JNIEXPORT jint JNICALL Java_com_jabber_audio_encoder_AudioWebrtcAecm_yy_1webrtc_1aecm_1initiate
(JNIEnv *env, jobject obj, jint samp_freq){
__android_log_print(ANDROID_LOG_ERROR ,TAG ,"in audioWebrtcAecm initiate function ");
int ret ;
void* AECM_instance;
if ((ret = WebRtcAecm_Create(&AECM_instance) )) {
__android_log_print(ANDROID_LOG_ERROR ,TAG ,"WebRtcAecm_Create failed with error code = %d", ret);
return ret;
}
if ((ret = WebRtcAecm_Init(AECM_instance, samp_freq) )) {
__android_log_print(ANDROID_LOG_ERROR ,TAG ,"WebRtcAecm_Init failed with error code = %d", ret);
return ret;
}
#if 0 //config
AecmConfig aecm_config;
aecm_config.cngMode = AECM_TRUE;
aecm_config.echoMode = 3;
if(( ret = WebRtcAecm_set_config(AECM_instance, aecm_config) )){
__android_log_print(ANDROID_LOG_ERROR ,TAG ,"WebRtcAecm_set_config failed with error code = %d", ret);
return ret;
}
#endif
return (jint)((int *)AECM_instance );
}
static void webrtc_aec_preprocess(MSFilter *f)
{
WebRTCAECState *s = (WebRTCAECState *) f->data;
AecmConfig config;
int delay_samples = 0;
mblk_t *m;
int error_code;
s->echostarted = FALSE;
delay_samples = s->delay_ms * s->samplerate / 1000;
s->framesize=(framesize*s->samplerate)/8000;
ms_message("Initializing WebRTC echo canceler with framesize=%i, delay_ms=%i, delay_samples=%i", s->framesize, s->delay_ms, delay_samples);
if ((s->aecmInst = WebRtcAecm_Create()) == NULL) {
s->bypass_mode = TRUE;
ms_error("WebRtcAecm_Create(): error, entering bypass mode");
return;
}
if ((error_code = WebRtcAecm_Init(s->aecmInst, s->samplerate)) < 0) {
if (error_code == AECM_BAD_PARAMETER_ERROR) {
ms_error("WebRtcAecm_Init(): WebRTC echo canceller does not support %d samplerate", s->samplerate);
}
s->bypass_mode = TRUE;
ms_error("Entering bypass mode");
return;
}
config.cngMode = TRUE;
config.echoMode = 3;
if (WebRtcAecm_set_config(s->aecmInst, config)!=0){
ms_error("WebRtcAecm_set_config(): failed.");
}
/* fill with zeroes for the time of the delay*/
m = allocb(delay_samples * 2, 0);
m->b_wptr += delay_samples * 2;
ms_bufferizer_put(&s->delayed_ref, m);
s->min_ref_samples = -1;
s->nominal_ref_samples = delay_samples;
ms_audio_flow_controller_init(&s->afc);
s->flow_control_time = f->ticker->time;
}