<div dir="ltr"><div><div>Arun,<br><br></div>Sorry for the delay.<br><br></div><div>Thanks for the answers.<br><br>"I'm not sure I follow correctly -- you're saying that if you do things<br> this way, the AEC works fine?"<br><br></div><div>Yes, I follow correcty your instruction is working very good.<br><br></div><div><br><br></div><div>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.<br>I'm having difficult to configure the pulseaudio.<br><br><br></div><div>Thank you very much.<br></div><div><br>The gstreamer log:<br><br>root@colibri-imx6-jfl:~/wavs# ./teste_pulseaudio_echo.sh input.48.wav <br>Setting pipeline to PAUSED ...<br>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<br>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<br>Pipeline is live and does not need PREROLL ...<br>Setting pipeline to PLAYING ...<br>New clock: GstPulseSrcClock<br>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<br>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<br>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<br>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<br>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<br>^Chandling interrupt.<br>Interrupt: Stopping pipeline ...<br>Execution ended after 0:00:04.584419001<br>Setting pipeline to PAUSED ...<br>Setting pipeline to READY ...<br>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<br>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<br>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<br>Setting pipeline to NULL ...<br>Freeing pipeline ...<br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-05-18 0:45 GMT-03:00 Arun Raghavan <span dir="ltr"><<a href="mailto:arun@arunraghavan.net" target="_blank">arun@arunraghavan.net</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Caio,<br> <span class=""><br> On Sat, 14 May 2016, at 02:19 AM, Caio wrote:<br> > Arun,<br> ><br> > Your test works very well! tks!<br> > When I listen the cancelled.wav The audio start very low and after the<br> > voice is very very low and some times disapers, now it's time to fix the<br> > mechanical part (speaker / mic) and mic volume.<br> <br> </span>The AGC should kick in and fix that eventually.<br> <div><div class="h5"><br> > The audio is cracking , but I think there's some problem in my<br> > configuration (I will need help to fix it).<br> ><br> ><br> > The log.<br> ><br> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Doing resync<br> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback after capture (-1638515), drop<br> > sink 104884<br> > Jan 01 02:52:43 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (146059), drop<br> > source 9346<br> > Jan 01 02:52:44 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (13025), drop<br> > source<br> > 832<br> > Jan 01 02:52:45 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10100), drop<br> > source<br> > 646<br> > Jan 01 02:52:46 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (9965), drop<br> > source<br> > 636<br> > Jan 01 02:52:47 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10089), drop<br> > source<br> > 644<br> > Jan 01 02:52:48 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback too far ahead (10024), drop<br> > source<br> > 640<br> > Jan 01 02:52:49 colibri-imx6-jfl pulseaudio[1051]: E: [alsa-source-HiFi<br> > wm8962-0] module-echo-cancel.c: Playback after capture (-34), drop sink<br> > 22<br> ><br> ><br> > This variable PULSE_PROP="filter.want=echo-cancel" fixed the problem.<br> > When I was testing just like I wrote I had problems with echo.<br> <br> </div></div>I'm not sure I follow correctly -- you're saying that if you do things<br> this way, the AEC works fine?<br> <span class=""><br> > pactl load-module module-echo-cancel source_name=echosource<br> > sink_name=echosink aec_method=webrtc sink_master=mastersink<br> > source_master=mastersource<br> > gst-launch-1.0 filesrc location=some-voice-file.wav ! wavparse !<br> > pulsesink device="echosink" \<br> > pulsesrc device="echosource" ! wavenc ! filesink<br> <br> </span>Can you confirm that the streams were actually connected to those<br> devices? Look at the output of pactl list sink-inputs and pactl list<br> source-outputs.<br> <span class=""><br> > How I can configure webrtc or speex using enviroment vars (PULSE_PROP)?<br> <br> </span>You can't configure this via the environment. With filter.want, the<br> default which is compiled in is used (webrtc if we have it, else speex).<br> You can only control this if you manually load the module as you ahve<br> above.<br> <div class="HOEnZb"><div class="h5"><br> -- Arun<br> _______________________________________________<br> pulseaudio-discuss mailing list<br> <a href="mailto:pulseaudio-discuss@lists.freedesktop.org">pulseaudio-discuss@lists.freedesktop.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss</a><br> </div></div></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>----------------------------------------------<br>Caio Pereira<br></div><div>@caio_rtp<br></div></div></div> </div>