SB
Severin Birrer
Tue, Aug 30, 2016 8:02 AM
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 (sip:203@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:203@192.168.139.152;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto:severin.birrer@fela.ch
http://www.fela.ch/
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
1. Get the depot_tools
2. Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
3. Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 (sip:203@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:203@192.168.139.152;user=phone)
-----------------------------------------------------------------------------------
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto:severin.birrer@fela.ch
http://www.fela.ch/
RS
Riza Sulistyo
Tue, Aug 30, 2016 9:08 AM
I'm trying to get some better Results with the AEC. (pjproject Version
2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time
(one minute or so) the background noise gets louder and I think the
cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in
Ticket #1888). But no luck there to compile it in, because of a missing
full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio,
system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which
Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_
ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|
PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic)
PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000,
channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 (sip:203@192.168.139.152;user=phone)
transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000)
transmitting to port 2 (sip:203@192.168.139.152;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto:severin.birrer@fela.ch
http://www.fela.ch/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Severin,
We have added webrtc aec on #1954 <https://trac.pjsip.org/repos/ticket/1954>
.
Maybe you can try that.
For tweaking purposes, the information on #1888
<https://trac.pjsip.org/repos/ticket/1888> is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer <severin.birrer@fela.ch>
wrote:
> I'm trying to get some better Results with the AEC. (pjproject Version
> 2.5.5)
> I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
> The cancelling works, but you still get a little echo and after some time
> (one minute or so) the background noise gets louder and I think the
> cancelling gets also worse.
> Do I have to set some more parameters than the ecTailLen and ecOptions?
>
> The other Thing i tried is to integrate WebRTC AEC (Mostly described in
> Ticket #1888). But no luck there to compile it in, because of a missing
> full-howto.
> Steps done so far with WebRTC:
> 1. Get the depot_tools
> 2. Get/Build webrtc:
> mkdir webrtc-checkout
> cd webrtc-checkout
> fetch --nohooks webrtc
> gclient sync
> gn gen out/Release
> ninja -C out/Release
> 3. Configure pjsip:
> ./configure --with-webrtc=~/webrtc-checkout
>
> Log output of configure:
> Using webrtc prefix... /home/yocto/webrtc-checkout
> checking for WebRtcAec_Process in -laudio_processing... no
>
>
> I think the webrtc aec libraries (audio_processing, common_audio,
> system_wrappers) doesn't get built.
> Is it due to the compatibility issue described in the ticket? Which
> Version do I have to get, or more, how do I get it?
>
>
> My setup:
> Buildsystem Ubuntu
> Target Yocto Linux on a i.MX6 (ARM)
> Loudspeaker & MIC
> SGTL5000 Audiocodec
> Latency: about 200ms (measured with pjsystest Latency Test)
>
> My initialisation routine of EC is:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=300;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_
> ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|
> PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
> ep.libInit( ep_cfg );
>
> Log output of SPEEX AEC:
> 10:37:54.669 pjsua_call.c ..Answering call 0: code=200
> 10:37:54.670 pjsua_media.c .....Call 0: updating media..
> 10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
> 10:37:54.672 pjsua_aud.c ......Audio channel update..
> 10:37:54.674 strm0x74d1452c .......Encoder stream started
> 10:37:54.674 strm0x74d1452c .......Decoder stream started
> 10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
> 10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
> 10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
> 10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic)
> PCM@16000/1/20ms
> 10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000,
> channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
> 10:37:54.715 conference.c ......Port 2 (sip:203@192.168.139.152;user=phone)
> transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
> 10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
> 10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000)
> transmitting to port 2 (sip:203@192.168.139.152;user=phone)
>
> ------------------------------------------------------------
> -----------------------
> Severin Birrer
>
> FELA Management AG Phone : +41 52 646 46 37
> Basadingerstrasse 18 Fax : +41 52 646 46 96
> CH-8253 Diessenhofen Mail : mailto:severin.birrer@fela.ch
> http://www.fela.ch/
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
SB
Severin Birrer
Wed, Aug 31, 2016 12:11 PM
Hi Riza,
With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
Do I miss something?
Kind Regards,
Severin
Config:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=25;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output:
03:38:48.615 pjsua_call.c ..Answering call 0: code=200
03:38:48.616 pjsua_media.c .....Call 0: updating media..
03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
03:38:48.617 pjsua_aud.c ......Audio channel update..
03:38:48.619 strm0x74e1452c .......Encoder stream started
03:38:48.620 strm0x74e1452c .......Decoder stream started
03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
----- Ursprüngliche Mail -----
Von: "Riza Sulistyo" riza@pjsip.org
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Dienstag, 30. August 2016 11:08:45
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
We have added webrtc aec on #1954 .
Maybe you can try that.
For tweaking purposes, the information on #1888 is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
http://www.fela.ch/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Riza,
With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
Do I miss something?
Kind Regards,
Severin
Config:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=25;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output:
03:38:48.615 pjsua_call.c ..Answering call 0: code=200
03:38:48.616 pjsua_media.c .....Call 0: updating media..
03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
03:38:48.617 pjsua_aud.c ......Audio channel update..
03:38:48.619 strm0x74e1452c .......Encoder stream started
03:38:48.620 strm0x74e1452c .......Decoder stream started
03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
*****
03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
----- Ursprüngliche Mail -----
Von: "Riza Sulistyo" <riza@pjsip.org>
An: "pjsip" <pjsip@lists.pjsip.org>
Gesendet: Dienstag, 30. August 2016 11:08:45
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
We have added webrtc aec on #1954 .
Maybe you can try that.
For tweaking purposes, the information on #1888 is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
1. Get the depot_tools
2. Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
3. Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
-----------------------------------------------------------------------------------
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
http://www.fela.ch/
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
BG
Bill Gardner
Wed, Aug 31, 2016 1:46 PM
Hi Severin,
It's very misleading that PSJIP passes the echo tail length to WebRTC.
WebRTC wants an estimate of the latency, not the echo tail length. If
you pass in too high a value, the echo cancellation will not work. I
think PJSIP implemented it this way because one can use the echo tail
parameter to pass a command line value all the way down to the AEC.
The mobile AEC does not have the automatic latency estimator, but it can
handle a much larger variation in latency and therefore does not need
an accurate estimate. I'd recommend using the mobile AEC and passing 0
as the latency estimate (in place of echo tail).
If not using mobile AEC, I'd still recommend using 0 as the latency
estimate. Note that the PJSIP echo framework automatically compensates
for audio buffer latency and passes signals to the echo canceller that
are more or less properly aligned. The recorded audio has to be delayed
somewhat from the playback audio or echo cancellation will completely
fail. But hopefully the echo framework has removed a lot of the recorded
delay due to buffering.
Regards,
Bill
On 8/31/2016 8:11 AM, Severin Birrer wrote:
Hi Riza,
With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
Do I miss something?
Kind Regards,
Severin
Config:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=25;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output:
03:38:48.615 pjsua_call.c ..Answering call 0: code=200
03:38:48.616 pjsua_media.c .....Call 0: updating media..
03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
03:38:48.617 pjsua_aud.c ......Audio channel update..
03:38:48.619 strm0x74e1452c .......Encoder stream started
03:38:48.620 strm0x74e1452c .......Decoder stream started
03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
----- Ursprüngliche Mail -----
Von: "Riza Sulistyo" riza@pjsip.org
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Dienstag, 30. August 2016 11:08:45
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
We have added webrtc aec on #1954 .
Maybe you can try that.
For tweaking purposes, the information on #1888 is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
http://www.fela.ch/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Severin,
It's very misleading that PSJIP passes the echo tail length to WebRTC.
WebRTC wants an estimate of the latency, not the echo tail length. If
you pass in too high a value, the echo cancellation will not work. I
think PJSIP implemented it this way because one can use the echo tail
parameter to pass a command line value all the way down to the AEC.
The mobile AEC does not have the automatic latency estimator, but it can
handle a much larger variation in latency and therefore does not need
an accurate estimate. I'd recommend using the mobile AEC and passing 0
as the latency estimate (in place of echo tail).
If not using mobile AEC, I'd still recommend using 0 as the latency
estimate. Note that the PJSIP echo framework automatically compensates
for audio buffer latency and passes signals to the echo canceller that
are more or less properly aligned. The recorded audio has to be delayed
somewhat from the playback audio or echo cancellation will completely
fail. But hopefully the echo framework has removed a lot of the recorded
delay due to buffering.
Regards,
Bill
On 8/31/2016 8:11 AM, Severin Birrer wrote:
> Hi Riza,
> With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
>
> The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
> But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
> Do I miss something?
>
> Kind Regards,
> Severin
>
> Config:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=25;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
> ep.libInit( ep_cfg );
>
> Log output:
>
> 03:38:48.615 pjsua_call.c ..Answering call 0: code=200
> 03:38:48.616 pjsua_media.c .....Call 0: updating media..
> 03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
> 03:38:48.617 pjsua_aud.c ......Audio channel update..
> 03:38:48.619 strm0x74e1452c .......Encoder stream started
> 03:38:48.620 strm0x74e1452c .......Decoder stream started
> 03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
> 03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
> 03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
> 03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
> 03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
> 03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
> 03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
> 03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
> 03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
> *****
> 03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
> 03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
> 03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
> 03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
> 03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
> 03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
> 03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>
>
>
>
> ----- Ursprüngliche Mail -----
> Von: "Riza Sulistyo" <riza@pjsip.org>
> An: "pjsip" <pjsip@lists.pjsip.org>
> Gesendet: Dienstag, 30. August 2016 11:08:45
> Betreff: Re: [pjsip] AEC echo cancelling
>
> Hi Severin,
> We have added webrtc aec on #1954 .
> Maybe you can try that.
> For tweaking purposes, the information on #1888 is still relevant.
>
> Best Regards,
>
> Riza
>
> On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
>
>
> I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
> I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
> The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
> Do I have to set some more parameters than the ecTailLen and ecOptions?
>
> The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
> Steps done so far with WebRTC:
> 1. Get the depot_tools
> 2. Get/Build webrtc:
> mkdir webrtc-checkout
> cd webrtc-checkout
> fetch --nohooks webrtc
> gclient sync
> gn gen out/Release
> ninja -C out/Release
> 3. Configure pjsip:
> ./configure --with-webrtc=~/webrtc-checkout
>
> Log output of configure:
> Using webrtc prefix... /home/yocto/webrtc-checkout
> checking for WebRtcAec_Process in -laudio_processing... no
>
>
> I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
> Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
>
>
> My setup:
> Buildsystem Ubuntu
> Target Yocto Linux on a i.MX6 (ARM)
> Loudspeaker & MIC
> SGTL5000 Audiocodec
> Latency: about 200ms (measured with pjsystest Latency Test)
>
> My initialisation routine of EC is:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=300;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
> ep.libInit( ep_cfg );
>
> Log output of SPEEX AEC:
> 10:37:54.669 pjsua_call.c ..Answering call 0: code=200
> 10:37:54.670 pjsua_media.c .....Call 0: updating media..
> 10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
> 10:37:54.672 pjsua_aud.c ......Audio channel update..
> 10:37:54.674 strm0x74d1452c .......Encoder stream started
> 10:37:54.674 strm0x74d1452c .......Decoder stream started
> 10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
> 10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
> 10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
> 10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
> 10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
> 10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
> 10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
> 10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
>
> -----------------------------------------------------------------------------------
> Severin Birrer
>
> FELA Management AG Phone : +41 52 646 46 37
> Basadingerstrasse 18 Fax : +41 52 646 46 96
> CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
> http://www.fela.ch/
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
>
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
SB
Severin Birrer
Wed, Aug 31, 2016 3:23 PM
Hi Bill,
With the mobile AEC we're getting much better result's. It is now just a little noisy in the start of a call for a few seconds, but there is no echo :)
Thank you all so far^^
With "passing 0 as the latency estimate (in place of echo tail)" do you mean, i have to set the ecTailLen=0?
But then the AEC get's disabled with pjsua2.
Or do you mean set sndPlayLatency/sndRecLatency to 0.
Or how do I set the latency estimate?
Best Regards
Severin
Best Options so far:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=1;
ep_cfg.medConfig.sndRecLatency=0;
ep_cfg.medConfig.sndPlayLatency=0;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
Log:
07:00:40.169 echo_webrtc.c ........WebRTC AEC mobile successfully created with options 899
07:00:40.170 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=1 ms, latency=25 ms
----- Ursprüngliche Mail -----
Von: "Bill Gardner" billg@wavearts.com
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Mittwoch, 31. August 2016 15:46:43
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
It's very misleading that PSJIP passes the echo tail length to WebRTC.
WebRTC wants an estimate of the latency, not the echo tail length. If
you pass in too high a value, the echo cancellation will not work. I
think PJSIP implemented it this way because one can use the echo tail
parameter to pass a command line value all the way down to the AEC.
The mobile AEC does not have the automatic latency estimator, but it can
handle a much larger variation in latency and therefore does not need
an accurate estimate. I'd recommend using the mobile AEC and passing 0
as the latency estimate (in place of echo tail).
If not using mobile AEC, I'd still recommend using 0 as the latency
estimate. Note that the PJSIP echo framework automatically compensates
for audio buffer latency and passes signals to the echo canceller that
are more or less properly aligned. The recorded audio has to be delayed
somewhat from the playback audio or echo cancellation will completely
fail. But hopefully the echo framework has removed a lot of the recorded
delay due to buffering.
Regards,
Bill
On 8/31/2016 8:11 AM, Severin Birrer wrote:
Hi Riza,
With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
Do I miss something?
Kind Regards,
Severin
Config:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=25;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output:
03:38:48.615 pjsua_call.c ..Answering call 0: code=200
03:38:48.616 pjsua_media.c .....Call 0: updating media..
03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
03:38:48.617 pjsua_aud.c ......Audio channel update..
03:38:48.619 strm0x74e1452c .......Encoder stream started
03:38:48.620 strm0x74e1452c .......Decoder stream started
03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
----- Ursprüngliche Mail -----
Von: "Riza Sulistyo" riza@pjsip.org
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Dienstag, 30. August 2016 11:08:45
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
We have added webrtc aec on #1954 .
Maybe you can try that.
For tweaking purposes, the information on #1888 is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
http://www.fela.ch/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Bill,
With the mobile AEC we're getting much better result's. It is now just a little noisy in the start of a call for a few seconds, but there is no echo :)
Thank you all so far^^
With "passing 0 as the latency estimate (in place of echo tail)" do you mean, i have to set the ecTailLen=0?
But then the AEC get's disabled with pjsua2.
Or do you mean set sndPlayLatency/sndRecLatency to 0.
Or how do I set the latency estimate?
Best Regards
Severin
Best Options so far:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=1;
ep_cfg.medConfig.sndRecLatency=0;
ep_cfg.medConfig.sndPlayLatency=0;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
Log:
07:00:40.169 echo_webrtc.c ........WebRTC AEC mobile successfully created with options 899
07:00:40.170 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=1 ms, latency=25 ms
----- Ursprüngliche Mail -----
Von: "Bill Gardner" <billg@wavearts.com>
An: "pjsip" <pjsip@lists.pjsip.org>
Gesendet: Mittwoch, 31. August 2016 15:46:43
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
It's very misleading that PSJIP passes the echo tail length to WebRTC.
WebRTC wants an estimate of the latency, not the echo tail length. If
you pass in too high a value, the echo cancellation will not work. I
think PJSIP implemented it this way because one can use the echo tail
parameter to pass a command line value all the way down to the AEC.
The mobile AEC does not have the automatic latency estimator, but it can
handle a much larger variation in latency and therefore does not need
an accurate estimate. I'd recommend using the mobile AEC and passing 0
as the latency estimate (in place of echo tail).
If not using mobile AEC, I'd still recommend using 0 as the latency
estimate. Note that the PJSIP echo framework automatically compensates
for audio buffer latency and passes signals to the echo canceller that
are more or less properly aligned. The recorded audio has to be delayed
somewhat from the playback audio or echo cancellation will completely
fail. But hopefully the echo framework has removed a lot of the recorded
delay due to buffering.
Regards,
Bill
On 8/31/2016 8:11 AM, Severin Birrer wrote:
> Hi Riza,
> With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
>
> The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
> But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
> Do I miss something?
>
> Kind Regards,
> Severin
>
> Config:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=25;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
> ep.libInit( ep_cfg );
>
> Log output:
>
> 03:38:48.615 pjsua_call.c ..Answering call 0: code=200
> 03:38:48.616 pjsua_media.c .....Call 0: updating media..
> 03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
> 03:38:48.617 pjsua_aud.c ......Audio channel update..
> 03:38:48.619 strm0x74e1452c .......Encoder stream started
> 03:38:48.620 strm0x74e1452c .......Decoder stream started
> 03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
> 03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
> 03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
> 03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
> 03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
> 03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
> 03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
> 03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
> 03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
> *****
> 03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
> 03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
> 03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
> 03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
> 03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
> 03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
> 03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
> 03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
> 03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>
>
>
>
> ----- Ursprüngliche Mail -----
> Von: "Riza Sulistyo" <riza@pjsip.org>
> An: "pjsip" <pjsip@lists.pjsip.org>
> Gesendet: Dienstag, 30. August 2016 11:08:45
> Betreff: Re: [pjsip] AEC echo cancelling
>
> Hi Severin,
> We have added webrtc aec on #1954 .
> Maybe you can try that.
> For tweaking purposes, the information on #1888 is still relevant.
>
> Best Regards,
>
> Riza
>
> On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
>
>
> I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
> I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
> The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
> Do I have to set some more parameters than the ecTailLen and ecOptions?
>
> The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
> Steps done so far with WebRTC:
> 1. Get the depot_tools
> 2. Get/Build webrtc:
> mkdir webrtc-checkout
> cd webrtc-checkout
> fetch --nohooks webrtc
> gclient sync
> gn gen out/Release
> ninja -C out/Release
> 3. Configure pjsip:
> ./configure --with-webrtc=~/webrtc-checkout
>
> Log output of configure:
> Using webrtc prefix... /home/yocto/webrtc-checkout
> checking for WebRtcAec_Process in -laudio_processing... no
>
>
> I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
> Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
>
>
> My setup:
> Buildsystem Ubuntu
> Target Yocto Linux on a i.MX6 (ARM)
> Loudspeaker & MIC
> SGTL5000 Audiocodec
> Latency: about 200ms (measured with pjsystest Latency Test)
>
> My initialisation routine of EC is:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=300;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
> ep.libInit( ep_cfg );
>
> Log output of SPEEX AEC:
> 10:37:54.669 pjsua_call.c ..Answering call 0: code=200
> 10:37:54.670 pjsua_media.c .....Call 0: updating media..
> 10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
> 10:37:54.672 pjsua_aud.c ......Audio channel update..
> 10:37:54.674 strm0x74d1452c .......Encoder stream started
> 10:37:54.674 strm0x74d1452c .......Decoder stream started
> 10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
> 10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
> 10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
> 10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
> 10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
> 10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
> 10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
> 10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
>
> -----------------------------------------------------------------------------------
> Severin Birrer
>
> FELA Management AG Phone : +41 52 646 46 37
> Basadingerstrasse 18 Fax : +41 52 646 46 96
> CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
> http://www.fela.ch/
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
>
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
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http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
BG
Bill Gardner
Wed, Aug 31, 2016 4:04 PM
I suggest editing echo_webrtc.c and remove passing echo tail to webrtc
process function and instead pass 0. I don't have code in front of me.
Alternatively you could use 1 which would be a negligible latency
estimate of 1 msec, and PJSIP will still enable AEC.
Don't change sndPlayLatency/andRecLatency, those define the audio buffer
sizes.
Regards,
Bill
On 8/31/2016 11:23 AM, Severin Birrer wrote:
Hi Bill,
With the mobile AEC we're getting much better result's. It is now just a little noisy in the start of a call for a few seconds, but there is no echo :)
Thank you all so far^^
With "passing 0 as the latency estimate (in place of echo tail)" do you mean, i have to set the ecTailLen=0?
But then the AEC get's disabled with pjsua2.
Or do you mean set sndPlayLatency/sndRecLatency to 0.
Or how do I set the latency estimate?
Best Regards
Severin
Best Options so far:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=1;
ep_cfg.medConfig.sndRecLatency=0;
ep_cfg.medConfig.sndPlayLatency=0;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
Log:
07:00:40.169 echo_webrtc.c ........WebRTC AEC mobile successfully created with options 899
07:00:40.170 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=1 ms, latency=25 ms
----- Ursprüngliche Mail -----
Von: "Bill Gardner" billg@wavearts.com
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Mittwoch, 31. August 2016 15:46:43
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
It's very misleading that PSJIP passes the echo tail length to WebRTC.
WebRTC wants an estimate of the latency, not the echo tail length. If
you pass in too high a value, the echo cancellation will not work. I
think PJSIP implemented it this way because one can use the echo tail
parameter to pass a command line value all the way down to the AEC.
The mobile AEC does not have the automatic latency estimator, but it can
handle a much larger variation in latency and therefore does not need
an accurate estimate. I'd recommend using the mobile AEC and passing 0
as the latency estimate (in place of echo tail).
If not using mobile AEC, I'd still recommend using 0 as the latency
estimate. Note that the PJSIP echo framework automatically compensates
for audio buffer latency and passes signals to the echo canceller that
are more or less properly aligned. The recorded audio has to be delayed
somewhat from the playback audio or echo cancellation will completely
fail. But hopefully the echo framework has removed a lot of the recorded
delay due to buffering.
Regards,
Bill
On 8/31/2016 8:11 AM, Severin Birrer wrote:
Hi Riza,
With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
Do I miss something?
Kind Regards,
Severin
Config:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=25;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output:
03:38:48.615 pjsua_call.c ..Answering call 0: code=200
03:38:48.616 pjsua_media.c .....Call 0: updating media..
03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
03:38:48.617 pjsua_aud.c ......Audio channel update..
03:38:48.619 strm0x74e1452c .......Encoder stream started
03:38:48.620 strm0x74e1452c .......Decoder stream started
03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
----- Ursprüngliche Mail -----
Von: "Riza Sulistyo" riza@pjsip.org
An: "pjsip" pjsip@lists.pjsip.org
Gesendet: Dienstag, 30. August 2016 11:08:45
Betreff: Re: [pjsip] AEC echo cancelling
Hi Severin,
We have added webrtc aec on #1954 .
Maybe you can try that.
For tweaking purposes, the information on #1888 is still relevant.
Best Regards,
Riza
On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
Do I have to set some more parameters than the ecTailLen and ecOptions?
The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
Steps done so far with WebRTC:
- Get the depot_tools
- Get/Build webrtc:
mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
gn gen out/Release
ninja -C out/Release
- Configure pjsip:
./configure --with-webrtc=~/webrtc-checkout
Log output of configure:
Using webrtc prefix... /home/yocto/webrtc-checkout
checking for WebRtcAec_Process in -laudio_processing... no
I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
My setup:
Buildsystem Ubuntu
Target Yocto Linux on a i.MX6 (ARM)
Loudspeaker & MIC
SGTL5000 Audiocodec
Latency: about 200ms (measured with pjsystest Latency Test)
My initialisation routine of EC is:
EpConfig ep_cfg;
ep_cfg.medConfig.noVad=true;
ep_cfg.medConfig.ecTailLen=300;
ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
ep.libInit( ep_cfg );
Log output of SPEEX AEC:
10:37:54.669 pjsua_call.c ..Answering call 0: code=200
10:37:54.670 pjsua_media.c .....Call 0: updating media..
10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
10:37:54.672 pjsua_aud.c ......Audio channel update..
10:37:54.674 strm0x74d1452c .......Encoder stream started
10:37:54.674 strm0x74d1452c .......Decoder stream started
10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
!*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
!*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
Severin Birrer
FELA Management AG Phone : +41 52 646 46 37
Basadingerstrasse 18 Fax : +41 52 646 46 96
CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
http://www.fela.ch/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
I suggest editing echo_webrtc.c and remove passing echo tail to webrtc
process function and instead pass 0. I don't have code in front of me.
Alternatively you could use 1 which would be a negligible latency
estimate of 1 msec, and PJSIP will still enable AEC.
Don't change sndPlayLatency/andRecLatency, those define the audio buffer
sizes.
Regards,
Bill
On 8/31/2016 11:23 AM, Severin Birrer wrote:
> Hi Bill,
>
> With the mobile AEC we're getting much better result's. It is now just a little noisy in the start of a call for a few seconds, but there is no echo :)
> Thank you all so far^^
>
> With "passing 0 as the latency estimate (in place of echo tail)" do you mean, i have to set the ecTailLen=0?
> But then the AEC get's disabled with pjsua2.
>
> Or do you mean set sndPlayLatency/sndRecLatency to 0.
>
> Or how do I set the latency estimate?
>
> Best Regards
> Severin
>
> Best Options so far:
> EpConfig ep_cfg;
> ep_cfg.medConfig.noVad=true;
> ep_cfg.medConfig.ecTailLen=1;
> ep_cfg.medConfig.sndRecLatency=0;
> ep_cfg.medConfig.sndPlayLatency=0;
> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
>
> Log:
>
> 07:00:40.169 echo_webrtc.c ........WebRTC AEC mobile successfully created with options 899
> 07:00:40.170 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=1 ms, latency=25 ms
>
>
> ----- Ursprüngliche Mail -----
> Von: "Bill Gardner" <billg@wavearts.com>
> An: "pjsip" <pjsip@lists.pjsip.org>
> Gesendet: Mittwoch, 31. August 2016 15:46:43
> Betreff: Re: [pjsip] AEC echo cancelling
>
> Hi Severin,
>
> It's very misleading that PSJIP passes the echo tail length to WebRTC.
> WebRTC wants an estimate of the latency, not the echo tail length. If
> you pass in too high a value, the echo cancellation will not work. I
> think PJSIP implemented it this way because one can use the echo tail
> parameter to pass a command line value all the way down to the AEC.
>
> The mobile AEC does not have the automatic latency estimator, but it can
> handle a much larger variation in latency and therefore does not need
> an accurate estimate. I'd recommend using the mobile AEC and passing 0
> as the latency estimate (in place of echo tail).
>
> If not using mobile AEC, I'd still recommend using 0 as the latency
> estimate. Note that the PJSIP echo framework automatically compensates
> for audio buffer latency and passes signals to the echo canceller that
> are more or less properly aligned. The recorded audio has to be delayed
> somewhat from the playback audio or echo cancellation will completely
> fail. But hopefully the echo framework has removed a lot of the recorded
> delay due to buffering.
>
> Regards,
>
> Bill
>
>
>
> On 8/31/2016 8:11 AM, Severin Birrer wrote:
>> Hi Riza,
>> With the Information in #1954 I could successfully build webrtc aec (SVN trunk rev 5435).
>>
>> The results were a lot better as with SPEEX aec. But only if I wait for about 20 seconds. (Because of wrong tail-length I think)
>> But I still have some trouble to find the correct tail-length value. I enabled the printing of the log line with the median (set SHOW_DELAY_METRICS 250 and disabled PJMEDIA_WEBRTC_AEC_USE_MOBILE). But still get the same results (about 20 seconds of waiting with echo) if I adjust the EC tail value to the median value of 4 (strange value?) i got from log.
>> Do I miss something?
>>
>> Kind Regards,
>> Severin
>>
>> Config:
>> EpConfig ep_cfg;
>> ep_cfg.medConfig.noVad=true;
>> ep_cfg.medConfig.ecTailLen=25;
>> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_WEBRTC|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
>> ep.libInit( ep_cfg );
>>
>> Log output:
>>
>> 03:38:48.615 pjsua_call.c ..Answering call 0: code=200
>> 03:38:48.616 pjsua_media.c .....Call 0: updating media..
>> 03:38:48.617 pjsua_media.c ......Audio updated, stream #0: (inactive)
>> 03:38:48.617 pjsua_aud.c ......Audio channel update..
>> 03:38:48.619 strm0x74e1452c .......Encoder stream started
>> 03:38:48.620 strm0x74e1452c .......Decoder stream started
>> 03:38:48.620 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
>> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
>> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
>> 03:38:48.623 pjsua_aud.c .....Conf connect: 2 --> 0
>> 03:38:48.623 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
>> 03:38:48.624 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
>> 03:38:48.656 echo_webrtc.c ........WebRTC AEC successfully created with options 899
>> 03:38:48.657 ec0x74e1bf50 ........WebRTC AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=25 ms, latency=100 ms
>> 03:38:48.659 conference.c ......Port 2 (sip:202@192.168.139.152;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
>> 03:38:48.660 pjsua_aud.c .....Conf connect: 0 --> 2
>> 03:38:48.661 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 (sip:202@192.168.139.152;user=phone)
>> *****
>> 03:38:48.687 Master/sound !Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.690 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.696 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.697 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.701 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.705 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.708 Master/sound Underflow, buf_cnt=0, will generate 1 frame
>> 03:38:48.924 stream.c !G722 codec used, remote samples per frame detected = 80
>> 03:38:53.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=48, fraction of poor delays=1.000000
>> 03:38:58.587 ec0x74e1bf50 !Buffer size adjusted from 960 to 513 (eff_cnt=635)
>> 03:38:58.695 echo_webrtc.c !WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
>> 03:39:03.695 echo_webrtc.c WebRTC delay metrics: --> median=-28, std=0, fraction of poor delays=1.000000
>> 03:39:08.695 echo_webrtc.c WebRTC delay metrics: --> median=-24, std=4, fraction of poor delays=0.919200
>> 03:39:13.695 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>> 03:39:18.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>> 03:39:23.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>> 03:39:28.694 echo_webrtc.c WebRTC delay metrics: --> median=4, std=0, fraction of poor delays=0.000000
>>
>>
>>
>>
>> ----- Ursprüngliche Mail -----
>> Von: "Riza Sulistyo" <riza@pjsip.org>
>> An: "pjsip" <pjsip@lists.pjsip.org>
>> Gesendet: Dienstag, 30. August 2016 11:08:45
>> Betreff: Re: [pjsip] AEC echo cancelling
>>
>> Hi Severin,
>> We have added webrtc aec on #1954 .
>> Maybe you can try that.
>> For tweaking purposes, the information on #1888 is still relevant.
>>
>> Best Regards,
>>
>> Riza
>>
>> On Tue, Aug 30, 2016 at 3:02 PM, Severin Birrer < severin.birrer@fela.ch > wrote:
>>
>>
>> I'm trying to get some better Results with the AEC. (pjproject Version 2.5.5)
>> I tried SPEEX AEC but the echo cancelling doesn't work as hoped for.
>> The cancelling works, but you still get a little echo and after some time (one minute or so) the background noise gets louder and I think the cancelling gets also worse.
>> Do I have to set some more parameters than the ecTailLen and ecOptions?
>>
>> The other Thing i tried is to integrate WebRTC AEC (Mostly described in Ticket #1888). But no luck there to compile it in, because of a missing full-howto.
>> Steps done so far with WebRTC:
>> 1. Get the depot_tools
>> 2. Get/Build webrtc:
>> mkdir webrtc-checkout
>> cd webrtc-checkout
>> fetch --nohooks webrtc
>> gclient sync
>> gn gen out/Release
>> ninja -C out/Release
>> 3. Configure pjsip:
>> ./configure --with-webrtc=~/webrtc-checkout
>>
>> Log output of configure:
>> Using webrtc prefix... /home/yocto/webrtc-checkout
>> checking for WebRtcAec_Process in -laudio_processing... no
>>
>>
>> I think the webrtc aec libraries (audio_processing, common_audio, system_wrappers) doesn't get built.
>> Is it due to the compatibility issue described in the ticket? Which Version do I have to get, or more, how do I get it?
>>
>>
>> My setup:
>> Buildsystem Ubuntu
>> Target Yocto Linux on a i.MX6 (ARM)
>> Loudspeaker & MIC
>> SGTL5000 Audiocodec
>> Latency: about 200ms (measured with pjsystest Latency Test)
>>
>> My initialisation routine of EC is:
>> EpConfig ep_cfg;
>> ep_cfg.medConfig.noVad=true;
>> ep_cfg.medConfig.ecTailLen=300;
>> ep_cfg.medConfig.ecOptions=PJMEDIA_ECHO_SPEEX|PJMEDIA_ECHO_USE_NOISE_SUPPRESSOR|PJMEDIA_ECHO_USE_SW_ECHO|PJMEDIA_ECHO_AGGRESSIVENESS_AGGRESSIVE;
>> ep.libInit( ep_cfg );
>>
>> Log output of SPEEX AEC:
>> 10:37:54.669 pjsua_call.c ..Answering call 0: code=200
>> 10:37:54.670 pjsua_media.c .....Call 0: updating media..
>> 10:37:54.671 pjsua_media.c ......Audio updated, stream #0: (inactive)
>> 10:37:54.672 pjsua_aud.c ......Audio channel update..
>> 10:37:54.674 strm0x74d1452c .......Encoder stream started
>> 10:37:54.674 strm0x74d1452c .......Decoder stream started
>> 10:37:54.675 pjsua_media.c ......Audio updated, stream #1: G722 (sendrecv)
>> !*** Recording device is: sysdefault:CARD=imxsgtl5000 With ID: -1
>> !*** Playback device is: sysdefault:CARD=imxsgtl5000 With ID: -2
>> 10:37:54.678 pjsua_aud.c .....Conf connect: 2 --> 0
>> 10:37:54.678 pjsua_aud.c ......Set sound device: capture=-1, playback=-2
>> 10:37:54.678 pjsua_aud.c .......Opening sound device (speaker + mic) PCM@16000/1/20ms
>> 10:37:54.713 ec0x74d1bf50 ........AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=300 ms, latency=0 ms
>> 10:37:54.715 conference.c ......Port 2 ( sip:203@192.168.139.152 ;user=phone) transmitting to port 0 (sysdefault:CARD=imxsgtl5000)
>> 10:37:54.716 pjsua_aud.c .....Conf connect: 0 --> 2
>> 10:37:54.717 conference.c ......Port 0 (sysdefault:CARD=imxsgtl5000) transmitting to port 2 ( sip:203@192.168.139.152 ;user=phone)
>>
>> -----------------------------------------------------------------------------------
>> Severin Birrer
>>
>> FELA Management AG Phone : +41 52 646 46 37
>> Basadingerstrasse 18 Fax : +41 52 646 46 96
>> CH-8253 Diessenhofen Mail : mailto: severin.birrer@fela.ch
>> http://www.fela.ch/
>>
>> _______________________________________________
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>>
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>>
>>
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>>
>>
>> _______________________________________________
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>>
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>>
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>
> _______________________________________________
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>
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>
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