WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze
Locked
 
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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 _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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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 > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Re: WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze
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In reply to this post by Jack
Those tap warnigns you can neglect.
Have you put queue after the tee ?? Below pipekine is working for me. I did not have pulse and other sink so replaced with fakesink. gst-launch-1.0 audiotestsrc wave=sine is-live=true ! audioconvert ! audioresample ! queue ! audiomixer ! tee name=t t.! queue ! audioconvert ! audioresample ! fakesink t.! queue ! volume ! audiopanorama ! audioconvert ! audioresample ! fakesink --gst-debug=2 Setting pipeline to PAUSED ... 0:00:00.074289300 20184 0000000002EC1CC0 WARN aggregator gstaggregator.c:1901:gst_aggregator_query_latency_unlocked:<audiomixer0> Latency query failed Pipeline is live and does not need PREROLL ... 0:00:00.077887600 20184 0000000002EC1D00 WARN audio-resampler audio-resampler.c:275:convert_taps_gint32_c: can't find exact taps Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock 0:00:00.095252200 20184 0000000002EC1C40 WARN audio-resampler audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps 0:00:00.095264900 20184 0000000002EC1B40 WARN audio-resampler audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps handling interrupt.9. Interrupt: Stopping pipeline ... Execution ended after 0:00:07.261243500 Setting pipeline to NULL ... Freeing pipeline ... -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hello Vinod,
Yes the Bin I add to the "tee" start with a "queue". Here the Bin : queue ! volume ! audiopanorama ! audioconvert ! audioresample ! shout2send Do you have an idea ? ++ Jack Le 23/07/2020 à 08:06, Vinod Kesti a écrit : > Those tap warnigns you can neglect. > Have you put queue after the tee ?? > > Below pipekine is working for me. I did not have pulse and other sink so > replaced with fakesink. > gst-launch-1.0 audiotestsrc wave=sine is-live=true ! audioconvert ! > audioresample ! queue ! audiomixer ! tee name=t t.! queue ! audioconvert ! > audioresample ! fakesink t.! queue ! volume ! audiopanorama ! audioconvert ! > audioresample ! fakesink --gst-debug=2 > Setting pipeline to PAUSED ... > 0:00:00.074289300 20184 0000000002EC1CC0 WARN aggregator > gstaggregator.c:1901:gst_aggregator_query_latency_unlocked:<audiomixer0> > Latency query failed > Pipeline is live and does not need PREROLL ... > 0:00:00.077887600 20184 0000000002EC1D00 WARN audio-resampler > audio-resampler.c:275:convert_taps_gint32_c: can't find exact taps > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > 0:00:00.095252200 20184 0000000002EC1C40 WARN audio-resampler > audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps > 0:00:00.095264900 20184 0000000002EC1B40 WARN audio-resampler > audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps > handling interrupt.9. > Interrupt: Stopping pipeline ... > Execution ended after 0:00:07.261243500 > Setting pipeline to NULL ... > Freeing pipeline ... > > > > > -- > Sent from: http://gstreamer-devel.966125.n4.nabble.com/ > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hum, even with :
audiotestsrc wave=sine is-live=false I can't listen anymore my sine when I add the Bin to the pipeline. This pipeline looks simple, then why my sine cut when I add this Bin ? That is the question. Any proposal is welcome ! ++ Jack Le 23/07/2020 à 09:54, Jack a écrit : > Hello Vinod, > > Yes the Bin I add to the "tee" start with a "queue". Here the Bin : > > queue ! volume ! audiopanorama ! audioconvert ! audioresample ! shout2send > > Do you have an idea ? > ++ > > Jack > > > > Le 23/07/2020 à 08:06, Vinod Kesti a écrit : >> Those tap warnigns you can neglect. >> Have you put queue after the tee ?? >> >> Below pipekine is working for me. I did not have pulse and other sink so >> replaced with fakesink. >> gst-launch-1.0 audiotestsrc wave=sine is-live=true ! audioconvert ! >> audioresample ! queue ! audiomixer ! tee name=t t.! queue ! audioconvert ! >> audioresample ! fakesink t.! queue ! volume ! audiopanorama ! audioconvert ! >> audioresample ! fakesink --gst-debug=2 >> Setting pipeline to PAUSED ... >> 0:00:00.074289300 20184 0000000002EC1CC0 WARN aggregator >> gstaggregator.c:1901:gst_aggregator_query_latency_unlocked:<audiomixer0> >> Latency query failed >> Pipeline is live and does not need PREROLL ... >> 0:00:00.077887600 20184 0000000002EC1D00 WARN audio-resampler >> audio-resampler.c:275:convert_taps_gint32_c: can't find exact taps >> Pipeline is PREROLLED ... >> Setting pipeline to PLAYING ... >> New clock: GstSystemClock >> 0:00:00.095252200 20184 0000000002EC1C40 WARN audio-resampler >> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps >> 0:00:00.095264900 20184 0000000002EC1B40 WARN audio-resampler >> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps >> handling interrupt.9. >> Interrupt: Stopping pipeline ... >> Execution ended after 0:00:07.261243500 >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> >> >> >> -- >> Sent from: http://gstreamer-devel.966125.n4.nabble.com/ >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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There was a problem with the samplerate in the pipeline :
audiotestsrc wave=sine is-live=true ! audioconvert ! audioresample ! queue ! audiomixer ! tee ! queue ! audioconvert ! audioresample ! pulsesink I change the value and now the pipeline is OK when I add the Bin : queue ! volume ! audiopanorama ! audioconvert ! audioresample ! shout2send and connect it to the "tee" element : no more freeze :) ++ Jack Le 23/07/2020 à 19:05, Jack a écrit : > Hum, even with : > > audiotestsrc wave=sine is-live=false > > I can't listen anymore my sine when I add the Bin to the pipeline. > > This pipeline looks simple, then why my sine cut when I add this Bin ? > That is the question. > > Any proposal is welcome ! > > ++ > > Jack > > > > Le 23/07/2020 à 09:54, Jack a écrit : >> Hello Vinod, >> >> Yes the Bin I add to the "tee" start with a "queue". Here the Bin : >> >> queue ! volume ! audiopanorama ! audioconvert ! audioresample ! shout2send >> >> Do you have an idea ? >> ++ >> >> Jack >> >> >> >> Le 23/07/2020 à 08:06, Vinod Kesti a écrit : >>> Those tap warnigns you can neglect. >>> Have you put queue after the tee ?? >>> >>> Below pipekine is working for me. I did not have pulse and other sink so >>> replaced with fakesink. >>> gst-launch-1.0 audiotestsrc wave=sine is-live=true ! audioconvert ! >>> audioresample ! queue ! audiomixer ! tee name=t t.! queue ! audioconvert ! >>> audioresample ! fakesink t.! queue ! volume ! audiopanorama ! audioconvert ! >>> audioresample ! fakesink --gst-debug=2 >>> Setting pipeline to PAUSED ... >>> 0:00:00.074289300 20184 0000000002EC1CC0 WARN aggregator >>> gstaggregator.c:1901:gst_aggregator_query_latency_unlocked:<audiomixer0> >>> Latency query failed >>> Pipeline is live and does not need PREROLL ... >>> 0:00:00.077887600 20184 0000000002EC1D00 WARN audio-resampler >>> audio-resampler.c:275:convert_taps_gint32_c: can't find exact taps >>> Pipeline is PREROLLED ... >>> Setting pipeline to PLAYING ... >>> New clock: GstSystemClock >>> 0:00:00.095252200 20184 0000000002EC1C40 WARN audio-resampler >>> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps >>> 0:00:00.095264900 20184 0000000002EC1B40 WARN audio-resampler >>> audio-resampler.c:274:convert_taps_gint16_c: can't find exact taps >>> handling interrupt.9. >>> Interrupt: Stopping pipeline ... >>> Execution ended after 0:00:07.261243500 >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> >>> >>> >>> -- >>> Sent from: http://gstreamer-devel.966125.n4.nabble.com/ >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >>> >> >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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