static void volume_process(MSFilter *f){
mblk_t *m;
Volume *v=(Volume*)f->data;
float target_gain;
/* Important notice: any processes called herein can modify v->target_gain, at
* end of this function apply_gain() is called, thus: later process calls can
* override this target gain, and order must be well thought out
*/
if (v->agc_enabled || v->peer!=NULL){
mblk_t *om;
int nbytes=v->nsamples*2;
ms_bufferizer_put_from_queue(v->buffer,f->inputs[0]);
while(ms_bufferizer_get_avail(v->buffer)>=nbytes){
om=allocb(nbytes,0);
ms_bufferizer_read(v->buffer,om->b_wptr,nbytes);
om->b_wptr+=nbytes;
update_energy((int16_t*)om->b_rptr, v->nsamples, v);
target_gain = v->static_gain;
if (v->peer) /* this ptr set = echo limiter enable flag */
target_gain = volume_echo_avoider_process(v, om);
/* Multiply with gain from echo limiter, not "choose smallest". Why?
* Remote talks, local echo suppress via mic path, but still audible in
* remote speaker. AGC operates fully, too (local speaker close to local mic!);
* having agc gain reduction also contribute to total reduction makes sense.
*/
if (v->agc_enabled) target_gain/= volume_agc_process(v, om);
if (v->noise_gate_enabled)
volume_noise_gate_process(v, v->level_pk, om);
apply_gain(v, om, target_gain);
ms_queue_put(f->outputs[0],om);
}
}else{
/*light processing: no agc. Work in place in the input buffer*/
while((m=ms_queue_get(f->inputs[0]))!=NULL){
update_energy((int16_t*)m->b_rptr, (m->b_wptr - m->b_rptr) / 2, v);
target_gain = v->static_gain;
if (v->noise_gate_enabled)
volume_noise_gate_process(v, v->level_pk, m);
apply_gain(v, m, target_gain);
ms_queue_put(f->outputs[0],m);
}
}
}
void iir_coeff::convert_to_ab() {
float_type hpf_z_gain = 0;
float_type hpf_p_gain = 0;
float_type z_gain = 0;
gain = (float_type)1.0;
z_root_to_ab(zeros);
z_gain = gain;
hpf_z_gain = hpf_gain;
gain = 1.0;
hpf_gain = 1.0;
z_root_to_ab(poles);
hpf_p_gain = hpf_gain;
gain /= z_gain;
if (odd) gain *= 0.5 * (1.0 - real(poles[0]));
hpf_gain = hpf_p_gain / hpf_z_gain;
if (lpf == filter_type::high) gain = hpf_gain;
state = filter_state::s3; // in Z-domain 2nd order A/B coefficients
a_tf = p2_to_poly(poles);
b_tf = p2_to_poly(zeros);
// Apply gain to b coefficents
apply_gain(gain);
}
static sf_count_t
sample_rate_convert (SNDFILE *infile, SNDFILE *outfile, int converter, double src_ratio, int channels, double * gain, int normalize)
{ static float input [BUFFER_LEN] ;
static float output [BUFFER_LEN] ;
SRC_STATE *src_state ;
SRC_DATA src_data ;
int error ;
double max = 0.0 ;
sf_count_t output_count = 0 ;
sf_seek (infile, 0, SEEK_SET) ;
sf_seek (outfile, 0, SEEK_SET) ;
/* Initialize the sample rate converter. */
if ((src_state = src_new (converter, channels, &error)) == NULL)
{ printf ("\n\nError : src_new() failed : %s.\n\n", src_strerror (error)) ;
exit (1) ;
} ;
src_data.end_of_input = 0 ; /* Set this later. */
/* Start with zero to force load in while loop. */
src_data.input_frames = 0 ;
src_data.data_in = input ;
src_data.src_ratio = src_ratio ;
src_data.data_out = output ;
src_data.output_frames = BUFFER_LEN /channels ;
while (1)
{
/* If the input buffer is empty, refill it. */
if (src_data.input_frames == 0)
{ src_data.input_frames = sf_readf_float (infile, input, BUFFER_LEN / channels) ;
src_data.data_in = input ;
/* The last read will not be a full buffer, so snd_of_input. */
if (src_data.input_frames < BUFFER_LEN / channels)
src_data.end_of_input = SF_TRUE ;
} ;
if ((error = src_process (src_state, &src_data)))
{ printf ("\nError : %s\n", src_strerror (error)) ;
exit (1) ;
} ;
/* Terminate if done. */
if (src_data.end_of_input && src_data.output_frames_gen == 0)
break ;
max = apply_gain (src_data.data_out, src_data.output_frames_gen, channels, max, *gain) ;
/* Write output. */
sf_writef_float (outfile, output, src_data.output_frames_gen) ;
output_count += src_data.output_frames_gen ;
src_data.data_in += src_data.input_frames_used * channels ;
src_data.input_frames -= src_data.input_frames_used ;
} ;
src_delete (src_state) ;
if (normalize && max > 1.0)
{ *gain = 1.0 / max ;
printf ("\nOutput has clipped. Restarting conversion to prevent clipping.\n\n") ;
return -1 ;
} ;
return output_count ;
} /* sample_rate_convert */
void iir_coeff::set_bandpass_gain() {
double w = 2.0*M_PI*get_center();
float_type gain = freqz_mag(w);
apply_gain(1.0 / gain);
}
