad_chain_data_unchecked:<alsasink0:sink> calling > chainfunction &gst_base_sink_chain with buffer buffer: > 0x7f3934017c00, pts 0:02:47.580000000, dts 99:99:99.999999999, dur > 0:00:00.010000000, size 320, offset 2681280, offset_end 2681440, > flags 0x0 > 0:02:52.285815505 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3360:gst_base_sink_chain_unlocked:<alsasink0> got times > start: 0:02:47.580000000, end: 0:02:47.590000000 > 0:02:52.285823219 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:1946:gst_base_sink_get_sync_times:<alsasink0> got times > start: 0:02:47.580000000, stop: 0:02:47.590000000, do_sync 0 > 0:02:52.285831300 5934 0x55dc774f8d40 LOG basesink > gstbasesink.c:2483:gst_base_sink_do_sync:<alsasink0> avg frame diff > 0:00:00.010000000 > 0:02:52.285837240 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3468:gst_base_sink_chain_unlocked:<alsasink0> rendering > object 0x7f3934017c00 > 0:02:52.285842645 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:1916:gst_audio_base_sink_render:<alsasink0> time > 0:02:47.580000000, start 0:00:00.000000000, samples 160 > 0:02:52.285856423 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:1953:gst_audio_base_sink_render:<alsasink0> sync- > offset +0:00:00.220000000, render-delay 0:00:00.000000000, ts-offset > +0:00:00.000000000 > 0:02:52.285866566 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2015:gst_audio_base_sink_render:<alsasink0> > running: start 0:02:47.580000000 - stop 0:02:47.590000000 > 0:02:52.285873485 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2030:gst_audio_base_sink_render:<alsasink0> > compensating for sync-offset 0:00:00.220000000 > 0:02:52.285879252 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2037:gst_audio_base_sink_render:<alsasink0> adding > base_time 1349:31:49.960166366 > 0:02:52.285888650 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:1045:gst_clock_get_internal_time:<GstSystemClock> internal > time 1349:34:41.936038218 > 0:02:52.285894620 5934 0x55dc774f8d40 DEBUG GST_CLOCK > gstclock.c:1090:gst_clock_get_time:<GstSystemClock> adjusted time > 1349:34:41.936038218 > 0:02:52.285905690 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:568:gst_audio_base_sink_get_time:<alsasink0> > processed samples: raw 72960, delay 3100, real 69860, time > 0:00:04.366250000 > 0:02:52.285913345 5934 0x55dc774f8d40 DEBUG audioclock > gstaudioclock.c:202:gst_audio_clock_get_time:<GstAudioSinkClock> > result 0:00:04.366250000, last_time 0:00:00.000000000 > 0:02:52.285921064 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:1452:gst_audio_base_sink_skew_slaving:<alsasink0> > internal 0:00:04.366250000 external 1349:34:41.936038218 cinternal > 0:00:00.000000000 cexternal 1349:34:37.567980323 > 0:02:52.285930548 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:1473:gst_audio_base_sink_skew_slaving:<alsasink0> > internal 0:00:04.366250000 external 0:00:04.368057895 skew > -0:00:00.001807895 avg -0:00:00.001807895 > 0:02:52.285940773 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2057:gst_audio_base_sink_render:<alsasink0> final > timestamps: start 0:00:00.192186043 - stop 0:00:00.202186043 > 0:02:52.285948429 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2126:gst_audio_base_sink_render:<alsasink0> no > align possible: no previous sample position known > 0:02:52.285952614 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2155:gst_audio_base_sink_render:<alsasink0> > rendering at 3074 160/160 > 0:02:52.285957857 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:1720:gst_buffer_map_range: buffer 0x7f3934017c00, idx 0, > length -1, flags 0001 > 0:02:52.285963384 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:212:_get_merged_memory: buffer 0x7f3934017c00, idx 0, > length 1 > 0:02:52.285969094 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1503:default_commit:<audiosinkringbuffer0> write > 160 : 160 > 0:02:52.285974366 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1523:default_commit:<audiosinkringbuffer0> > pointer at 456, write to 19-68, diff -437, segtotal 20, segsize 320, > base 0 > 0:02:52.285980927 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1553:default_commit:<audiosinkringbuffer0> write > @0x7f393c0078f0 seg 19, sps 160, off 68, avail 252 > 0:02:52.285986614 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1579:default_commit: copy 252 bytes > 0:02:52.285991302 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1523:default_commit:<audiosinkringbuffer0> > pointer at 456, write to 20-0, diff -436, segtotal 20, segsize 320, > base 0 > 0:02:52.285997607 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1553:default_commit:<audiosinkringbuffer0> write > @0x7f393c006130 seg 0, sps 160, off 0, avail 68 > 0:02:52.286003049 5934 0x55dc774f8d40 DEBUG ringbuffer > gstaudioringbuffer.c:1579:default_commit: copy 68 bytes > 0:02:52.286007709 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2166:gst_audio_base_sink_render:<alsasink0> wrote > 160 of 160 > 0:02:52.286012918 5934 0x55dc774f8d40 DEBUG audiobasesink > gstaudiobasesink.c:2198:gst_audio_base_sink_render:<alsasink0> next > sample expected at 3234 > 0:02:52.286017810 5934 0x55dc774f8d40 DEBUG basesink > gstbasesink.c:3507:gst_base_sink_chain_unlocked:<alsasink0> object > unref after render 0x7f3934017c00 > 0:02:52.286022216 5934 0x55dc774f8d40 LOG GST_BUFFER > gstbuffer.c:724:_gst_buffer_free: finalize 0x7f3934017c00 > 0:02:52.286026444 5934 0x55dc774f8d40 DEBUG GST_MEMORY > gstmemory.c:87:_gst_memory_free: free memory 0x7f393c012260 > > > Any ideas why this delay occurs ? > Ajit. > > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.freedesktop.org" class="gmail_msg" target="_blank">gstreamer-devel@lists.freedesktop.org</a> > <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" class="gmail_msg" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a>_______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" class="gmail_msg" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" class="gmail_msg" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div></div></div></div>