Yes it does. The device is handsfree.
Thanks.
Alexandre Gonçalves
............................................................................................................................
Email: alexandre@xxxxxxxxxxxxxxxxxx
2016-04-11 16:09 GMT+01:00 Bill Gardner <billg@xxxxxxxxxxxx>:
Hi Alexandre,
I don't know what the problem is, but you shouldn't need to create a separate EC object. pjsua_init will create an echo canceller and connect it to the sound port.
Does the call work without EC?
Regards,
Bill
On 4/11/2016 10:33 AM, Alexandre Gonçalves wrote:
Hello,
I've made a small C application to run on a linux machine, based on "Simple PJSUA" example.
What I need to add is echo cancellation, and I tryed to do it like the code attached:
But according to the logs I get it seems that something is missing. I suspect it's the manipulation of the capture and playback frames, but I have no idea how to do it!
#include <pjsua-lib/pjsua.h>#define THIS_FILE "APP"
#define SIP_DOMAIN "raspbx"#define SIP_USER "511"#define SIP_PASSWD "aaa511"#define SIP_REALM "asterisk"#define SIP_SCHEME "digest"
pjmedia_echo_state *ec;pjmedia_frame play_frame, rec_frame;
/* Callback called by the library upon receiving incoming call */static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,pjsip_rx_data *rdata) {pjsua_call_info ci;
PJ_UNUSED_ARG(acc_id);PJ_UNUSED_ARG(rdata);
pjsua_call_get_info(call_id, &ci);
PJ_LOG(2, (THIS_FILE, "Incoming call from %.*s!!",(int) ci.remote_info.slen,ci.remote_info.ptr));
/* Automatically answer incoming calls with 200/OK */pjsua_call_answer(call_id, 200, NULL, NULL);}
/* Callback called by the library when call's state has changed */static void on_call_state(pjsua_call_id call_id, pjsip_event *e) {pjsua_call_info ci;
PJ_UNUSED_ARG(e);
pjsua_call_get_info(call_id, &ci);PJ_LOG(2, (THIS_FILE, "Call %d state=%.*s", call_id,(int) ci.state_text.slen,ci.state_text.ptr));
}
/* Callback called by the library when call's media state has changed */static void on_call_media_state(pjsua_call_id call_id) {pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {// When media is active, connect call to sound device.pjsua_conf_connect(ci.conf_slot, 0);pjsua_conf_connect(0, ci.conf_slot);}}
/* Display error and exit application */static void error_exit(const char *title, pj_status_t status) {pjsua_perror(THIS_FILE, title, status);pjsua_destroy();exit(1);}
/** main()** argv[1] may contain URL to call.*/int main(int argc, char *argv[]) {pjsua_acc_id acc_id;pj_status_t status;
/* Create pjsua first! */status = pjsua_create();if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
/* Init pjsua */{pjsua_config cfg;pjsua_logging_config log_cfg;pjsua_media_config media_cfg;
pjsua_config_default(&cfg);cfg.cb._on_incoming_call_ = &on_incoming_call;cfg.cb._on_call_media_state_ = &on_call_media_state;cfg.cb._on_call_state_ = &on_call_state;cfg.max_calls = 1;
pjsua_logging_config_default(&log_cfg);log_cfg.console_level = 6;
pjsua_media_config_default(&media_cfg);media_cfg.ec_options = PJMEDIA_ECHO_DEFAULT;media_cfg.ec_tail_len = 250;
status = pjsua_init(&cfg, &log_cfg, &media_cfg);// status = pjsua_init(&cfg, &log_cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
pj_pool_t *pool = pjsua_pool_create("my_echo", 1000, 1000);status = pjmedia_echo_create(pool, 16000, 320, 500, 500, PJMEDIA_ECHO_DEFAULT, &ec);play_frame.buf = pj_pool_alloc(pool, 320);rec_frame.buf = pj_pool_alloc(pool, 320);
}
/* Add UDP transport. */{pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);cfg.port = 5060;status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);if (status != PJ_SUCCESS) error_exit("Error creating transport", status);}
/* Initialization is done, now start pjsua */status = pjsua_start();if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
/* Register to SIP server by creating SIP account. */{pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);cfg.cred_count = 1;cfg.cred_info[0].realm = pj_str(SIP_REALM);cfg.cred_info[0].scheme = pj_str(SIP_SCHEME);cfg.cred_info[0].username = pj_str(SIP_USER);cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;cfg.cred_info[0].data = "">
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);if (status != PJ_SUCCESS) error_exit("Error adding account", status);}
if (argc > 1) {pj_str_t uri = pj_str(argv[1]);status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);if (status != PJ_SUCCESS) error_exit("Error making call", status);}
char option[10];while (1) {
if (fgets(option, sizeof (option), stdin) == NULL) {puts("EOF while reading stdin, will quit now..");break;}
if (option[0] == 'q')break;
}
/* Destroy pjsua */pjsua_destroy();
return 0;}
The logs:
16:10:45.203 speex !warning: No playback frame available (your application is buggy and/or got xruns)16:10:45.207 speex !warning: Auto-filling the buffer (your application is buggy and/or got xruns)16:10:45.216 speex !warning: internal playback buffer corruption?16:10:45.221 speex !warning: Auto-filling the buffer (your application is buggy and/or got xruns)16:10:45.279 speex warning: Had to discard a playback frame (your application is buggy and/or got xruns)16:10:45.393 speex warning: Auto-filling the buffer (your application is buggy and/or got xruns)16:10:45.462 speex warning: Had to discard a playback frame (your application is buggy and/or got xruns)16:10:45.522 speex warning: Auto-filling the buffer (your application is buggy and/or got xruns)16:10:45.562 speex warning: Had to discard a playback frame (your application is buggy and/or got xruns)
Pjsip version: 2.4
Can anyone help out?
Thanks.
Alexandre Gonçalves
............................................................................................................................
Email: alexandre@xxxxxxxxxxxxxxxxxx
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