I forgot the pulseaudio log. Jan 01 05:39:46 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Doing resync Jan 01 05:39:46 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Playback after capture (-1705478), drop sink 109170 Jan 01 05:39:46 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Playback too far ahead (261821), drop source 16756 Jan 01 05:39:48 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Playback after capture (-168495), drop sink 10802 Jan 01 05:48:55 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Doing resync Jan 01 05:48:55 colibri-imx6 pulseaudio[1230]: E: [alsa-source-HiFi wm8962-0] module-echo-cancel.c: Playback after capture (-2897082), drop sink 185432 2016-05-27 15:28 GMT-03:00 Caio <caiortp at gmail.com>: > Arun, > > Sorry for the delay. > > Thanks for the answers. > > "I'm not sure I follow correctly -- you're saying that if you do things > this way, the AEC works fine?" > > Yes, I follow correcty your instruction is working very good. > > > > But Now I'm trying to avoid the "audio underrun" (the audio is cracking). > When I use the paplay the audio is ok and the cpu is around 14%, when I > start the test the cpu goes to 26% and the audio start to cracking. > I'm having difficult to configure the pulseaudio. > > > Thank you very much. > > The gstreamer log: > > root at colibri-imx6-jfl:~/wavs# ./teste_pulseaudio_echo.sh input.48.wav > Setting pipeline to PAUSED ... > 0:00:00.093887667 1258 0x1bac100 WARN basesrc > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0/1.4.1-r0/gstreamer-1.4.1/libs/gst/base/gstbasesrc.c:3460:gst_base_src_start_complete:<filesrc0> > pad not activated yet > 0:00:00.094892667 1258 0x1bb5750 FIXME default > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0/1.4.1-r0/gstreamer-1.4.1/gst/gstutils.c:3643:gst_pad_create_stream_id_internal:<wavparse0:src> > Creating random stream-id, consider implementing a deterministic way of > creating a stream-id > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstPulseSrcClock > 0:00:00.098627667 1258 0x1bb5780 FIXME default > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0/1.4.1-r0/gstreamer-1.4.1/gst/gstutils.c:3643:gst_pad_create_stream_id_internal:<pulsesrc0:src> > Creating random stream-id, consider implementing a deterministic way of > creating a stream-id > 0:00:01.350294000 1258 0x1bb5750 WARN audiobasesink > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-mx6qdl-angstrom-linux-gnueabi/gstreamer1.0-plugins-base/1.4.1-r0/gst-plugins-base-1.4.1/gst-libs/gst/audio/gstaudiobasesink.c:1288:gst_audio_base_sink_skew_slaving:<pulsesink0> > correct clock skew 22728553 > 20000000 > 0:00:01.904267667 1258 0x1bb5750 WARN audiobasesink > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-mx6qdl-angstrom-linux-gnueabi/gstreamer1.0-plugins-base/1.4.1-r0/gst-plugins-base-1.4.1/gst-libs/gst/audio/gstaudiobasesink.c:1288:gst_audio_base_sink_skew_slaving:<pulsesink0> > correct clock skew 21428670 > 20000000 > 0:00:03.019438001 1258 0x1bb5750 WARN audiobasesink > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-mx6qdl-angstrom-linux-gnueabi/gstreamer1.0-plugins-base/1.4.1-r0/gst-plugins-base-1.4.1/gst-libs/gst/audio/gstaudiobasesink.c:1288:gst_audio_base_sink_skew_slaving:<pulsesink0> > correct clock skew 20980685 > 20000000 > 0:00:03.947815667 1258 0x1bcb5a0 WARN pulse > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0-plugins-good/1.4.1-r0/gst-plugins-good-1.4.1/ext/pulse/pulsesink.c:678:gst_pulsering_stream_underflow_cb:<pulsesink0> > Got underflow > ^Chandling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 0:00:04.584419001 > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > 0:00:04.692690334 1258 0x1bc8430 WARN audiosrc > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-mx6qdl-angstrom-linux-gnueabi/gstreamer1.0-plugins-base/1.4.1-r0/gst-plugins-base-1.4.1/gst-libs/gst/audio/gstaudiosrc.c:244:audioringbuffer_thread_func:<pulsesrc0> > error reading data -1 (reason: Success), skipping segment > 0:00:04.694879001 1258 0x1bac100 WARN wavenc > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0-plugins-good/1.4.1-r0/gst-plugins-good-1.4.1/gst/wavenc/gstwavenc.c:922:gst_wavenc_change_state:<wavenc0> > warning: Wav stream not finished properly > 0:00:04.694963334 1258 0x1bac100 WARN wavenc > /home/icchw/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-angstrom-linux-gnueabi/gstreamer1.0-plugins-good/1.4.1-r0/gst-plugins-good-1.4.1/gst/wavenc/gstwavenc.c:922:gst_wavenc_change_state:<wavenc0> > warning: Wav stream not finished properly, no EOS received before shutdown > Setting pipeline to NULL ... > Freeing pipeline ... > > > 2016-05-18 0:45 GMT-03:00 Arun Raghavan <arun at arunraghavan.net>: > >> Hi Caio, >> >> On Sat, 14 May 2016, at 02:19 AM, Caio wrote: >> > Arun, >> > >> > Your test works very well! tks! >> > When I listen the cancelled.wav The audio start very low and after the >> > voice is very very low and some times disapers, now it's time to fix the >> > mechanical part (speaker / mic) and mic volume. >> >> The AGC should kick in and fix that eventually. >> >> > The audio is cracking , but I think there's some problem in my >> > configuration (I will need help to fix it). >> > >> > >> > The log. >> > >> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Doing resync >> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback after capture (-1638515), drop >> > sink 104884 >> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (146059), drop >> > source 9346 >> > Jan 01 02:52:44 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (13025), drop >> > source >> > 832 >> > Jan 01 02:52:45 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10100), drop >> > source >> > 646 >> > Jan 01 02:52:46 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (9965), drop >> > source >> > 636 >> > Jan 01 02:52:47 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10089), drop >> > source >> > 644 >> > Jan 01 02:52:48 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10024), drop >> > source >> > 640 >> > Jan 01 02:52:49 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi >> > wm8962-0] module-echo-cancel.c: Playback after capture (-34), drop sink >> > 22 >> > >> > >> > This variable PULSE_PROP="filter.want=echo-cancel" fixed the >> problem. >> > When I was testing just like I wrote I had problems with echo. >> >> I'm not sure I follow correctly -- you're saying that if you do things >> this way, the AEC works fine? >> >> > pactl load-module module-echo-cancel source_name=echosource >> > sink_name=echosink aec_method=webrtc sink_master=mastersink >> > source_master=mastersource >> > gst-launch-1.0 filesrc location=some-voice-file.wav ! wavparse ! >> > pulsesink device="echosink" \ >> > pulsesrc device="echosource" ! wavenc ! filesink >> >> Can you confirm that the streams were actually connected to those >> devices? Look at the output of pactl list sink-inputs and pactl list >> source-outputs. >> >> > How I can configure webrtc or speex using enviroment vars (PULSE_PROP)? >> >> You can't configure this via the environment. With filter.want, the >> default which is compiled in is used (webrtc if we have it, else speex). >> You can only control this if you manually load the module as you ahve >> above. >> >> -- Arun >> _______________________________________________ >> pulseaudio-discuss mailing list >> pulseaudio-discuss at lists.freedesktop.org >> https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss >> > > > > -- > ---------------------------------------------- > Caio Pereira > @caio_rtp > -- ---------------------------------------------- Caio Pereira @caio_rtp -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20160527/0e557d23/attachment.html>
