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authorAlex Converse <alex.converse@gmail.com>2009-07-08 20:01:31 +0000
committerAlex Converse <alex.converse@gmail.com>2009-07-08 20:01:31 +0000
commit78e65cd7726942a1615ead039abe0bfa79341212 (patch)
tree7003e32f0234d3fb6d7959e9f193e2ec733df5c6 /libavcodec
parent5e039e1b4c0fe25c76faa7ea107db60264edb757 (diff)
downloadffmpeg-78e65cd7726942a1615ead039abe0bfa79341212.tar.gz
Merge the AAC encoder from SoC svn. It is still considered experimental.
Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/aac.h19
-rw-r--r--libavcodec/aaccoder.c1037
-rw-r--r--libavcodec/aacenc.c402
-rw-r--r--libavcodec/aacenc.h71
-rw-r--r--libavcodec/aacpsy.c286
-rw-r--r--libavcodec/allcodecs.c2
-rw-r--r--libavcodec/psymodel.c130
-rw-r--r--libavcodec/psymodel.h158
9 files changed, 1991 insertions, 115 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 002fe0bc37..62819bc9b5 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -36,6 +36,7 @@ OBJS-$(CONFIG_VDPAU) += vdpau.o
# decoders/encoders/hardware accelerators
OBJS-$(CONFIG_AAC_DECODER) += aac.o aactab.o mpeg4audio.o aac_parser.o aac_ac3_parser.o
+OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacpsy.o aactab.o psymodel.o iirfilter.o mdct.o fft.o mpeg4audio.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index 95adcb3922..1d24a59667 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -116,6 +116,12 @@ typedef struct {
#define MAX_PREDICTORS 672
+#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
+#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
+#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
+#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
+#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
+
/**
* Individual Channel Stream
*/
@@ -126,6 +132,7 @@ typedef struct {
int num_window_groups;
uint8_t group_len[8];
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
+ const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
@@ -165,6 +172,7 @@ typedef struct {
typedef struct {
int num_pulse;
+ int start;
int pos[4];
int amp[4];
} Pulse;
@@ -189,11 +197,14 @@ typedef struct {
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
- enum BandType band_type[120]; ///< band types
+ Pulse pulse;
+ enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
+ int sf_idx[128]; ///< scalefactor indices (used by encoder)
+ uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
- DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
+ DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap
DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
@@ -203,7 +214,9 @@ typedef struct {
*/
typedef struct {
// CPE specific
- uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
+ int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
+ int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
+ uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
new file mode 100644
index 0000000000..763d3b90dd
--- /dev/null
+++ b/libavcodec/aaccoder.c
@@ -0,0 +1,1037 @@
+/*
+ * AAC coefficients encoder
+ * Copyright (C) 2008-2009 Konstantin Shishkov
+ *
+ * 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 libavcodec/aaccoder.c
+ * AAC coefficients encoder
+ */
+
+/***********************************
+ * TODOs:
+ * speedup quantizer selection
+ * add sane pulse detection
+ ***********************************/
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "aac.h"
+#include "aacenc.h"
+#include "aactab.h"
+
+/** bits needed to code codebook run value for long windows */
+static const uint8_t run_value_bits_long[64] = {
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
+ 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
+ 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
+};
+
+/** bits needed to code codebook run value for short windows */
+static const uint8_t run_value_bits_short[16] = {
+ 3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
+};
+
+static const uint8_t* run_value_bits[2] = {
+ run_value_bits_long, run_value_bits_short
+};
+
+
+/**
+ * Quantize one coefficient.
+ * @return absolute value of the quantized coefficient
+ * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
+ */
+static av_always_inline int quant(float coef, const float Q)
+{
+ return pow(coef * Q, 0.75) + 0.4054;
+}
+
+static void quantize_bands(int (*out)[2], const float *in, const float *scaled, int size, float Q34, int is_signed, int maxval)
+{
+ int i;
+ double qc;
+ for (i = 0; i < size; i++) {
+ qc = scaled[i] * Q34;
+ out[i][0] = (int)FFMIN((int)qc, maxval);
+ out[i][1] = (int)FFMIN((int)(qc + 0.4054), maxval);
+ if (is_signed && in[i] < 0.0f) {
+ out[i][0] = -out[i][0];
+ out[i][1] = -out[i][1];
+ }
+ }
+}
+
+static void abs_pow34_v(float *out, const float* in, const int size)
+{
+#ifndef USE_REALLY_FULL_SEARCH
+ int i;
+ for (i = 0; i < size; i++) {
+ out[i] = pow(fabsf(in[i]), 0.75);
+ }
+#endif /* USE_REALLY_FULL_SEARCH */
+}
+
+static av_always_inline int quant2(float coef, const float Q)
+{
+ return pow(coef * Q, 0.75);
+}
+
+static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
+static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
+
+/**
+ * Calculate rate distortion cost for quantizing with given codebook
+ *
+ * @return quantization distortion
+ */
+static float quantize_band_cost(struct AACEncContext *s, const float *in, const float *scaled, int size, int scale_idx, int cb,
+ const float lambda, const float uplim, int *bits)
+{
+ const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+ const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
+ const float CLIPPED_ESCAPE = 165140.0f*IQ;
+ int i, j, k;
+ float cost = 0;
+ const int dim = cb < FIRST_PAIR_BT ? 4 : 2;
+ int resbits = 0;
+#ifndef USE_REALLY_FULL_SEARCH
+ const float Q34 = pow(Q, 0.75);
+ const int range = aac_cb_range[cb];
+ const int maxval = aac_cb_maxval[cb];
+ int offs[4];
+#endif /* USE_REALLY_FULL_SEARCH */
+
+ if(!cb){
+ for(i = 0; i < size; i++)
+ cost += in[i]*in[i]*lambda;
+ return cost;
+ }
+#ifndef USE_REALLY_FULL_SEARCH
+ offs[0] = 1;
+ for(i = 1; i < dim; i++)
+ offs[i] = offs[i-1]*range;
+ quantize_bands(s->qcoefs, in, scaled, size, Q34, !IS_CODEBOOK_UNSIGNED(cb), maxval);
+#endif /* USE_REALLY_FULL_SEARCH */
+ for(i = 0; i < size; i += dim){
+ float mincost;
+ int minidx = 0;
+ int minbits = 0;
+ const float *vec;
+#ifndef USE_REALLY_FULL_SEARCH
+ int (*quants)[2] = &s->qcoefs[i];
+ mincost = 0.0f;
+ for(j = 0; j < dim; j++){
+ mincost += in[i+j]*in[i+j]*lambda;
+ }
+ minidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
+ minbits = ff_aac_spectral_bits[cb-1][minidx];
+ mincost += minbits;
+ for(j = 0; j < (1<<dim); j++){
+ float rd = 0.0f;
+ int curbits;
+ int curidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
+ int same = 0;
+ for(k = 0; k < dim; k++){
+ if((j & (1 << k)) && quants[k][0] == quants[k][1]){
+ same = 1;
+ break;
+ }
+ }
+ if(same)
+ continue;
+ for(k = 0; k < dim; k++)
+ curidx += quants[k][!!(j & (1 << k))] * offs[dim - 1 - k];
+ curbits = ff_aac_spectral_bits[cb-1][curidx];
+ vec = &ff_aac_codebook_vectors[cb-1][curidx*dim];
+#else
+ mincost = INFINITY;
+ vec = ff_aac_codebook_vectors[cb-1];
+ for(j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim){
+ float rd = 0.0f;
+ int curbits = ff_aac_spectral_bits[cb-1][j];
+#endif /* USE_REALLY_FULL_SEARCH */
+ if(IS_CODEBOOK_UNSIGNED(cb)){
+ for(k = 0; k < dim; k++){
+ float t = fabsf(in[i+k]);
+ float di;
+ //do not code with escape sequence small values
+ if(vec[k] == 64.0f && t < 39.0f*IQ){
+ rd = INFINITY;
+ break;
+ }
+ if(vec[k] == 64.0f){//FIXME: slow
+ if (t >= CLIPPED_ESCAPE) {
+ di = t - CLIPPED_ESCAPE;
+ curbits += 21;
+ }else{
+ int c = av_clip(quant(t, Q), 0, 8191);
+ di = t - c*cbrt(c)*IQ;
+ curbits += av_log2(c)*2 - 4 + 1;
+ }
+ }else{
+ di = t - vec[k]*IQ;
+ }
+ if(vec[k] != 0.0f)
+ curbits++;
+ rd += di*di*lambda;
+ }
+ }else{
+ for(k = 0; k < dim; k++){
+ float di = in[i+k] - vec[k]*IQ;
+ rd += di*di*lambda;
+ }
+ }
+ rd += curbits;
+ if(rd < mincost){
+ mincost = rd;
+ minidx = j;
+ minbits = curbits;
+ }
+ }
+ cost += mincost;
+ resbits += minbits;
+ if(cost >= uplim)
+ return uplim;
+ }
+
+ if(bits)
+ *bits = resbits;
+ return cost;
+}
+
+static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+ int scale_idx, int cb, const float lambda)
+{
+ const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+ const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
+ const float CLIPPED_ESCAPE = 165140.0f*IQ;
+ const int dim = (cb < FIRST_PAIR_BT) ? 4 : 2;
+ int i, j, k;
+#ifndef USE_REALLY_FULL_SEARCH
+ const float Q34 = pow(Q, 0.75);
+ const int range = aac_cb_range[cb];
+ const int maxval = aac_cb_maxval[cb];
+ int offs[4];
+ float *scaled = s->scoefs;
+#endif /* USE_REALLY_FULL_SEARCH */
+
+//START_TIMER
+ if(!cb)
+ return;
+
+#ifndef USE_REALLY_FULL_SEARCH
+ offs[0] = 1;
+ for(i = 1; i < dim; i++)
+ offs[i] = offs[i-1]*range;
+ abs_pow34_v(scaled, in, size);
+ quantize_bands(s->qcoefs, in, scaled, size, Q34, !IS_CODEBOOK_UNSIGNED(cb), maxval);
+#endif /* USE_REALLY_FULL_SEARCH */
+ for(i = 0; i < size; i += dim){
+ float mincost;
+ int minidx = 0;
+ int minbits = 0;
+ const float *vec;
+#ifndef USE_REALLY_FULL_SEARCH
+ int (*quants)[2] = &s->qcoefs[i];
+ mincost = 0.0f;
+ for(j = 0; j < dim; j++){
+ mincost += in[i+j]*in[i+j]*lambda;
+ }
+ minidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
+ minbits = ff_aac_spectral_bits[cb-1][minidx];
+ mincost += minbits;
+ for(j = 0; j < (1<<dim); j++){
+ float rd = 0.0f;
+ int curbits;
+ int curidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
+ int same = 0;
+ for(k = 0; k < dim; k++){
+ if((j & (1 << k)) && quants[k][0] == quants[k][1]){
+ same = 1;
+ break;
+ }
+ }
+ if(same)
+ continue;
+ for(k = 0; k < dim; k++)
+ curidx += quants[k][!!(j & (1 << k))] * offs[dim - 1 - k];
+ curbits = ff_aac_spectral_bits[cb-1][curidx];
+ vec = &ff_aac_codebook_vectors[cb-1][curidx*dim];
+#else
+ vec = ff_aac_codebook_vectors[cb-1];
+ mincost = INFINITY;
+ for(j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim){
+ float rd = 0.0f;
+ int curbits = ff_aac_spectral_bits[cb-1][j];
+ int curidx = j;
+#endif /* USE_REALLY_FULL_SEARCH */
+ if(IS_CODEBOOK_UNSIGNED(cb)){
+ for(k = 0; k < dim; k++){
+ float t = fabsf(in[i+k]);
+ float di;
+ //do not code with escape sequence small values
+ if(vec[k] == 64.0f && t < 39.0f*IQ){
+ rd = INFINITY;
+ break;
+ }
+ if(vec[k] == 64.0f){//FIXME: slow
+ if (t >= CLIPPED_ESCAPE) {
+ di = t - CLIPPED_ESCAPE;
+ curbits += 21;
+ }else{
+ int c = av_clip(quant(t, Q), 0, 8191);
+ di = t - c*cbrt(c)*IQ;
+ curbits += av_log2(c)*2 - 4 + 1;
+ }
+ }else{
+ di = t - vec[k]*IQ;
+ }
+ if(vec[k] != 0.0f)
+ curbits++;
+ rd += di*di*lambda;
+ }
+ }else{
+ for(k = 0; k < dim; k++){
+ float di = in[i+k] - vec[k]*IQ;
+ rd += di*di*lambda;
+ }
+ }
+ rd += curbits;
+ if(rd < mincost){
+ mincost = rd;
+ minidx = curidx;
+ minbits = curbits;
+ }
+ }
+ put_bits(pb, ff_aac_spectral_bits[cb-1][minidx], ff_aac_spectral_codes[cb-1][minidx]);
+ if(IS_CODEBOOK_UNSIGNED(cb))
+ for(j = 0; j < dim; j++)
+ if(ff_aac_codebook_vectors[cb-1][minidx*dim+j] != 0.0f)
+ put_bits(pb, 1, in[i+j] < 0.0f);
+ if(cb == ESC_BT){
+ for(j = 0; j < 2; j++){
+ if(ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f){
+ int coef = av_clip(quant(fabsf(in[i+j]), Q), 0, 8191);
+ int len = av_log2(coef);
+
+ put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
+ put_bits(pb, len, coef & ((1 << len) - 1));
+ }
+ }
+ }
+ }
+//STOP_TIMER("quantize_and_encode")
+}
+
+/**
+ * structure used in optimal codebook search
+ */
+typedef struct BandCodingPath {
+ int prev_idx; ///< pointer to the previous path point
+ int codebook; ///< codebook for coding band run
+ float cost; ///< path cost
+ int run;
+} BandCodingPath;
+
+/**
+ * Encode band info for single window group bands.
+ */
+static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
+ int win, int group_len, const float lambda)
+{
+ BandCodingPath path[120][12];
+ int w, swb, cb, start, start2, size;
+ int i, j;
+ const int max_sfb = sce->ics.max_sfb;
+ const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
+ const int run_esc = (1 << run_bits) - 1;
+ int idx, ppos, count;
+ int stackrun[120], stackcb[120], stack_len;
+ float next_minrd = INFINITY;
+ int next_mincb = 0;
+
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+ start = win*128;
+ for(cb = 0; cb < 12; cb++){
+ path[0][cb].cost = 0.0f;
+ path[0][cb].prev_idx = -1;
+ path[0][cb].run = 0;
+ }
+ for(swb = 0; swb < max_sfb; swb++){
+ start2 = start;
+ size = sce->ics.swb_sizes[swb];
+ if(sce->zeroes[win*16 + swb]){
+ for(cb = 0; cb < 12; cb++){
+ path[swb+1][cb].prev_idx = cb;
+ path[swb+1][cb].cost = path[swb][cb].cost;
+ path[swb+1][cb].run = path[swb][cb].run + 1;
+ }
+ }else{
+ float minrd = next_minrd;
+ int mincb = next_mincb;
+ next_minrd = INFINITY;
+ next_mincb = 0;
+ for(cb = 0; cb < 12; cb++){
+ float cost_stay_here, cost_get_here;
+ float rd = 0.0f;
+ for(w = 0; w < group_len; w++){
+ FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(win+w)*16+swb];
+ rd += quantize_band_cost(s, sce->coeffs + start + w*128,
+ s->scoefs + start + w*128, size,
+ sce->sf_idx[(win+w)*16+swb], cb,
+ lambda / band->threshold, INFINITY, NULL);
+ }
+ cost_stay_here = path[swb][cb].cost + rd;
+ cost_get_here = minrd + rd + run_bits + 4;
+ if( run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
+ != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
+ cost_stay_here += run_bits;
+ if (cost_get_here < cost_stay_here) {
+ path[swb+1][cb].prev_idx = mincb;
+ path[swb+1][cb].cost = cost_get_here;
+ path[swb+1][cb].run = 1;
+ } else {
+ path[swb+1][cb].prev_idx = cb;
+ path[swb+1][cb].cost = cost_stay_here;
+ path[swb+1][cb].run = path[swb][cb].run + 1;
+ }
+ if (path[swb+1][cb].cost < next_minrd) {
+ next_minrd = path[swb+1][cb].cost;
+ next_mincb = cb;
+ }
+ }
+ }
+ start += sce->ics.swb_sizes[swb];
+ }
+
+ //convert resulting path from backward-linked list
+ stack_len = 0;
+ idx = 0;
+ for(cb = 1; cb < 12; cb++){
+ if(path[max_sfb][cb].cost < path[max_sfb][idx].cost)
+ idx = cb;
+ }
+ ppos = max_sfb;
+ while(ppos > 0){
+ cb = idx;
+ stackrun[stack_len] = path[ppos][cb].run;
+ stackcb [stack_len] = cb;
+ idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
+ ppos -= path[ppos][cb].run;
+ stack_len++;
+ }
+ //perform actual band info encoding
+ start = 0;
+ for(i = stack_len - 1; i >= 0; i--){
+ put_bits(&s->pb, 4, stackcb[i]);
+ count = stackrun[i];
+ memset(sce->zeroes + win*16 + start, !stackcb[i], count);
+ //XXX: memset when band_type is also uint8_t
+ for(j = 0; j < count; j++){
+ sce->band_type[win*16 + start] = stackcb[i];
+ start++;
+ }
+ while(count >= run_esc){
+ put_bits(&s->pb, run_bits, run_esc);
+ count -= run_esc;
+ }
+ put_bits(&s->pb, run_bits, count);
+ }
+}
+
+static void encode_window_bands_info_fixed(AACEncContext *s, SingleChannelElement *sce,
+ int win, int group_len, const float lambda)
+{
+ encode_window_bands_info(s, sce, win, group_len, 1.0f);
+}
+
+
+typedef struct TrellisPath {
+ float cost;
+ int prev;
+ int min_val;
+ int max_val;
+} TrellisPath;
+
+static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce, const float lambda)
+{
+ int q, w, w2, g, start = 0;
+ int i;
+ int idx;
+ TrellisPath paths[256*121];
+ int bandaddr[121];
+ int minq;
+ float mincost;
+
+ for(i = 0; i < 256; i++){
+ paths[i].cost = 0.0f;
+ paths[i].prev = -1;
+ paths[i].min_val = i;
+ paths[i].max_val = i;
+ }
+ for(i = 256; i < 256*121; i++){
+ paths[i].cost = INFINITY;
+ paths[i].prev = -2;
+ paths[i].min_val = INT_MAX;
+ paths[i].max_val = 0;
+ }
+ idx = 256;
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ const float *coefs = sce->coeffs + start;
+ float qmin, qmax;
+ int nz = 0;
+
+ bandaddr[idx >> 8] = w*16+g;
+ qmin = INT_MAX;
+ qmax = 0.0f;
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
+ if(band->energy <= band->threshold || band->threshold == 0.0f){
+ sce->zeroes[(w+w2)*16+g] = 1;
+ continue;
+ }
+ sce->zeroes[(w+w2)*16+g] = 0;
+ nz = 1;
+ for(i = 0; i < sce->ics.swb_sizes[g]; i++){
+ float t = fabsf(coefs[w2*128+i]);
+ if(t > 0.0f) qmin = fminf(qmin, t);
+ qmax = fmaxf(qmax, t);
+ }
+ }
+ if(nz){
+ int minscale, maxscale;
+ float minrd = INFINITY;
+ //minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
+ minscale = av_clip_uint8(log2(qmin)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
+ //maximum scalefactor index is when maximum coefficient after quantizing is still not zero
+ maxscale = av_clip_uint8(log2(qmax)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
+ for(q = minscale; q < maxscale; q++){
+ float dists[12], dist;
+ memset(dists, 0, sizeof(dists));
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
+ int cb;
+ for(cb = 0; cb <= ESC_BT; cb++){
+ dists[cb] += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
+ q, cb, lambda / band->threshold, INFINITY, NULL);
+ }
+ }
+ dist = dists[0];
+ for(i = 1; i <= ESC_BT; i++)
+ dist = fminf(dist, dists[i]);
+ minrd = fminf(minrd, dist);
+
+ for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
+ float cost;
+ int minv, maxv;
+ if(isinf(paths[idx - 256 + i].cost))
+ continue;
+ cost = paths[idx - 256 + i].cost + dist
+ + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
+ minv = FFMIN(paths[idx - 256 + i].min_val, q);
+ maxv = FFMAX(paths[idx - 256 + i].max_val, q);
+ if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
+ paths[idx + q].cost = cost;
+ paths[idx + q].prev = idx - 256 + i;
+ paths[idx + q].min_val = minv;
+ paths[idx + q].max_val = maxv;
+ }
+ }
+ }
+ }else{
+ for(q = 0; q < 256; q++){
+ if(!isinf(paths[idx - 256 + q].cost)){
+ paths[idx + q].cost = paths[idx - 256 + q].cost + 1;
+ paths[idx + q].prev = idx - 256 + q;
+ paths[idx + q].min_val = FFMIN(paths[idx - 256 + q].min_val, q);
+ paths[idx + q].max_val = FFMAX(paths[idx - 256 + q].max_val, q);
+ continue;
+ }
+ for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
+ float cost;
+ int minv, maxv;
+ if(isinf(paths[idx - 256 + i].cost))
+ continue;
+ cost = paths[idx - 256 + i].cost + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
+ minv = FFMIN(paths[idx - 256 + i].min_val, q);
+ maxv = FFMAX(paths[idx - 256 + i].max_val, q);
+ if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
+ paths[idx + q].cost = cost;
+ paths[idx + q].prev = idx - 256 + i;
+ paths[idx + q].min_val = minv;
+ paths[idx + q].max_val = maxv;
+ }
+ }
+ }
+ }
+ sce->zeroes[w*16+g] = !nz;
+ start += sce->ics.swb_sizes[g];
+ idx += 256;
+ }
+ }
+ idx -= 256;
+ mincost = paths[idx].cost;
+ minq = idx;
+ for(i = 1; i < 256; i++){
+ if(paths[idx + i].cost < mincost){
+ mincost = paths[idx + i].cost;
+ minq = idx + i;
+ }
+ }
+ while(minq >= 256){
+ sce->sf_idx[bandaddr[minq>>8]] = minq & 0xFF;
+ minq = paths[minq].prev;
+ }
+ //set the same quantizers inside window groups
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+ for(g = 0; g < sce->ics.num_swb; g++)
+ for(w2 = 1; w2 < sce->ics.group_len[w]; w2++)
+ sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
+}
+
+/**
+ * two-loop quantizers search taken from ISO 13818-7 Appendix C
+ */
+static void search_for_quantizers_twoloop(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce, const float lambda)
+{
+ int start = 0, i, w, w2, g;
+ int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
+ float dists[128], uplims[128];
+ int fflag, minscaler;
+ int its = 0;
+ int allz = 0;
+ float minthr = INFINITY;
+
+ //XXX: some heuristic to determine initial quantizers will reduce search time
+ memset(dists, 0, sizeof(dists));
+ //determine zero bands and upper limits
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ for(g = 0; g < sce->ics.num_swb; g++){
+ int nz = 0;
+ float uplim = 0.0f;
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
+ uplim += band->threshold;
+ if(band->energy <= band->threshold || band->threshold == 0.0f){
+ sce->zeroes[(w+w2)*16+g] = 1;
+ continue;
+ }
+ nz = 1;
+ }
+ uplims[w*16+g] = uplim *512;
+ sce->zeroes[w*16+g] = !nz;
+ if(nz)
+ minthr = fminf(minthr, uplim);
+ allz = FFMAX(allz, nz);
+ }
+ }
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ for(g = 0; g < sce->ics.num_swb; g++){
+ if(sce->zeroes[w*16+g]){
+ sce->sf_idx[w*16+g] = SCALE_ONE_POS;
+ continue;
+ }
+ sce->sf_idx[w*16+g] = SCALE_ONE_POS + fminf(log2(uplims[w*16+g]/minthr)*4,59);
+ }
+ }
+
+ if(!allz)
+ return;
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+ //perform two-loop search
+ //outer loop - improve quality
+ do{
+ int tbits, qstep;
+ minscaler = sce->sf_idx[0];
+ //inner loop - quantize spectrum to fit into given number of bits
+ qstep = its ? 1 : 32;
+ do{
+ int prev = -1;
+ tbits = 0;
+ fflag = 0;
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ const float *coefs = sce->coeffs + start;
+ const float *scaled = s->scoefs + start;
+ int bits = 0;
+ int cb;
+ float mindist = INFINITY;
+ int minbits = 0;
+
+ if(sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218)
+ continue;
+ minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
+ for(cb = 0; cb <= ESC_BT; cb++){
+ float dist = 0.0f;
+ int bb = 0;
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ int b;
+ dist += quantize_band_cost(s, coefs + w2*128,
+ scaled + w2*128,
+ sce->ics.swb_sizes[g],
+ sce->sf_idx[w*16+g],
+ ESC_BT,
+ 1.0,
+ INFINITY,
+ &b);
+ bb += b;
+ }
+ if(dist < mindist){
+ mindist = dist;
+ minbits = bb;
+ }
+ }
+ dists[w*16+g] = mindist - minbits;
+ bits = minbits;
+ if(prev != -1){
+ bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
+ }
+ tbits += bits;
+ start += sce->ics.swb_sizes[g];
+ prev = sce->sf_idx[w*16+g];
+ }
+ }
+ if(tbits > destbits){
+ for(i = 0; i < 128; i++){
+ if(sce->sf_idx[i] < 218 - qstep){
+ sce->sf_idx[i] += qstep;
+ }
+ }
+ }else{
+ for(i = 0; i < 128; i++){
+ if(sce->sf_idx[i] > 60 - qstep){
+ sce->sf_idx[i] -= qstep;
+ }
+ }
+ }
+ qstep >>= 1;
+ if(!qstep && tbits > destbits*1.02)
+ qstep = 1;
+ if(sce->sf_idx[0] >= 217)break;
+ }while(qstep);
+
+ fflag = 0;
+ minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ int prevsc = sce->sf_idx[w*16+g];
+ if(dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60)
+ sce->sf_idx[w*16+g]--;
+ sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
+ sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
+ if(sce->sf_idx[w*16+g] != prevsc)
+ fflag = 1;
+ }
+ }
+ its++;
+ }while(fflag && its < 10);
+}
+
+static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce, const float lambda)
+{
+ int start = 0, i, w, w2, g;
+ float uplim[128], maxq[128];
+ int minq, maxsf;
+ float distfact = ((sce->ics.num_windows > 1) ? 85.80 : 147.84) / lambda;
+ int last = 0, lastband = 0, curband = 0;
+ float avg_energy = 0.0;
+ if(sce->ics.num_windows == 1){
+ start = 0;
+ for(i = 0; i < 1024; i++){
+ if(i - start >= sce->ics.swb_sizes[curband]){
+ start += sce->ics.swb_sizes[curband];
+ curband++;
+ }
+ if(sce->coeffs[i]){
+ avg_energy += sce->coeffs[i] * sce->coeffs[i];
+ last = i;
+ lastband = curband;
+ }
+ }
+ }else{
+ for(w = 0; w < 8; w++){
+ const float *coeffs = sce->coeffs + w*128;
+ start = 0;
+ for(i = 0; i < 128; i++){
+ if(i - start >= sce->ics.swb_sizes[curband]){
+ start += sce->ics.swb_sizes[curband];
+ curband++;
+ }
+ if(coeffs[i]){
+ avg_energy += coeffs[i] * coeffs[i];
+ last = FFMAX(last, i);
+ lastband = FFMAX(lastband, curband);
+ }
+ }
+ }
+ }
+ last++;
+ avg_energy /= last;
+ if(avg_energy == 0.0f){
+ for(i = 0; i < FF_ARRAY_ELEMS(sce->sf_idx); i++)
+ sce->sf_idx[i] = SCALE_ONE_POS;
+ return;
+ }
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ float *coefs = sce->coeffs + start;
+ const int size = sce->ics.swb_sizes[g];
+ int start2 = start, end2 = start + size, peakpos = start;
+ float maxval = -1, thr = 0.0f, t;
+ maxq[w*16+g] = 0.0f;
+ if(g > lastband){
+ maxq[w*16+g] = 0.0f;
+ start += size;
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++)
+ memset(coefs + w2*128, 0, sizeof(coefs[0])*size);
+ continue;
+ }
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ for(i = 0; i < size; i++){
+ float t = coefs[w2*128+i]*coefs[w2*128+i];
+ maxq[w*16+g] = fmaxf(maxq[w*16+g], fabsf(coefs[w2*128 + i]));
+ thr += t;
+ if(sce->ics.num_windows == 1 && maxval < t){
+ maxval = t;
+ peakpos = start+i;
+ }
+ }
+ }
+ if(sce->ics.num_windows == 1){
+ start2 = FFMAX(peakpos - 2, start2);
+ end2 = FFMIN(peakpos + 3, end2);
+ }else{
+ start2 -= start;
+ end2 -= start;
+ }
+ start += size;
+ thr = pow(thr / (avg_energy * (end2 - start2)), 0.3 + 0.1*(lastband - g) / lastband);
+ t = 1.0 - (1.0 * start2 / last);
+ uplim[w*16+g] = distfact / (1.4 * thr + t*t*t + 0.075);
+ }
+ }
+ memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ const float *coefs = sce->coeffs + start;
+ const float *scaled = s->scoefs + start;
+ const int size = sce->ics.swb_sizes[g];
+ int scf, prev_scf, step;
+ int min_scf = 0, max_scf = 255;
+ float curdiff;
+ if(maxq[w*16+g] < 21.544){
+ sce->zeroes[w*16+g] = 1;
+ start += size;
+ continue;
+ }
+ sce->zeroes[w*16+g] = 0;
+ scf = prev_scf = av_clip(SCALE_ONE_POS - SCALE_DIV_512 - log2(1/maxq[w*16+g])*16/3, 60, 218);
+ step = 16;
+ for(;;){
+ float dist = 0.0f;
+ int quant_max;
+
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ int b;
+ dist += quantize_band_cost(s, coefs + w2*128,
+ scaled + w2*128,
+ sce->ics.swb_sizes[g],
+ scf,
+ ESC_BT,
+ 1.0,
+ INFINITY,
+ &b);
+ dist -= b;
+ }
+ dist *= 1.0f/512.0f;
+ quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
+ if(quant_max >= 8191){ // too much, return to the previous quantizer
+ sce->sf_idx[w*16+g] = prev_scf;
+ break;
+ }
+ prev_scf = scf;
+ curdiff = fabsf(dist - uplim[w*16+g]);
+ if(curdiff == 0.0f)
+ step = 0;
+ else
+ step = fabsf(log2(curdiff));
+ if(dist > uplim[w*16+g])
+ step = -step;
+ if(FFABS(step) <= 1 || (step > 0 && scf >= max_scf) || (step < 0 && scf <= min_scf)){
+ sce->sf_idx[w*16+g] = scf;
+ break;
+ }
+ scf += step;
+ if(step > 0)
+ min_scf = scf;
+ else
+ max_scf = scf;
+ }
+ start += size;
+ }
+ }
+ minq = sce->sf_idx[0] ? sce->sf_idx[0] : INT_MAX;
+ for(i = 1; i < 128; i++){
+ if(!sce->sf_idx[i])
+ sce->sf_idx[i] = sce->sf_idx[i-1];
+ else
+ minq = FFMIN(minq, sce->sf_idx[i]);
+ }
+ if(minq == INT_MAX) minq = 0;
+ minq = FFMIN(minq, SCALE_MAX_POS);
+ maxsf = FFMIN(minq + SCALE_MAX_DIFF, SCALE_MAX_POS);
+ for(i = 126; i >= 0; i--){
+ if(!sce->sf_idx[i])
+ sce->sf_idx[i] = sce->sf_idx[i+1];
+ sce->sf_idx[i] = av_clip(sce->sf_idx[i], minq, maxsf);
+ }
+}
+
+static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce, const float lambda)
+{
+ int start = 0, i, w, w2, g;
+ int minq = 255;
+
+ memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ start = w*128;
+ for(g = 0; g < sce->ics.num_swb; g++){
+ for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
+ FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
+ if(band->energy <= band->threshold){
+ sce->sf_idx[(w+w2)*16+g] = 218;
+ sce->zeroes[(w+w2)*16+g] = 1;
+ }else{
+ sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - SCALE_DIV_512 + log2(band->threshold), 80, 218);
+ sce->zeroes[(w+w2)*16+g] = 0;
+ }
+ minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]);
+ }
+ }
+ }
+ for(i = 0; i < 128; i++){
+ sce->sf_idx[i] = 140;//av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1);
+ }
+ //set the same quantizers inside window groups
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+ for(g = 0; g < sce->ics.num_swb; g++)
+ for(w2 = 1; w2 < sce->ics.group_len[w]; w2++)
+ sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
+}
+
+static void search_for_ms(AACEncContext *s, ChannelElement *cpe, const float lambda)
+{
+ int start = 0, i, w, w2, g;
+ float M[128], S[128];
+ float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
+ SingleChannelElement *sce0 = &cpe->ch[0];
+ SingleChannelElement *sce1 = &cpe->ch[1];
+ if(!cpe->common_window)
+ return;
+ for(w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]){
+ for(g = 0; g < sce0->ics.num_swb; g++){
+ if(!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]){
+ float dist1 = 0.0f, dist2 = 0.0f;
+ for(w2 = 0; w2 < sce0->ics.group_len[w]; w2++){
+ FFPsyBand *band0 = &s->psy.psy_bands[(s->cur_channel+0)*PSY_MAX_BANDS+(w+w2)*16+g];
+ FFPsyBand *band1 = &s->psy.psy_bands[(s->cur_channel+1)*PSY_MAX_BANDS+(w+w2)*16+g];
+ float minthr = fminf(band0->threshold, band1->threshold);
+ float maxthr = fmaxf(band0->threshold, band1->threshold);
+ for(i = 0; i < sce0->ics.swb_sizes[g]; i++){
+ M[i] = (sce0->coeffs[start+w2*128+i]
+ + sce1->coeffs[start+w2*128+i])*0.5;
+ S[i] = sce0->coeffs[start+w2*128+i]
+ - sce1->coeffs[start+w2*128+i];
+ }
+ abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
+ abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
+ abs_pow34_v(M34, M, sce0->ics.swb_sizes[g]);
+ abs_pow34_v(S34, S, sce0->ics.swb_sizes[g]);
+ dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
+ L34,
+ sce0->ics.swb_sizes[g],
+ sce0->sf_idx[(w+w2)*16+g],
+ sce0->band_type[(w+w2)*16+g],
+ lambda / band0->threshold, INFINITY, NULL);
+ dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
+ R34,
+ sce1->ics.swb_sizes[g],
+ sce1->sf_idx[(w+w2)*16+g],
+ sce1->band_type[(w+w2)*16+g],
+ lambda / band1->threshold, INFINITY, NULL);
+ dist2 += quantize_band_cost(s, M,
+ M34,
+ sce0->ics.swb_sizes[g],
+ sce0->sf_idx[(w+w2)*16+g],
+ sce0->band_type[(w+w2)*16+g],
+ lambda / maxthr, INFINITY, NULL);
+ dist2 += quantize_band_cost(s, S,
+ S34,
+ sce1->ics.swb_sizes[g],
+ sce1->sf_idx[(w+w2)*16+g],
+ sce1->band_type[(w+w2)*16+g],
+ lambda / minthr, INFINITY, NULL);
+ }
+ cpe->ms_mask[w*16+g] = dist2 < dist1;
+ }
+ start += sce0->ics.swb_sizes[g];
+ }
+ }
+}
+
+AACCoefficientsEncoder ff_aac_coders[] = {
+ {
+ search_for_quantizers_faac,
+ encode_window_bands_info_fixed,
+ quantize_and_encode_band,
+// search_for_ms,
+ },
+ {
+ search_for_quantizers_anmr,
+ encode_window_bands_info,
+ quantize_and_encode_band,
+// search_for_ms,
+ },
+ {
+ search_for_quantizers_twoloop,
+ encode_window_bands_info,
+ quantize_and_encode_band,
+// search_for_ms,
+ },
+ {
+ search_for_quantizers_fast,
+ encode_window_bands_info,
+ quantize_and_encode_band,
+// search_for_ms,
+ },
+};
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 5537b7eac4..bc18b73a2e 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -26,19 +26,20 @@
/***********************************
* TODOs:
- * psy model selection with some option
* add sane pulse detection
* add temporal noise shaping
***********************************/
#include "avcodec.h"
-#include "get_bits.h"
+#include "put_bits.h"
#include "dsputil.h"
#include "mpeg4audio.h"
-#include "aacpsy.h"
#include "aac.h"
#include "aactab.h"
+#include "aacenc.h"
+
+#include "psymodel.h"
static const uint8_t swb_size_1024_96[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
@@ -83,7 +84,7 @@ static const uint8_t swb_size_1024_8[] = {
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
-static const uint8_t * const swb_size_1024[] = {
+static const uint8_t *swb_size_1024[] = {
swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
@@ -110,7 +111,7 @@ static const uint8_t swb_size_128_8[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
-static const uint8_t * const swb_size_128[] = {
+static const uint8_t *swb_size_128[] = {
/* the last entry on the following row is swb_size_128_64 but is a
duplicate of swb_size_128_96 */
swb_size_128_96, swb_size_128_96, swb_size_128_96,
@@ -119,23 +120,6 @@ static const uint8_t * const swb_size_128[] = {
swb_size_128_16, swb_size_128_16, swb_size_128_8
};
-/** bits needed to code codebook run value for long windows */
-static const uint8_t run_value_bits_long[64] = {
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
- 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
- 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
-};
-
-/** bits needed to code codebook run value for short windows */
-static const uint8_t run_value_bits_short[16] = {
- 3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
-};
-
-static const uint8_t* const run_value_bits[2] = {
- run_value_bits_long, run_value_bits_short
-};
-
/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
{1, TYPE_SCE}, // 1 channel - single channel element
@@ -147,33 +131,6 @@ static const uint8_t aac_chan_configs[6][5] = {
};
/**
- * structure used in optimal codebook search
- */
-typedef struct BandCodingPath {
- int prev_idx; ///< pointer to the previous path point
- int codebook; ///< codebook for coding band run
- int bits; ///< number of bit needed to code given number of bands
-} BandCodingPath;
-
-/**
- * AAC encoder context
- */
-typedef struct {
- PutBitContext pb;
- MDCTContext mdct1024; ///< long (1024 samples) frame transform context
- MDCTContext mdct128; ///< short (128 samples) frame transform context
- DSPContext dsp;
- DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
- int16_t* samples; ///< saved preprocessed input
-
- int samplerate_index; ///< MPEG-4 samplerate index
-
- ChannelElement *cpe; ///< channel elements
- AACPsyContext psy; ///< psychoacoustic model context
- int last_frame;
-} AACEncContext;
-
-/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
@@ -197,6 +154,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i;
+ const uint8_t *sizes[2];
+ int lengths[2];
avctx->frame_size = 1024;
@@ -224,25 +183,90 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
- if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
- aac_chan_configs[avctx->channels-1][0], 0,
- swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
- av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
- return -1;
- }
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
+
+ sizes[0] = swb_size_1024[i];
+ sizes[1] = swb_size_128[i];
+ lengths[0] = ff_aac_num_swb_1024[i];
+ lengths[1] = ff_aac_num_swb_128[i];
+ ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
+ s->psypp = ff_psy_preprocess_init(avctx);
+ s->coder = &ff_aac_coders[0];
+
+ s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+#if !CONFIG_HARDCODED_TABLES
+ for (i = 0; i < 428; i++)
+ ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
+#endif /* CONFIG_HARDCODED_TABLES */
+
+ if (avctx->channels > 5)
+ av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
+ "The output will most likely be an illegal bitstream.\n");
+
return 0;
}
+static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce, short *audio, int channel)
+{
+ int i, j, k;
+ const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ memcpy(s->output, sce->saved, sizeof(float)*1024);
+ if(sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE){
+ memset(s->output, 0, sizeof(s->output[0]) * 448);
+ for(i = 448; i < 576; i++)
+ s->output[i] = sce->saved[i] * pwindow[i - 448];
+ for(i = 576; i < 704; i++)
+ s->output[i] = sce->saved[i];
+ }
+ if(sce->ics.window_sequence[0] != LONG_START_SEQUENCE){
+ j = channel;
+ for (i = 0; i < 1024; i++, j += avctx->channels){
+ s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
+ sce->saved[i] = audio[j] * lwindow[i];
+ }
+ }else{
+ j = channel;
+ for(i = 0; i < 448; i++, j += avctx->channels)
+ s->output[i+1024] = audio[j];
+ for(i = 448; i < 576; i++, j += avctx->channels)
+ s->output[i+1024] = audio[j] * swindow[576 - i - 1];
+ memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
+ j = channel;
+ for(i = 0; i < 1024; i++, j += avctx->channels)
+ sce->saved[i] = audio[j];
+ }
+ ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
+ }else{
+ j = channel;
+ for (k = 0; k < 1024; k += 128) {
+ for(i = 448 + k; i < 448 + k + 256; i++)
+ s->output[i - 448 - k] = (i < 1024)
+ ? sce->saved[i]
+ : audio[channel + (i-1024)*avctx->channels];
+ s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
+ s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
+ ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
+ }
+ j = channel;
+ for(i = 0; i < 1024; i++, j += avctx->channels)
+ sce->saved[i] = audio[j];
+ }
+}
+
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
- int i;
+ int w;
put_bits(&s->pb, 1, 0); // ics_reserved bit
put_bits(&s->pb, 2, info->window_sequence[0]);
@@ -252,27 +276,118 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
put_bits(&s->pb, 1, 0); // no prediction
}else{
put_bits(&s->pb, 4, info->max_sfb);
- for(i = 1; i < info->num_windows; i++)
- put_bits(&s->pb, 1, info->group_len[i]);
+ for(w = 1; w < 8; w++){
+ put_bits(&s->pb, 1, !info->group_len[w]);
+ }
}
}
/**
- * Calculate the number of bits needed to code all coefficient signs in current band.
+ * Encode MS data.
+ * @see 4.6.8.1 "Joint Coding - M/S Stereo"
*/
-static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
- int group_len, int start, int size)
+static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
- int bits = 0;
int i, w;
- for(w = 0; w < group_len; w++){
- for(i = 0; i < size; i++){
- if(sce->icoefs[start + i])
- bits++;
+
+ put_bits(pb, 2, cpe->ms_mode);
+ if(cpe->ms_mode == 1){
+ for(w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]){
+ for(i = 0; i < cpe->ch[0].ics.max_sfb; i++)
+ put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
+ }
+ }
+}
+
+/**
+ * Produce integer coefficients from scalefactors provided by the model.
+ */
+static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
+{
+ int i, w, w2, g, ch;
+ int start, sum, maxsfb, cmaxsfb;
+
+ for(ch = 0; ch < chans; ch++){
+ IndividualChannelStream *ics = &cpe->ch[ch].ics;
+ start = 0;
+ maxsfb = 0;
+ cpe->ch[ch].pulse.num_pulse = 0;
+ for(w = 0; w < ics->num_windows*16; w += 16){
+ for(g = 0; g < ics->num_swb; g++){
+ sum = 0;
+ //apply M/S
+ if(!ch && cpe->ms_mask[w + g]){
+ for(i = 0; i < ics->swb_sizes[g]; i++){
+ cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
+ cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
+ }
+ }
+ start += ics->swb_sizes[g];
+ }
+ for(cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--);
+ maxsfb = FFMAX(maxsfb, cmaxsfb);
+ }
+ ics->max_sfb = maxsfb;
+
+ //adjust zero bands for window groups
+ for(w = 0; w < ics->num_windows; w += ics->group_len[w]){
+ for(g = 0; g < ics->max_sfb; g++){
+ i = 1;
+ for(w2 = w; w2 < w + ics->group_len[w]; w2++){
+ if(!cpe->ch[ch].zeroes[w2*16 + g]){
+ i = 0;
+ break;
+ }
+ }
+ cpe->ch[ch].zeroes[w*16 + g] = i;
+ }
+ }
+ }
+
+ if(chans > 1 && cpe->common_window){
+ IndividualChannelStream *ics0 = &cpe->ch[0].ics;
+ IndividualChannelStream *ics1 = &cpe->ch[1].ics;
+ int msc = 0;
+ ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
+ ics1->max_sfb = ics0->max_sfb;
+ for(w = 0; w < ics0->num_windows*16; w += 16)
+ for(i = 0; i < ics0->max_sfb; i++)
+ if(cpe->ms_mask[w+i]) msc++;
+ if(msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0;
+ else cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
+ }
+}
+
+/**
+ * Encode scalefactor band coding type.
+ */
+static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
+{
+ int w;
+
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
+ }
+}
+
+/**
+ * Encode scalefactors.
+ */
+static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
+{
+ int off = sce->sf_idx[0], diff;
+ int i, w;
+
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
+ for(i = 0; i < sce->ics.max_sfb; i++){
+ if(!sce->zeroes[w*16 + i]){
+ diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
+ if(diff < 0 || diff > 120) av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
+ off = sce->sf_idx[w*16 + i];
+ put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
+ }
}
- start += 128;
}
- return bits;
}
/**
@@ -298,28 +413,44 @@ static void encode_pulses(AACEncContext *s, Pulse *pulse)
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
- int start, i, w, w2, wg;
+ int start, i, w, w2;
- w = 0;
- for(wg = 0; wg < sce->ics.num_window_groups; wg++){
+ for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
start = 0;
for(i = 0; i < sce->ics.max_sfb; i++){
if(sce->zeroes[w*16 + i]){
start += sce->ics.swb_sizes[i];
continue;
}
- for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
- encode_band_coeffs(s, sce, start + w2*128,
- sce->ics.swb_sizes[i],
- sce->band_type[w*16 + i]);
+ for(w2 = w; w2 < w + sce->ics.group_len[w]; w2++){
+ s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
+ sce->ics.swb_sizes[i],
+ sce->sf_idx[w*16 + i],
+ sce->band_type[w*16 + i],
+ s->lambda);
}
start += sce->ics.swb_sizes[i];
}
- w += sce->ics.group_len[wg];
}
}
/**
+ * Encode one channel of audio data.
+ */
+static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
+{
+ put_bits(&s->pb, 8, sce->sf_idx[0]);
+ if(!common_window) put_ics_info(s, &sce->ics);
+ encode_band_info(s, sce);
+ encode_scale_factors(avctx, s, sce);
+ encode_pulses(s, &sce->pulse);
+ put_bits(&s->pb, 1, 0); //tns
+ put_bits(&s->pb, 1, 0); //ssr
+ encode_spectral_coeffs(s, sce);
+ return 0;
+}
+
+/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
@@ -339,13 +470,130 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const ch
put_bits(&s->pb, 12 - padbits, 0);
}
+static int aac_encode_frame(AVCodecContext *avctx,
+ uint8_t *frame, int buf_size, void *data)
+{
+ AACEncContext *s = avctx->priv_data;
+ int16_t *samples = s->samples, *samples2, *la;
+ ChannelElement *cpe;
+ int i, j, chans, tag, start_ch;
+ const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
+ int chan_el_counter[4];
+
+ if(s->last_frame)
+ return 0;
+ if(data){
+ if(!s->psypp){
+ memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
+ }else{
+ start_ch = 0;
+ samples2 = s->samples + 1024 * avctx->channels;
+ for(i = 0; i < chan_map[0]; i++){
+ tag = chan_map[i+1];
+ chans = tag == TYPE_CPE ? 2 : 1;
+ ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch, chans);
+ start_ch += chans;
+ }
+ }
+ }
+ if(!avctx->frame_number){
+ memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
+ return 0;
+ }
+
+ init_put_bits(&s->pb, frame, buf_size*8);
+ if((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)){
+ put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
+ }
+ start_ch = 0;
+ memset(chan_el_counter, 0, sizeof(chan_el_counter));
+ for(i = 0; i < chan_map[0]; i++){
+ FFPsyWindowInfo wi[2];
+ tag = chan_map[i+1];
+ chans = tag == TYPE_CPE ? 2 : 1;
+ cpe = &s->cpe[i];
+ samples2 = samples + start_ch;
+ la = samples2 + 1024 * avctx->channels + start_ch;
+ if(!data) la = NULL;
+ for(j = 0; j < chans; j++){
+ IndividualChannelStream *ics = &cpe->ch[j].ics;
+ int k;
+ wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = wi[j].window_type[0];
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = wi[j].window_shape;
+ ics->num_windows = wi[j].num_windows;
+ ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
+ ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
+ for(k = 0; k < ics->num_windows; k++)
+ ics->group_len[k] = wi[j].grouping[k];
+
+ s->cur_channel = start_ch + j;
+ apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
+ s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
+ }
+ cpe->common_window = 0;
+ if(chans > 1
+ && wi[0].window_type[0] == wi[1].window_type[0]
+ && wi[0].window_shape == wi[1].window_shape){
+
+ cpe->common_window = 1;
+ for(j = 0; j < wi[0].num_windows; j++){
+ if(wi[0].grouping[j] != wi[1].grouping[j]){
+ cpe->common_window = 0;
+ break;
+ }
+ }
+ }
+ if(cpe->common_window && s->coder->search_for_ms)
+ s->coder->search_for_ms(s, cpe, s->lambda);
+ adjust_frame_information(s, cpe, chans);
+ put_bits(&s->pb, 3, tag);
+ put_bits(&s->pb, 4, chan_el_counter[tag]++);
+ if(chans == 2){
+ put_bits(&s->pb, 1, cpe->common_window);
+ if(cpe->common_window){
+ put_ics_info(s, &cpe->ch[0].ics);
+ encode_ms_info(&s->pb, cpe);
+ }
+ }
+ for(j = 0; j < chans; j++){
+ s->cur_channel = start_ch + j;
+ ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
+ encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
+ }
+ start_ch += chans;
+ }
+
+ put_bits(&s->pb, 3, TYPE_END);
+ flush_put_bits(&s->pb);
+ avctx->frame_bits = put_bits_count(&s->pb);
+
+ // rate control stuff
+ if(!(avctx->flags & CODEC_FLAG_QSCALE)){
+ float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
+ s->lambda *= ratio;
+ }
+
+ if (avctx->frame_bits > 6144*avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n", avctx->frame_bits, 6144*avctx->channels);
+ }
+
+ if(!data)
+ s->last_frame = 1;
+ memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
+ return put_bits_count(&s->pb)>>3;
+}
+
static av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
- ff_aac_psy_end(&s->psy);
+ ff_psy_end(&s->psy);
+ ff_psy_preprocess_end(s->psypp);
av_freep(&s->samples);
av_freep(&s->cpe);
return 0;
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
new file mode 100644
index 0000000000..458d3d5d99
--- /dev/null
+++ b/libavcodec/aacenc.h
@@ -0,0 +1,71 @@
+/*
+ * AAC encoder
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * 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
+ */
+
+#ifndef AVCODEC_AACENC_H
+#define AVCODEC_AACENC_H
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "dsputil.h"
+
+#include "aac.h"
+
+#include "psymodel.h"
+
+struct AACEncContext;
+
+typedef struct AACCoefficientsEncoder{
+ void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
+ SingleChannelElement *sce, const float lambda);
+ void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
+ int win, int group_len, const float lambda);
+ void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+ int scale_idx, int cb, const float lambda);
+ void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
+}AACCoefficientsEncoder;
+
+extern AACCoefficientsEncoder ff_aac_coders[];
+
+/**
+ * AAC encoder context
+ */
+typedef struct AACEncContext {
+ PutBitContext pb;
+ MDCTContext mdct1024; ///< long (1024 samples) frame transform context
+ MDCTContext mdct128; ///< short (128 samples) frame transform context
+ DSPContext dsp;
+ DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
+ int16_t* samples; ///< saved preprocessed input
+
+ int samplerate_index; ///< MPEG-4 samplerate index
+
+ ChannelElement *cpe; ///< channel elements
+ FFPsyContext psy;
+ struct FFPsyPreprocessContext* psypp;
+ AACCoefficientsEncoder *coder;
+ int cur_channel;
+ int last_frame;
+ float lambda;
+ DECLARE_ALIGNED_16(int, qcoefs[96][2]); ///< quantized coefficients
+ DECLARE_ALIGNED_16(float, scoefs[1024]); ///< scaled coefficients
+} AACEncContext;
+
+#endif /* AVCODEC_AACENC_H */
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 45fcad46be..3880266784 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -25,54 +25,25 @@
*/
#include "avcodec.h"
-#include "aacpsy.h"
#include "aactab.h"
+#include "psymodel.h"
/***********************************
* TODOs:
- * General:
- * better audio preprocessing (add DC highpass filter?)
- * more psy models
- * maybe improve coefficient quantization function in some way
- *
- * 3GPP-based psy model:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static av_always_inline int quant(float coef, const float Q)
-{
- return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
-}
-
-static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
-{
- int i;
- int q;
- float coef, unquant, sum = 0.0f;
- const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
- const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
- for(i = 0; i < size; i++){
- coef = fabs(c[i]);
- q = quant(c[i], Q);
- unquant = (q * cbrt(q)) * IQ;
- sum += (coef - unquant) * (coef - unquant);
- }
- return sum;
-}
-
-/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
+
+#define PSY_3GPP_RPEMIN 0.01f
+#define PSY_3GPP_RPELEV 2.0f
/**
* @}
*/
@@ -83,9 +54,25 @@ static inline float get_approximate_quant_error(float *c, int size, int scale_id
typedef struct Psy3gppBand{
float energy; ///< band energy
float ffac; ///< form factor
+ float thr; ///< energy threshold
+ float min_snr; ///< minimal SNR
+ float thr_quiet; ///< threshold in quiet
}Psy3gppBand;
/**
+ * single/pair channel context for psychoacoustic model
+ */
+typedef struct Psy3gppChannel{
+ Psy3gppBand band[128]; ///< bands information
+ Psy3gppBand prev_band[128]; ///< bands information from the previous frame
+
+ float win_energy; ///< sliding average of channel energy
+ float iir_state[2]; ///< hi-pass IIR filter state
+ uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
+ enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
+}Psy3gppChannel;
+
+/**
* psychoacoustic model frame type-dependent coefficients
*/
typedef struct Psy3gppCoeffs{
@@ -96,9 +83,240 @@ typedef struct Psy3gppCoeffs{
}Psy3gppCoeffs;
/**
+ * 3GPP TS26.403-inspired psychoacoustic model specific data
+ */
+typedef struct Psy3gppContext{
+ Psy3gppCoeffs psy_coef[2];
+ Psy3gppChannel *ch;
+}Psy3gppContext;
+
+/**
* Calculate Bark value for given line.
*/
-static inline float calc_bark(float f)
+static av_cold float calc_bark(float f)
{
return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}
+
+#define ATH_ADD 4
+/**
+ * Calculate ATH value for given frequency.
+ * Borrowed from Lame.
+ */
+static av_cold float ath(float f, float add)
+{
+ f /= 1000.0f;
+ return 3.64 * pow(f, -0.8)
+ - 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ + 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ + (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
+}
+
+static av_cold int psy_3gpp_init(FFPsyContext *ctx){
+ Psy3gppContext *pctx;
+ float barks[1024];
+ int i, j, g, start;
+ float prev, minscale, minath;
+
+ ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
+ pctx = (Psy3gppContext*) ctx->model_priv_data;
+
+ for(i = 0; i < 1024; i++)
+ barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
+ minath = ath(3410, ATH_ADD);
+ for(j = 0; j < 2; j++){
+ Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
+ i = 0;
+ prev = 0.0;
+ for(g = 0; g < ctx->num_bands[j]; g++){
+ i += ctx->bands[j][g];
+ coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
+ prev = barks[i - 1];
+ }
+ for(g = 0; g < ctx->num_bands[j] - 1; g++){
+ coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
+ coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
+ }
+ start = 0;
+ for(g = 0; g < ctx->num_bands[j]; g++){
+ minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
+ for(i = 1; i < ctx->bands[j][g]; i++){
+ minscale = fminf(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
+ }
+ coeffs->ath[g] = minscale - minath;
+ start += ctx->bands[j][g];
+ }
+ }
+
+ pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
+ return 0;
+}
+
+/**
+ * IIR filter used in block switching decision
+ */
+static float iir_filter(int in, float state[2])
+{
+ float ret;
+
+ ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
+ state[0] = in;
+ state[1] = ret;
+ return ret;
+}
+
+/**
+ * window grouping information stored as bits (0 - new group, 1 - group continues)
+ */
+static const uint8_t window_grouping[9] = {
+ 0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
+};
+
+/**
+ * Tell encoder which window types to use.
+ * @see 3GPP TS26.403 5.4.1 "Blockswitching"
+ */
+static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
+ const int16_t *audio, const int16_t *la,
+ int channel, int prev_type)
+{
+ int i, j;
+ int br = ctx->avctx->bit_rate / ctx->avctx->channels;
+ int attack_ratio = br <= 16000 ? 18 : 10;
+ Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
+ Psy3gppChannel *pch = &pctx->ch[channel];
+ uint8_t grouping = 0;
+ FFPsyWindowInfo wi;
+
+ memset(&wi, 0, sizeof(wi));
+ if(la){
+ float s[8], v;
+ int switch_to_eight = 0;
+ float sum = 0.0, sum2 = 0.0;
+ int attack_n = 0;
+ for(i = 0; i < 8; i++){
+ for(j = 0; j < 128; j++){
+ v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
+ sum += v*v;
+ }
+ s[i] = sum;
+ sum2 += sum;
+ }
+ for(i = 0; i < 8; i++){
+ if(s[i] > pch->win_energy * attack_ratio){
+ attack_n = i + 1;
+ switch_to_eight = 1;
+ break;
+ }
+ }
+ pch->win_energy = pch->win_energy*7/8 + sum2/64;
+
+ wi.window_type[1] = prev_type;
+ switch(prev_type){
+ case ONLY_LONG_SEQUENCE:
+ wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
+ break;
+ case LONG_START_SEQUENCE:
+ wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
+ grouping = pch->next_grouping;
+ break;
+ case LONG_STOP_SEQUENCE:
+ wi.window_type[0] = ONLY_LONG_SEQUENCE;
+ break;
+ case EIGHT_SHORT_SEQUENCE:
+ wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
+ grouping = switch_to_eight ? pch->next_grouping : 0;
+ break;
+ }
+ pch->next_grouping = window_grouping[attack_n];
+ }else{
+ for(i = 0; i < 3; i++)
+ wi.window_type[i] = prev_type;
+ grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
+ }
+
+ wi.window_shape = 1;
+ if(wi.window_type[0] != EIGHT_SHORT_SEQUENCE){
+ wi.num_windows = 1;
+ wi.grouping[0] = 1;
+ }else{
+ int lastgrp = 0;
+ wi.num_windows = 8;
+ for(i = 0; i < 8; i++){
+ if(!((grouping >> i) & 1))
+ lastgrp = i;
+ wi.grouping[lastgrp]++;
+ }
+ }
+
+ return wi;
+}
+
+/**
+ * Calculate band thresholds as suggested in 3GPP TS26.403
+ */
+static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
+ FFPsyWindowInfo *wi)
+{
+ Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
+ Psy3gppChannel *pch = &pctx->ch[channel];
+ int start = 0;
+ int i, w, g;
+ const int num_bands = ctx->num_bands[wi->num_windows == 8];
+ const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
+ Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
+
+ //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+ for(w = 0; w < wi->num_windows*16; w += 16){
+ for(g = 0; g < num_bands; g++){
+ Psy3gppBand *band = &pch->band[w+g];
+ band->energy = 0.0f;
+ for(i = 0; i < band_sizes[g]; i++)
+ band->energy += coefs[start+i] * coefs[start+i];
+ band->energy *= 1.0f / (512*512);
+ band->thr = band->energy * 0.001258925f;
+ start += band_sizes[g];
+
+ ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
+ }
+ }
+ //modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
+ for(w = 0; w < wi->num_windows*16; w += 16){
+ Psy3gppBand *band = &pch->band[w];
+ for(g = 1; g < num_bands; g++){
+ band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
+ }
+ for(g = num_bands - 2; g >= 0; g--){
+ band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g]);
+ }
+ for(g = 0; g < num_bands; g++){
+ band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
+ if(wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE){
+ band[g].thr_quiet = fmaxf(PSY_3GPP_RPEMIN*band[g].thr_quiet,
+ fminf(band[g].thr_quiet,
+ PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+ }
+ band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
+
+ ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
+ }
+ }
+ memcpy(pch->prev_band, pch->band, sizeof(pch->band));
+}
+
+static av_cold void psy_3gpp_end(FFPsyContext *apc)
+{
+ Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
+ av_freep(&pctx->ch);
+ av_freep(&apc->model_priv_data);
+}
+
+
+const FFPsyModel ff_aac_psy_model =
+{
+ .name = "3GPP TS 26.403-inspired model",
+ .init = psy_3gpp_init,
+ .window = psy_3gpp_window,
+ .analyze = psy_3gpp_analyze,
+ .end = psy_3gpp_end,
+};
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 6a171147f8..c289523a8f 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -195,7 +195,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (ZMBV, zmbv);
/* audio codecs */
- REGISTER_DECODER (AAC, aac);
+ REGISTER_ENCDEC (AAC, aac);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (APE, ape);
diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
new file mode 100644
index 0000000000..623254e531
--- /dev/null
+++ b/libavcodec/psymodel.c
@@ -0,0 +1,130 @@
+/*
+ * audio encoder psychoacoustic model
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * 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
+ */
+
+#include "avcodec.h"
+#include "psymodel.h"
+#include "iirfilter.h"
+
+extern const FFPsyModel ff_aac_psy_model;
+
+av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
+ int num_lens,
+ const uint8_t **bands, const int* num_bands)
+{
+ ctx->avctx = avctx;
+ ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
+ ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
+ ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
+ memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
+ memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
+ switch(ctx->avctx->codec_id){
+ case CODEC_ID_AAC:
+ ctx->model = &ff_aac_psy_model;
+ break;
+ }
+ if(ctx->model->init)
+ return ctx->model->init(ctx);
+ return 0;
+}
+
+FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
+ const int16_t *audio, const int16_t *la,
+ int channel, int prev_type)
+{
+ return ctx->model->window(ctx, audio, la, channel, prev_type);
+}
+
+void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
+ const float *coeffs, FFPsyWindowInfo *wi)
+{
+ ctx->model->analyze(ctx, channel, coeffs, wi);
+}
+
+av_cold void ff_psy_end(FFPsyContext *ctx)
+{
+ if(ctx->model->end)
+ ctx->model->end(ctx);
+ av_freep(&ctx->bands);
+ av_freep(&ctx->num_bands);
+ av_freep(&ctx->psy_bands);
+}
+
+typedef struct FFPsyPreprocessContext{
+ AVCodecContext *avctx;
+ float stereo_att;
+ struct FFIIRFilterCoeffs *fcoeffs;
+ struct FFIIRFilterState **fstate;
+}FFPsyPreprocessContext;
+
+#define FILT_ORDER 4
+
+av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
+{
+ FFPsyPreprocessContext *ctx;
+ int i;
+ float cutoff_coeff;
+ ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
+ ctx->avctx = avctx;
+
+ if(avctx->flags & CODEC_FLAG_QSCALE)
+ cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
+ else
+ cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
+
+ ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
+ FILT_ORDER, cutoff_coeff, 0.0, 0.0);
+ if(ctx->fcoeffs){
+ ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
+ for(i = 0; i < avctx->channels; i++)
+ ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
+ }
+ return ctx;
+}
+
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
+ const int16_t *audio, int16_t *dest,
+ int tag, int channels)
+{
+ int ch, i;
+ if(ctx->fstate){
+ for(ch = 0; ch < channels; ch++){
+ ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
+ audio + ch, ctx->avctx->channels,
+ dest + ch, ctx->avctx->channels);
+ }
+ }else{
+ for(ch = 0; ch < channels; ch++){
+ for(i = 0; i < ctx->avctx->frame_size; i++)
+ dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+ }
+ }
+}
+
+av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
+{
+ int i;
+ ff_iir_filter_free_coeffs(ctx->fcoeffs);
+ if (ctx->fstate)
+ for (i = 0; i < ctx->avctx->channels; i++)
+ ff_iir_filter_free_state(ctx->fstate[i]);
+ av_freep(&ctx->fstate);
+}
+
diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
new file mode 100644
index 0000000000..5bcf556f0b
--- /dev/null
+++ b/libavcodec/psymodel.h
@@ -0,0 +1,158 @@
+/*
+ * audio encoder psychoacoustic model
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * 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
+ */
+
+#ifndef AVCODEC_PSYMODEL_H
+#define AVCODEC_PSYMODEL_H
+
+#include "avcodec.h"
+
+/** maximum possible number of bands */
+#define PSY_MAX_BANDS 128
+
+/**
+ * single band psychoacoustic information
+ */
+typedef struct FFPsyBand{
+ int bits;
+ float energy;
+ float threshold;
+ float distortion;
+ float perceptual_weight;
+}FFPsyBand;
+
+/**
+ * windowing related information
+ */
+typedef struct FFPsyWindowInfo{
+ int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
+ int window_shape; ///< window shape (sine/KBD/whatever)
+ int num_windows; ///< number of windows in a frame
+ int grouping[8]; ///< window grouping (for e.g. AAC)
+ int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
+}FFPsyWindowInfo;
+
+/**
+ * context used by psychoacoustic model
+ */
+typedef struct FFPsyContext{
+ AVCodecContext *avctx; ///< encoder context
+ const struct FFPsyModel *model; ///< encoder-specific model functions
+
+ FFPsyBand *psy_bands; ///< frame bands information
+
+ uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
+ int *num_bands; ///< number of scalefactor bands for possible frame sizes
+ int num_lens; ///< number of scalefactor band sets
+
+ void* model_priv_data; ///< psychoacoustic model implementation private data
+}FFPsyContext;
+
+/**
+ * codec-specific psychoacoustic model implementation
+ */
+typedef struct FFPsyModel {
+ const char *name;
+ int (*init) (FFPsyContext *apc);
+ FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
+ void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi);
+ void (*end) (FFPsyContext *apc);
+}FFPsyModel;
+
+/**
+ * Initialize psychoacoustic model.
+ *
+ * @param ctx model context
+ * @param avctx codec context
+ * @param num_lens number of possible frame lengths
+ * @param bands scalefactor band lengths for all frame lengths
+ * @param num_bands number of scalefactor bands for all frame lengths
+ *
+ * @return zero if successful, a negative value if not
+ */
+av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
+ int num_lens,
+ const uint8_t **bands, const int* num_bands);
+
+/**
+ * Suggest window sequence for channel.
+ *
+ * @param ctx model context
+ * @param audio samples for the current frame
+ * @param la lookahead samples (NULL when unavailable)
+ * @param channel number of channel element to analyze
+ * @param prev_type previous window type
+ *
+ * @return suggested window information in a structure
+ */
+FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
+ const int16_t *audio, const int16_t *la,
+ int channel, int prev_type);
+
+
+/**
+ * Perform psychoacoustic analysis and set band info (threshold, energy).
+ *
+ * @param ctx model context
+ * @param channel audio channel number
+ * @param coeffs pointer to the transformed coefficients
+ * @param wi window information
+ */
+void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs,
+ FFPsyWindowInfo *wi);
+
+/**
+ * Cleanup model context at the end.
+ *
+ * @param ctx model context
+ */
+av_cold void ff_psy_end(FFPsyContext *ctx);
+
+
+/**************************************************************************
+ * Audio preprocessing stuff. *
+ * This should be moved into some audio filter eventually. *
+ **************************************************************************/
+struct FFPsyPreprocessContext;
+
+/**
+ * psychoacoustic model audio preprocessing initialization
+ */
+av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);
+
+/**
+ * Preprocess several channel in audio frame in order to compress it better.
+ *
+ * @param ctx preprocessing context
+ * @param audio samples to preprocess
+ * @param dest place to put filtered samples
+ * @param tag channel number
+ * @param channels number of channel to preprocess (some additional work may be done on stereo pair)
+ */
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
+ const int16_t *audio, int16_t *dest,
+ int tag, int channels);
+
+/**
+ * Cleanup audio preprocessing module.
+ */
+av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
+
+#endif /* AVCODEC_PSYMODEL_H */
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