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sems/core/AmPlayoutBuffer.cpp

410 lines
8.5 KiB

#include "AmPlayoutBuffer.h"
#include "AmAudio.h"
#define SEARCH_OFFSET 140
#define SEARCH_REGION 110
#define DELTA 5
#define TSM_MAX_SCALE 2.0
#define TSM_MIN_SCALE 0.5
// only scale if 0.9 < f < 1.1
#define SCALE_FACTOR_START 0.1
#define PI 3.14
#define DEBUG_PLAYOUTBUF
AmPlayoutBuffer::AmPlayoutBuffer()
: r_ts(0),w_ts(0)
{
}
void AmPlayoutBuffer::direct_write(unsigned int ts, ShortSample* buf, unsigned int len)
{
#ifndef USE_ADAPTIVE_JB
buffer_put(w_ts,buf,len);
#else
buffer_put(ts, buf, len);
#endif // USE_ADAPTIVE_JB
}
void AmPlayoutBuffer::write(u_int32_t ref_ts, u_int32_t ts, int16_t* buf, u_int32_t len)
{
buffer_put(w_ts,buf,len);
}
u_int32_t AmPlayoutBuffer::read(u_int32_t ts, int16_t* buf, u_int32_t len)
{
#ifndef USE_ADAPTIVE_JB
if(ts_less()(r_ts,w_ts)){
u_int32_t rlen=0;
if(ts_less()(r_ts+PCM16_B2S(AUDIO_BUFFER_SIZE),w_ts))
rlen = PCM16_B2S(AUDIO_BUFFER_SIZE);
else
rlen = w_ts - r_ts;
buffer_get(r_ts,buf,rlen);
return rlen;
}
return 0;
#else
buffer_get(ts, buf, len);
return len;
#endif // USE_ADAPTIVE_JB
}
AmAdaptivePlayout::AmAdaptivePlayout()
: idx(0),
loss_rate(ORDER_STAT_LOSS_RATE),
wsola_off(WSOLA_START_OFF),
shr_threshold(SHR_THRESHOLD),
plc_cnt(0),
short_scaled(WSOLA_SCALED_WIN)
{
memset(n_stat,0,sizeof(int32_t)*ORDER_STAT_WIN_SIZE);
}
u_int32_t AmAdaptivePlayout::next_delay(u_int32_t ref_ts, u_int32_t ts)
{
int32_t n = (int32_t)(ref_ts - ts);
multiset<int32_t>::iterator it = o_stat.find(n_stat[idx]);
if(it != o_stat.end())
o_stat.erase(it);
n_stat[idx] = n;
o_stat.insert(n);
int32_t D_r=0,D_r1=0;
int r = int((double(o_stat.size()) + 1.0)*(1.0 - loss_rate));
if((r == 0) || (r >= (int)o_stat.size())){
StddevValue n_std;
for(int i=0; i<ORDER_STAT_WIN_SIZE; i++){
n_std.push(double(n_stat[i]));
}
if(r == 0){
D_r = (*o_stat.begin()) - (int32_t)(2.0*n_std.stddev());
D_r1 = (*o_stat.begin());
}
else {
D_r = (*o_stat.rbegin());
D_r1 = (*o_stat.rbegin()) + (int32_t)(2.0*n_std.stddev());
}
}
else {
int i=0;
for(it = o_stat.begin(); it != o_stat.end(); it++){
if(++i == r){
D_r = (*it);
++it;
D_r1 = (*it);
break;
}
}
}
int32_t D =
int32_t(D_r + double(D_r1 - D_r)
* ( (double(o_stat.size()) + 1.0)
*(1.0-loss_rate) - double(r)));
if(++idx >= ORDER_STAT_WIN_SIZE)
idx = 0;
return D;
}
void AmAdaptivePlayout::write(u_int32_t ref_ts, u_int32_t ts,
int16_t* buf, u_int32_t len)
{
// predict next delay
u_int32_t p_delay = next_delay(ref_ts,ts);
u_int32_t old_off = wsola_off;
ts += old_off;
if(short_scaled.mean() > 2.0){
if(shr_threshold < 3000)
shr_threshold += 10;
}
else if(short_scaled.mean() < 1.0){
if(shr_threshold > 100)
shr_threshold -= 2;
}
// need to scale?
if( ts_less()(wsola_off+EXP_THRESHOLD,p_delay) || // expand packet
ts_less()(p_delay+shr_threshold,wsola_off) ) { // shrink packet
wsola_off = p_delay;
}
else {
if(ts_less()(r_ts,ts+len)){
plc_cnt = 0;
buffer_put(ts,buf,len);
}
else {
// lost
}
// statistics
short_scaled.push(0.0);
return;
}
int32_t n_len = len + wsola_off - old_off;
if(n_len < 0)
n_len = 1;
float f = float(n_len) / float(len);
if(f > TSM_MAX_SCALE)
f = TSM_MAX_SCALE;
n_len = (int32_t)(float(len) * f);
if(ts_less()(ts+n_len,r_ts)){
// statistics
short_scaled.push(0.0);
return;
}
u_int32_t old_wts = w_ts;
buffer_put(ts,buf,len);
n_len = time_scale(ts,f,len);
wsola_off = old_off + n_len - len;
//ts += n_len - len;
if(w_ts != old_wts)
plc_cnt = 0;
// statistics
short_scaled.push(100.0);
}
u_int32_t AmAdaptivePlayout::read(u_int32_t ts, int16_t* buf, u_int32_t len)
{
bool do_plc=false;
if(ts_less()(w_ts,ts+len) && (plc_cnt < 6)){
if(!plc_cnt){
time_scale(w_ts-len,2.0, len);
}
else {
do_plc = true;
}
plc_cnt++;
}
if(do_plc){
short plc_buf[FRAMESZ];
for(unsigned int i=0; i<len/FRAMESZ; i++){
fec.dofe(plc_buf);
buffer_put(w_ts,plc_buf,FRAMESZ);
}
buffer_get(ts,buf,len);
}
else {
buffer_get(ts,buf,len);
for(unsigned int i=0; i<len/FRAMESZ; i++)
fec.addtohistory(buf + i*FRAMESZ);
}
return len;
}
void AmAdaptivePlayout::direct_write(unsigned int ts, ShortSample* buf, unsigned int len)
{
buffer_put(ts+wsola_off,buf,len);
}
void AmPlayoutBuffer::buffer_put(unsigned int ts, ShortSample* buf, unsigned int len)
{
buffer.put(ts,buf,len);
if(ts_less()(w_ts,ts+len))
w_ts = ts + len;
}
void AmPlayoutBuffer::buffer_get(unsigned int ts, ShortSample* buf, unsigned int len)
{
buffer.get(ts,buf,len);
if(ts_less()(r_ts,ts+len))
r_ts = ts + len;
}
/**
* find best cross correlation of a TEMPLATE_SEG samples
* long frame
* * starting between sr_beg ... sr_end
* * to TEMPLATE_SEG samples frame starting from ts
*
*/
short* find_best_corr(short *ts, short *sr_beg, short* sr_end)
{
// find best correlation
float corr=0.f,best_corr=0.f;
short *best_sr=ts;
short *sr;
for(sr = sr_beg; sr != sr_end; sr++){
corr=0.f;
for(int i=0; i<TEMPLATE_SEG; i++)
corr += float(sr[i]) * float(ts[i]);
if((best_sr == 0) || (corr > best_corr)){
best_corr = corr;
best_sr = sr;
}
}
return best_sr;
}
u_int32_t AmAdaptivePlayout::time_scale(u_int32_t ts, float factor,
u_int32_t packet_len)
{
// current position in strech buffer
short *tmpl = p_buf + packet_len;
// begin and end of strech buffer
short *p_buf_beg = p_buf;
short *p_buf_end;
// initially size is packet_len
unsigned int s = packet_len;
// we start from beginning of frame
unsigned int cur_ts = ts;
// safety
if (packet_len > MAX_PACKET_SAMPLES)
return s;
if (fabs(factor - 1.0) <= SCALE_FACTOR_START) {
#ifdef DEBUG_PLAYOUTBUF
DBG("not scaling - too little f difference \n");
#endif
return s;
}
// boundaries of scaling
if(factor > TSM_MAX_SCALE)
factor = TSM_MAX_SCALE;
else if(factor < TSM_MIN_SCALE)
factor = TSM_MIN_SCALE;
short *srch_beg, *srch_end, *srch;
while(true){
// get previous packet_len frame + scaled frame
// (with size s) into p_buf
buffer_get(ts - packet_len, p_buf_beg, s + packet_len);
p_buf_end = p_buf_beg + s + packet_len;
// determine search region for template seg
// as srch_beg ... srch_end
if (factor > 1.0){
// expansion
srch_beg = tmpl - (int)((float)TEMPLATE_SEG * (factor - 1.0)) - SEARCH_REGION/2;
srch_end = srch_beg + SEARCH_REGION;
if(srch_beg < p_buf_beg)
srch_beg = p_buf_beg;
if(srch_end + DELTA >= tmpl)
srch_end = tmpl - DELTA;
}
else {
// compression
srch_end = tmpl + (int)((float)TEMPLATE_SEG * (1.0 - factor)) + SEARCH_REGION/2;
srch_beg = srch_end - SEARCH_REGION;
if(srch_end + TEMPLATE_SEG > p_buf_end)
srch_end = p_buf_end - TEMPLATE_SEG;
if(srch_beg - DELTA < tmpl)
srch_beg = tmpl + DELTA;
}
// stop if search region size < 0
if (srch_beg >= srch_end)
break;
// find best correlation to tmpl in srch_beg..srch_end
srch = find_best_corr(tmpl,srch_beg,srch_end);
// merge original segment (starting from tmpl) and
// best correlation (starting from srch) into merge_buf
float f,v;
for(int k=0; k<TEMPLATE_SEG; k++){
f = 0.5 - 0.5 * cos( PI*float(k) / float(TEMPLATE_SEG) );
v = (float)srch[k] * f + (float)tmpl[k] * (1.0 - f);
if(v > 32767.)
v = 32767.;
else if(v < -32768.)
v = -32768.;
merge_buf[k] = (short)v;
}
// put merged segment into buffer
buffer_put( cur_ts, merge_buf, TEMPLATE_SEG);
// add after merged segment audio from after srch
buffer_put( cur_ts + TEMPLATE_SEG, srch + TEMPLATE_SEG,
p_buf_end - srch - TEMPLATE_SEG );
// size s has changed
s += tmpl - srch;
// go to next segment
cur_ts += TEMPLATE_SEG/2;
tmpl += TEMPLATE_SEG/2;
// calculate current factor
float act_fact = s / (float)packet_len;
#ifdef DEBUG_PLAYOUTBUF
DBG("at ts %u: new size = %u, ratio = %f, requested = %f\n", ts, s, act_fact, factor);
#endif
// break condition: coming to the end of the frame (with safety margin)
if(p_buf_end - tmpl < TEMPLATE_SEG + DELTA)
break;
// streched enough?
if((factor > 1.0) && (act_fact >= factor))
break;
else if((factor < 1.0) && (act_fact <= factor))
break;
// streched over maximum already?
else if(act_fact >= TSM_MAX_SCALE || f <= TSM_MIN_SCALE)
break;
}
return s;
}