High initial playback delay when using audiomixer

Wed Nov 30 01:37:14 UTC 2016

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) -4195220096
> 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
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
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