WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze

Jack jack at rybn.org
Wed Jul 22 23:24:17 UTC 2020


Re,

At debug level 4, I get a FIXME (if it can help to solve this issue) :

0:00:41.789219531 20253 0x5557d5315a30 FIXME               basesink
gstbasesink.c:3270:gst_base_sink_default_event:<shout2send0>
stream-start event without group-id. Consider implementing group-id
handling in the upstream elements


Here, a part of the logs with the FIXME :

:00:41.786198201 20253 0x5557d52794a0 WARN         audio-resampler
audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
0:00:41.786244237 20253 0x5557d52794a0 WARN         audio-resampler
audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
0:00:41.786284717 20253 0x5557d52794a0 WARN         audio-resampler
audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
0:00:41.786318412 20253 0x5557d52794a0 INFO         audio-converter
audio-converter.c:962:chain_quantize: depth in 16, out 16
0:00:41.786324134 20253 0x5557d52794a0 INFO         audio-converter
audio-converter.c:974:chain_quantize: using no dither and noise shaping
0:00:41.786331009 20253 0x5557d52794a0 INFO         audio-converter
audio-converter.c:1032:chain_pack: pack format S16LE to S16LE
0:00:41.786335748 20253 0x5557d52794a0 INFO         audio-converter
audio-converter.c:1393:gst_audio_converter_new: same formats, and
passthrough mixing -> only resampling
0:00:41.786842674 20253 0x5557d5315a30 INFO               structure
gststructure.c:2634:gst_structure_get_valist: Expected field
'channel-mask' in structure: audio/x-raw, format=(string)S16LE,
rate=(int)48000, channels=(int)2, layout=(string)interleaved;
0:00:41.786876331 20253 0x5557d5279450 WARN           audiobasesink
gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<sink> correct
clock skew -0:00:00.030115297 < -+0:00:00.020000000
0:00:41.787167764 20253 0x5557d5315a30 INFO                 opusenc
gstopusenc.c:531:gst_opus_enc_setup_channel_mappings:<opusenc0> Stereo,
trivial RTP mapping
0:00:41.787194573 20253 0x5557d5315a30 INFO                 opusenc
gstopusenc.c:709:gst_opus_enc_setup:<opusenc0> Mapping tables built: 2
channels, 1 stereo streams
0:00:41.787209619 20253 0x5557d5315a30 INFO                 opusenc
gstopuscommon.c:109:gst_opus_common_log_channel_mapping_table:<opusenc0>
Encoding mapping table: [ 0 1 ]
0:00:41.787216359 20253 0x5557d5315a30 INFO                 opusenc
gstopuscommon.c:109:gst_opus_common_log_channel_mapping_table:<opusenc0>
Decoding mapping table: [ 0 1 ]
0:00:41.787582502 20253 0x5557d5315a30 INFO               GST_EVENT
gstevent.c:820:gst_event_new_caps: creating caps event audio/x-opus,
rate=(int)48000, channels=(int)2, channel-mapping-family=(int)0,
stream-count=(int)1, coupled-count=(int)1, streamheader=(buffer)<
4f707573486561640102380180bb0000000000,
4f707573546167731e000000456e636f6465642077697468204753747265616d6572206f707573656e630000000001
>
0:00:41.789219531 20253 0x5557d5315a30 FIXME               basesink
gstbasesink.c:3270:gst_base_sink_default_event:<shout2send0>
stream-start event without group-id. Consider implementing group-id
handling in the upstream elements
0:00:41.789305991 20253 0x5557d5315a30 INFO                oggdemux
gstoggstream.c:2734:gst_ogg_stream_setup_map_from_caps_headers: Checking
streamheader on caps audio/x-opus, rate=(int)48000, channels=(int)2,
channel-mapping-family=(int)0, stream-count=(int)1,
coupled-count=(int)1, streamheader=(buffer)<
4f707573486561640102380180bb0000000000,
4f707573546167731e000000456e636f6465642077697468204753747265616d6572206f707573656e630000000001
>
0:00:41.789340686 20253 0x5557d5315a30 INFO                oggdemux
gstoggstream.c:2780:gst_ogg_stream_setup_map_from_caps_headers: Found
headers on caps, using those to determine type
0:00:41.789351256 20253 0x5557d5315a30 INFO                oggdemux
gstoggstream.c:2092:setup_opus_mapper: Opus has a pre-skip of 312 samples
0:00:41.790075375 20253 0x5557d5315a30 INFO               GST_EVENT
gstevent.c:820:gst_event_new_caps: creating caps event application/ogg,
streamheader=(buffer)<
4f67675300020000000000000000329c18360000000043436c8101134f707573486561640102380180bb0000000000,
4f67675300000000000000000000329c183601000000d80e4c3b012f4f707573546167731e000000456e636f6465642077697468204753747265616d6572206f707573656e630000000001
>
0:00:41.790156057 20253 0x5557d5315a30 INFO               GST_EVENT
gstevent.c:901:gst_event_new_segment: creating segment event time
segment start=0:00:00.000000000, offset=0:00:00.000000000,
stop=99:99:99.999999999, rate=1.000000, applied_rate=1.000000,
flags=0x00, time=0:00:00.000000000, base=0:00:00.000000000, position
0:00:00.000000000, duration 99:99:99.999999999
0:00:41.790211717 20253 0x5557d5315a30 INFO              GST_STATES
gstbin.c:3421:bin_handle_async_done:<bin_0> committing state from READY
to PAUSED, old pending PLAYING
0:00:41.790221923 20253 0x5557d5315a30 INFO              GST_STATES
gstbin.c:3441:bin_handle_async_done:<bin_0> completed state change,
pending VOID
0:00:41.790232708 20253 0x5557d5315a30 INFO              GST_STATES
gstelement.c:2660:_priv_gst_element_state_changed:<bin_0> notifying
about state-changed READY to PAUSED (VOID_PENDING pending)
0:00:41.790250987 20253 0x5557d5315a30 INFO              GST_STATES
gstbin.c:3421:bin_handle_async_done:<mainpipeline> committing state from
PAUSED to PAUSED, old pending PLAYING

++

Jack




Le 22/07/2020 à 23:36, Jack a écrit :
> Hello,
> 
> I have a playing PIPELINE build with Python :
> 
> audiotestsrc wave=sine is-live=true ! audioconvert ! audioresample !
> queue ! audiomixer ! tee ! queue ! audioconvert ! audioresample ! pulsesink
> 
> And I can listen the sound produced by "audiotestsrc" with is-live=true.
> So the PIPELINE is working properly.
> 
> But, when I connect a Bin to the "tee" element, I get a lot of warning
> with the message "convert_taps_gint16_c: can't find exact taps" and the
> pipeline freeze.
> 
> The Bin connected to the "tee" element is (I don't put the properties of
> "shout2send" here but the configuration is OK) :
> 
> queue ! volume ! audiopanorama ! audioconvert ! audioresample ! shout2send
> 
> Then I use this code to connect the "tee" source pad to the Bin sink
> ghostpad :
> 
> self.ch_pad = PIPELINE.tee.get_request_pad('src_%u')
> self.ch_pad.link(self.get_static_pad("sink"))
> self.set_state(Gst.State.PLAYING)
> 
> My configuration :
> Ubuntu 18.04
> GStreamer 1.17.0 (GIT)
> Python 3.7
> 
> 
> Maybe gstreamer gurus can help me quickly to solve this problem ?
> Best!
> ++
> 
> Jack
> 
> 
> The log with "WARN" messages :
> 
> 0:00:35.420044895 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.420085966 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.420121114 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.421678578 26903 0x7fd6cc009c00 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.422157037 26903 0x7fd6a4002940 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.513347981 26903 0x55924d620850 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.513574203 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.513634501 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.513683464 26903 0x55924d620800 WARN         audio-resampler
> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps
> 0:00:35.513840570 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.094561258 < -+0:00:00.020000000
> 0:00:36.781925073 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.039647889 < -+0:00:00.020000000
> 0:00:36.782056563 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.038393044 < -+0:00:00.020000000
> 0:00:36.782078111 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.037194009 < -+0:00:00.020000000
> 0:00:36.782242838 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.036036641 < -+0:00:00.020000000
> 0:00:36.782258107 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.034911235 < -+0:00:00.020000000
> 0:00:36.782305295 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.033821672 < -+0:00:00.020000000
> 0:00:36.782347019 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.032766027 < -+0:00:00.020000000
> 0:00:36.782423489 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.031744493 < -+0:00:00.020000000
> 0:00:36.782463233 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.030753779 < -+0:00:00.020000000
> 0:00:36.782552115 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.029795436 < -+0:00:00.020000000
> 0:00:36.782566567 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.028864880 < -+0:00:00.020000000
> 0:00:36.782720108 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.027967339 < -+0:00:00.020000000
> 0:00:36.782749513 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.027094454 < -+0:00:00.020000000
> 0:00:36.782874796 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.026251637 < -+0:00:00.020000000
> 0:00:36.782898129 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.025432083 < -+0:00:00.020000000
> 0:00:36.782949734 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.024638925 < -+0:00:00.020000000
> 0:00:36.783174104 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.023875883 < -+0:00:00.020000000
> 0:00:36.783192484 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.023130495 < -+0:00:00.020000000
> 0:00:36.783282290 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.022410440 < -+0:00:00.020000000
> 0:00:36.783370287 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.021712858 < -+0:00:00.020000000
> 0:00:36.783415671 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.021035786 < -+0:00:00.020000000
> 0:00:36.800458440 26903 0x55924d670f20 WARN            audiobasesrc
> gstaudiobasesrc.c:840:gst_audio_base_src_create:<input_audio_6> create
> DISCONT of 13752 samples at sample 1622736
> 0:00:36.800478927 26903 0x55924d670f20 WARN            audiobasesrc
> gstaudiobasesrc.c:845:gst_audio_base_src_create:<input_audio_6> warning:
> Can't record audio fast enough
> 0:00:36.800483241 26903 0x55924d670f20 WARN            audiobasesrc
> gstaudiobasesrc.c:845:gst_audio_base_src_create:<input_audio_6> warning:
> Dropped 13752 samples. This is most likely because downstream can't keep
> up and is consuming samples too slowly.
> 0:00:37.301391183 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.036563800 < -+0:00:00.020000000
> 0:00:37.301470584 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.035424539 < -+0:00:00.020000000
> 0:00:37.302518433 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.034349393 < -+0:00:00.020000000
> 0:00:37.302973096 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.033290506 < -+0:00:00.020000000
> 0:00:37.303043088 26903 0x7fd6cc009c00 WARN           audiobasesink
> gstaudiobasesink.c:1491:gst_audio_base_sink_skew_slaving:<output_audio_7>
> correct clock skew -0:00:00.032252949 < -+0:00:00.020000000
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