I jus followed the ticket, that I mentioned.
And disabled speex and speex-aec (actually I disabled all audio codecs except opus =) at compile time (with ./configure).
Here is my ./configure command:
TARGET_ABI=armeabi-v7a APP_PLATFORM=android-16 ./configure-android --use-ndk-cflags --disable-speex-aec --disable-g711-codec --disable-l16-codec --disable-gsm-codec --disable-g722-codec --disable-g7221-codec --disable-speex-codec --disable-ilbc-codec --disable-ffmpeg --disable-opencore-amr --disable-silk --with-openh264=/projects/openh264-1.0.0/android --with-libyuv=/projects/libyuv-android/jni --with-opus=/projects/opus/build --disable-webrtc
Feel free to adopt it for your case.
Also I have nothing special in config_site.h:
#define PJ_CONFIG_ANDROID 1
#define PJMEDIA_HAS_VIDEO 1
#include <pj/config_site_sample.h>
What OS system are you using with pjsip？
For Android， the speex AEC that came with pjmedia_snd_port seem's have problem to work properly even with the latest pjproject2.4.5.
I have to turned off the default AEC, and manually do it within the callback functions. Such as
> snd_port_param.base.ec_enabled=PJ_FALSE;
>snd_port_param._on_rec_frame_=rec_cb; >snd_port_param._on_play_frame_=play_cb;
> status=pjmedia_echo_create2(....);You have to be careful with all the parameters, because the default setting would not work for SPEEX.I would really hope if there were some easier ways for Android developers.Сергей Митрофанов <goretz.m@xxxxxxxxx>于2016年4月9日周六 上午1:50写道：_______________________________________________Hi, I had same issue 2 weeks ago.
I just use the new webrtc echo canceller from the last commits. And it works greate!
Also I disabled al other EC and unused audio codecs...
Check this ticket for help:
https://trac.pjsip.org/repos/ticket/1888пт, 8 апр. 2016 г. в 20:36, João Resende <ms92resende@xxxxxxxxx>:______________________________________________________________________________________________I need to activate the echo cancelation algorithm and I try to add #define PJMEDIA_HAS_SPEEX_AEC 1 in the config_site.h but then I realise that this is the default behaviour of PJSIP. But I'm listening a lot of echo during a call.
Can someone tell me what variable should I set to solve this issue?
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