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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20160527/f9eae3db/attachment-0001.html>
