Add libswresample.

Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 352 insertions, 0 deletions

diff --git a/libswresample/resample2.c b/libswresample/resample2.c
new file mode 100644
index 0000000000..8623877c7c
--- /dev/null
+++ b/libswresample/resample2.c
@@ -0,0 +1,352 @@
+/*
+ * audio resampling
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio resampling
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "libavutil/log.h"
+#include "swresample_internal.h"
+
+#ifndef CONFIG_RESAMPLE_HP
+#define FILTER_SHIFT 15
+
+#define FELEM int16_t
+#define FELEM2 int32_t
+#define FELEML int64_t
+#define FELEM_MAX INT16_MAX
+#define FELEM_MIN INT16_MIN
+#define WINDOW_TYPE 9
+#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
+#define FILTER_SHIFT 30
+
+#define FELEM int32_t
+#define FELEM2 int64_t
+#define FELEML int64_t
+#define FELEM_MAX INT32_MAX
+#define FELEM_MIN INT32_MIN
+#define WINDOW_TYPE 12
+#else
+#define FILTER_SHIFT 0
+
+#define FELEM double
+#define FELEM2 double
+#define FELEML double
+#define WINDOW_TYPE 24
+#endif
+
+
+typedef struct AVResampleContext{
+    const AVClass *av_class;
+    FELEM *filter_bank;
+    int filter_length;
+    int ideal_dst_incr;
+    int dst_incr;
+    int index;
+    int frac;
+    int src_incr;
+    int compensation_distance;
+    int phase_shift;
+    int phase_mask;
+    int linear;
+    double factor;
+}AVResampleContext;
+
+/**
+ * 0th order modified bessel function of the first kind.
+ */
+static double bessel(double x){
+    double v=1;
+    double lastv=0;
+    double t=1;
+    int i;
+    static const double inv[100]={
+ 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
+ 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
+ 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
+ 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
+ 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
+ 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
+ 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
+ 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
+ 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
+ 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
+    };
+
+    x= x*x/4;
+    for(i=0; v != lastv; i++){
+        lastv=v;
+        t *= x*inv[i];
+        v += t;
+    }
+    return v;
+}
+
+/**
+ * builds a polyphase filterbank.
+ * @param factor resampling factor
+ * @param scale wanted sum of coefficients for each filter
+ * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
+ * @return 0 on success, negative on error
+ */
+static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
+    int ph, i;
+    double x, y, w;
+    double *tab = av_malloc(tap_count * sizeof(*tab));
+    const int center= (tap_count-1)/2;
+
+    if (!tab)
+        return AVERROR(ENOMEM);
+
+    /* if upsampling, only need to interpolate, no filter */
+    if (factor > 1.0)
+        factor = 1.0;
+
+    for(ph=0;ph<phase_count;ph++) {
+        double norm = 0;
+        for(i=0;i<tap_count;i++) {
+            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
+            if (x == 0) y = 1.0;
+            else        y = sin(x) / x;
+            switch(type){
+            case 0:{
+                const float d= -0.5; //first order derivative = -0.5
+                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
+                if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
+                else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
+                break;}
+            case 1:
+                w = 2.0*x / (factor*tap_count) + M_PI;
+                y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
+                break;
+            default:
+                w = 2.0*x / (factor*tap_count*M_PI);
+                y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
+                break;
+            }
+
+            tab[i] = y;
+            norm += y;
+        }
+
+        /* normalize so that an uniform color remains the same */
+        for(i=0;i<tap_count;i++) {
+#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
+            filter[ph * tap_count + i] = tab[i] / norm;
+#else
+            filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
+#endif
+        }
+    }
+#if 0
+    {
+#define LEN 1024
+        int j,k;
+        double sine[LEN + tap_count];
+        double filtered[LEN];
+        double maxff=-2, minff=2, maxsf=-2, minsf=2;
+        for(i=0; i<LEN; i++){
+            double ss=0, sf=0, ff=0;
+            for(j=0; j<LEN+tap_count; j++)
+                sine[j]= cos(i*j*M_PI/LEN);
+            for(j=0; j<LEN; j++){
+                double sum=0;
+                ph=0;
+                for(k=0; k<tap_count; k++)
+                    sum += filter[ph * tap_count + k] * sine[k+j];
+                filtered[j]= sum / (1<<FILTER_SHIFT);
+                ss+= sine[j + center] * sine[j + center];
+                ff+= filtered[j] * filtered[j];
+                sf+= sine[j + center] * filtered[j];
+            }
+            ss= sqrt(2*ss/LEN);
+            ff= sqrt(2*ff/LEN);
+            sf= 2*sf/LEN;
+            maxff= FFMAX(maxff, ff);
+            minff= FFMIN(minff, ff);
+            maxsf= FFMAX(maxsf, sf);
+            minsf= FFMIN(minsf, sf);
+            if(i%11==0){
+                av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
+                minff=minsf= 2;
+                maxff=maxsf= -2;
+            }
+        }
+    }
+#endif
+
+    av_free(tab);
+    return 0;
+}
+
+AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
+    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
+    int phase_count= 1<<phase_shift;
+
+    if(!c || c->phase_shift!=phase_shift || c->linear!=linear || c->factor != factor
+         || c->filter_length!=FFMAX((int)ceil(filter_size/factor), 1)){
+    c= av_mallocz(sizeof(AVResampleContext));
+    if (!c)
+        return NULL;
+
+    c->phase_shift= phase_shift;
+    c->phase_mask= phase_count-1;
+    c->linear= linear;
+    c->factor= factor;
+
+    c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
+    c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
+    if (!c->filter_bank)
+        goto error;
+    if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
+        goto error;
+    memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
+    c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
+    }
+
+    c->compensation_distance= 0;
+    c->src_incr= out_rate;
+    c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
+    c->index= -phase_count*((c->filter_length-1)/2);
+    c->frac= 0;
+
+    return c;
+error:
+    av_free(c->filter_bank);
+    av_free(c);
+    return NULL;
+}
+
+void swr_resample_free(AVResampleContext **c){
+    if(!*c)
+        return;
+    av_freep(&(*c)->filter_bank);
+    av_freep(c);
+}
+
+void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
+    AVResampleContext *c= s->resample;
+//    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
+    c->compensation_distance= compensation_distance;
+    c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
+}
+
+int swr_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
+    int dst_index, i;
+    int index= c->index;
+    int frac= c->frac;
+    int dst_incr_frac= c->dst_incr % c->src_incr;
+    int dst_incr=      c->dst_incr / c->src_incr;
+    int compensation_distance= c->compensation_distance;
+
+  if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
+        int64_t index2= ((int64_t)index)<<32;
+        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
+        dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
+
+        for(dst_index=0; dst_index < dst_size; dst_index++){
+            dst[dst_index] = src[index2>>32];
+            index2 += incr;
+        }
+        frac += dst_index * dst_incr_frac;
+        index += dst_index * dst_incr;
+        index += frac / c->src_incr;
+        frac %= c->src_incr;
+  }else{
+    for(dst_index=0; dst_index < dst_size; dst_index++){
+        FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
+        int sample_index= index >> c->phase_shift;
+        FELEM2 val=0;
+
+        if(sample_index < 0){
+            for(i=0; i<c->filter_length; i++)
+                val += src[FFABS(sample_index + i) % src_size] * filter[i];
+        }else if(sample_index + c->filter_length > src_size){
+            break;
+        }else if(c->linear){
+            FELEM2 v2=0;
+            for(i=0; i<c->filter_length; i++){
+                val += src[sample_index + i] * (FELEM2)filter[i];
+                v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
+            }
+            val+=(v2-val)*(FELEML)frac / c->src_incr;
+        }else{
+            for(i=0; i<c->filter_length; i++){
+                val += src[sample_index + i] * (FELEM2)filter[i];
+            }
+        }
+
+#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
+        dst[dst_index] = av_clip_int16(lrintf(val));
+#else
+        val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
+        dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
+#endif
+
+        frac += dst_incr_frac;
+        index += dst_incr;
+        if(frac >= c->src_incr){
+            frac -= c->src_incr;
+            index++;
+        }
+
+        if(dst_index + 1 == compensation_distance){
+            compensation_distance= 0;
+            dst_incr_frac= c->ideal_dst_incr % c->src_incr;
+            dst_incr=      c->ideal_dst_incr / c->src_incr;
+        }
+    }
+  }
+    *consumed= FFMAX(index, 0) >> c->phase_shift;
+    if(index>=0) index &= c->phase_mask;
+
+    if(compensation_distance){
+        compensation_distance -= dst_index;
+        assert(compensation_distance > 0);
+    }
+    if(update_ctx){
+        c->frac= frac;
+        c->index= index;
+        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
+        c->compensation_distance= compensation_distance;
+    }
+#if 0
+    if(update_ctx && !c->compensation_distance){
+#undef rand
+        av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
+av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
+    }
+#endif
+
+    return dst_index;
+}
+
+int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
+    int i, ret= -1;
+
+    for(i=0; i<dst->ch_count; i++){
+        ret= swr_resample(c, dst->ch[i], src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
+    }
+
+    return ret;
+}