aaccoder: Implement Perceptual Noise Substitution for AAC

This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
made after some problems were noted out. Any changes made since the previous revisions have been indicated.

In order to extend the encoder to use an additional codebook, the array holding each codebook has been
modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
extended to allow for intensity stereo encoding, which uses additional codebooks.

The 12th entry in the codebook function array points to a function which stops the execution of the program
by calling an assert with an always 'false' argument. It was pointed out in an email discussion with
Claudio Freire that having a 'NULL' entry can result in unexpected behaviour and could be used as
a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.

Another change from version 1 of the patch is the addition of an argument to the encoder, '-aac_pns' to
enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
The switch will be removed in the future, when the algorithm to select noise bands has been improved.
The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.

Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
a value of '2.2'. Claudio Freire reported the problem's source could be in the range of the scalefactor
indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
value. This has been implemented and all the files which used to trigger the asserion now encode without error.

The third revision of the problem also removes unneded variabes and comparisons. All of them were
redundant and were of little use for when the PNS implementation would be extended.

The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
variable due to the fact that rounding errors can prove to be a problem at low frequencies.
Considerations were taken whether the entire expression could be evaluated inside the expression
, but in the end it was decided that it would be for the best if just the type of the variable were
to change. Claudio Freire reported the two problems. There is no change of functionality
(except for low sampling frequencies) so the spectral demonstrations at the end of this commit's message were not updated.

Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
ranges of the scalefactor indices.

Some spectral comparisons: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
The constant is the value which multiplies the threshold when it gets compared to the energy, larger
values means more noise will be substituded by PNS values. Example when const = 2.2:
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.pngReviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/aaccoder.c

@@ -40,6 +40,12 @@
 #include "aacenc.h"
 #include "aactab.h"
 
+/** Frequency in Hz for lower limit of noise substitution **/
+#define NOISE_LOW_LIMIT 4000
+
+/** Total number of usable codebooks **/
+#define CB_TOT 13
+
 /** 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,
@@ -57,6 +63,10 @@ static const uint8_t * const run_value_bits[2] = {
     run_value_bits_long, run_value_bits_short
 };
 
+/** Map to convert values from BandCodingPath index to a codebook index **/
+static const uint8_t aac_cb_out_map[CB_TOT]  = {0,1,2,3,4,5,6,7,8,9,10,11,13};
+/** Inverse map to convert from codebooks to BandCodingPath indices **/
+static const uint8_t aac_cb_in_map[CB_TOT+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12};
 
 /**
  * Quantize one coefficient.
@@ -108,7 +118,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
                                 const float *scaled, int size, int scale_idx,
                                 int cb, const float lambda, const float uplim,
                                 int *bits, int BT_ZERO, int BT_UNSIGNED,
-                                int BT_PAIR, int BT_ESC)
+                                int BT_PAIR, int BT_ESC, int BT_NOISE)
 {
     const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
     const float Q   = ff_aac_pow2sf_tab [q_idx];
@@ -119,8 +129,6 @@ static av_always_inline float quantize_and_encode_band_cost_template(
     float cost = 0;
     const int dim = BT_PAIR ? 2 : 4;
     int resbits = 0;
-    const int range  = aac_cb_range[cb];
-    const int maxval = aac_cb_maxval[cb];
     int off;
 
     if (BT_ZERO) {
@@ -130,15 +138,22 @@ static av_always_inline float quantize_and_encode_band_cost_template(
             *bits = 0;
         return cost * lambda;
     }
+    if (BT_NOISE) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        return cost * lambda;
+    }
     if (!scaled) {
         abs_pow34_v(s->scoefs, in, size);
         scaled = s->scoefs;
     }
-    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, maxval);
+    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb]);
     if (BT_UNSIGNED) {
         off = 0;
     } else {
-        off = maxval;
+        off = aac_cb_maxval[cb];
     }
     for (i = 0; i < size; i += dim) {
         const float *vec;
@@ -147,7 +162,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
         int curbits;
         float rd = 0.0f;
         for (j = 0; j < dim; j++) {
-            curidx *= range;
+            curidx *= aac_cb_range[cb];
             curidx += quants[j] + off;
         }
         curbits =  ff_aac_spectral_bits[cb-1][curidx];
@@ -207,8 +222,17 @@ static av_always_inline float quantize_and_encode_band_cost_template(
     return cost;
 }
 
-#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC) \
-static float quantize_and_encode_band_cost_ ## NAME(                                        \
+static float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
+                                                const float *in, const float *scaled,
+                                                int size, int scale_idx, int cb,
+                                                const float lambda, const float uplim,
+                                                int *bits) {
+    av_assert0(0);
+    return 0.0f;
+}
+
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE) \
+static float quantize_and_encode_band_cost_ ## NAME(                                    \
                                 struct AACEncContext *s,                                \
                                 PutBitContext *pb, const float *in,                     \
                                 const float *scaled, int size, int scale_idx,           \
@@ -217,15 +241,16 @@ static float quantize_and_encode_band_cost_ ## NAME(
     return quantize_and_encode_band_cost_template(                                      \
                                 s, pb, in, scaled, size, scale_idx,                     \
                                 BT_ESC ? ESC_BT : cb, lambda, uplim, bits,              \
-                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC);                 \
+                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE);       \
 }
 
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1)
 
 static float (*const quantize_and_encode_band_cost_arr[])(
                                 struct AACEncContext *s,
@@ -245,6 +270,8 @@ static float (*const quantize_and_encode_band_cost_arr[])(
     quantize_and_encode_band_cost_UPAIR,
     quantize_and_encode_band_cost_UPAIR,
     quantize_and_encode_band_cost_ESC,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
 };
 
 #define quantize_and_encode_band_cost(                                  \
@@ -312,7 +339,7 @@ typedef struct BandCodingPath {
 static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                      int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][12];
+    BandCodingPath path[120][CB_TOT];
     int w, swb, cb, start, size;
     int i, j;
     const int max_sfb  = sce->ics.max_sfb;
@@ -325,7 +352,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
     start = win*128;
-    for (cb = 0; cb < 12; cb++) {
+    for (cb = 0; cb < CB_TOT; cb++) {
         path[0][cb].cost     = 0.0f;
         path[0][cb].prev_idx = -1;
         path[0][cb].run      = 0;
@@ -333,7 +360,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     for (swb = 0; swb < max_sfb; swb++) {
         size = sce->ics.swb_sizes[swb];
         if (sce->zeroes[win*16 + swb]) {
-            for (cb = 0; cb < 12; cb++) {
+            for (cb = 0; cb < CB_TOT; 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;
@@ -343,14 +370,14 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
             int mincb = next_mincb;
             next_minrd = INFINITY;
             next_mincb = 0;
-            for (cb = 0; cb < 12; cb++) {
+            for (cb = 0; cb < CB_TOT; cb++) {
                 float cost_stay_here, cost_get_here;
                 float rd = 0.0f;
                 for (w = 0; w < group_len; w++) {
                     FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_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,
+                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
                                              lambda / band->threshold, INFINITY, NULL);
                 }
                 cost_stay_here = path[swb][cb].cost + rd;
@@ -379,7 +406,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     //convert resulting path from backward-linked list
     stack_len = 0;
     idx       = 0;
-    for (cb = 1; cb < 12; cb++)
+    for (cb = 1; cb < CB_TOT; cb++)
         if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
             idx = cb;
     ppos = max_sfb;
@@ -394,12 +421,13 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     //perform actual band info encoding
     start = 0;
     for (i = stack_len - 1; i >= 0; i--) {
-        put_bits(&s->pb, 4, stackcb[i]);
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
         count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
+        memset(sce->zeroes + win*16 + start, !cb, count);
         //XXX: memset when band_type is also uint8_t
         for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] =  stackcb[i];
+            sce->band_type[win*16 + start] = cb;
             start++;
         }
         while (count >= run_esc) {
@@ -413,7 +441,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                                   int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][12];
+    BandCodingPath path[120][CB_TOT];
     int w, swb, cb, start, size;
     int i, j;
     const int max_sfb  = sce->ics.max_sfb;
@@ -426,7 +454,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
     start = win*128;
-    for (cb = 0; cb < 12; cb++) {
+    for (cb = 0; cb < CB_TOT; cb++) {
         path[0][cb].cost     = run_bits+4;
         path[0][cb].prev_idx = -1;
         path[0][cb].run      = 0;
@@ -450,7 +478,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             }
             next_minbits = path[swb+1][0].cost;
             next_mincb = 0;
-            for (cb = 1; cb < 12; cb++) {
+            for (cb = 1; cb < CB_TOT; cb++) {
                 path[swb+1][cb].cost = 61450;
                 path[swb+1][cb].prev_idx = -1;
                 path[swb+1][cb].run = 0;
@@ -459,6 +487,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             float minbits = next_minbits;
             int mincb = next_mincb;
             int startcb = sce->band_type[win*16+swb];
+            startcb = aac_cb_in_map[startcb];
             next_minbits = INFINITY;
             next_mincb = 0;
             for (cb = 0; cb < startcb; cb++) {
@@ -466,13 +495,20 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][cb].prev_idx = -1;
                 path[swb+1][cb].run = 0;
             }
-            for (cb = startcb; cb < 12; cb++) {
+            for (cb = startcb; cb < CB_TOT; cb++) {
                 float cost_stay_here, cost_get_here;
                 float bits = 0.0f;
+                if (cb == 12 && sce->band_type[win*16+swb] != NOISE_BT) {
+                    path[swb+1][cb].cost = 61450;
+                    path[swb+1][cb].prev_idx = -1;
+                    path[swb+1][cb].run = 0;
+                    continue;
+                }
                 for (w = 0; w < group_len; w++) {
                     bits += quantize_band_cost(s, sce->coeffs + start + w*128,
                                                s->scoefs + start + w*128, size,
-                                               sce->sf_idx[(win+w)*16+swb], cb,
+                                               sce->sf_idx[(win+w)*16+swb],
+                                               aac_cb_out_map[cb],
                                                0, INFINITY, NULL);
                 }
                 cost_stay_here = path[swb][cb].cost + bits;
@@ -501,7 +537,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     //convert resulting path from backward-linked list
     stack_len = 0;
     idx       = 0;
-    for (cb = 1; cb < 12; cb++)
+    for (cb = 1; cb < CB_TOT; cb++)
         if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
             idx = cb;
     ppos = max_sfb;
@@ -517,12 +553,13 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     //perform actual band info encoding
     start = 0;
     for (i = stack_len - 1; i >= 0; i--) {
-        put_bits(&s->pb, 4, stackcb[i]);
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
         count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
+        memset(sce->zeroes + win*16 + start, !cb, count);
         //XXX: memset when band_type is also uint8_t
         for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] =  stackcb[i];
+            sce->band_type[win*16 + start] = cb;
             start++;
         }
         while (count >= run_esc) {
@@ -711,9 +748,11 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 {
     int start = 0, i, w, w2, g;
     int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
-    float dists[128] = { 0 }, uplims[128];
+    const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
+    float dists[128] = { 0 }, uplims[128] = { 0 };
     float maxvals[128];
-    int fflag, minscaler;
+    int noise_sf[128] = { 0 };
+    int fflag, minscaler, minscaler_n;
     int its  = 0;
     int allz = 0;
     float minthr = INFINITY;
@@ -724,12 +763,14 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
     //XXX: some heuristic to determine initial quantizers will reduce search time
     //determine zero bands and upper limits
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
         for (g = 0;  g < sce->ics.num_swb; g++) {
             int nz = 0;
-            float uplim = 0.0f;
+            float uplim = 0.0f, energy = 0.0f;
             for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                 FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
                 uplim += band->threshold;
+                energy += band->energy;
                 if (band->energy <= band->threshold || band->threshold == 0.0f) {
                     sce->zeroes[(w+w2)*16+g] = 1;
                     continue;
@@ -737,10 +778,18 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                 nz = 1;
             }
             uplims[w*16+g] = uplim *512;
+            if (s->options.pns && start*freq_mult > NOISE_LOW_LIMIT && energy < uplim * 1.2f) {
+                noise_sf[w*16+g] = av_clip(4+FFMIN(log2f(energy)*2,255), -100, 155);
+                sce->band_type[w*16+g] = NOISE_BT;
+                nz= 1;
+            } else { /** Band type will be determined by the twoloop algorithm */
+                sce->band_type[w*16+g] = 0;
+            }
             sce->zeroes[w*16+g] = !nz;
             if (nz)
                 minthr = FFMIN(minthr, uplim);
             allz |= nz;
+            start += sce->ics.swb_sizes[g];
         }
     }
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
@@ -771,6 +820,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
     do {
         int tbits, qstep;
         minscaler = sce->sf_idx[0];
+        minscaler_n = sce->sf_idx[0];
         //inner loop - quantize spectrum to fit into given number of bits
         qstep = its ? 1 : 32;
         do {
@@ -785,7 +835,11 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                     int cb;
                     float dist = 0.0f;
 
-                    if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+                    if (sce->band_type[w*16+g] == NOISE_BT) {
+                        minscaler_n = FFMIN(minscaler_n, noise_sf[w*16+g]);
+                        start += sce->ics.swb_sizes[g];
+                        continue;
+                    } else if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
                         start += sce->ics.swb_sizes[g];
                         continue;
                     }
@@ -828,9 +882,17 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 
         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])
+            for (g = 0; g < sce->ics.num_swb; g++)
+                if (sce->band_type[w*16+g] == NOISE_BT)
+                    sce->sf_idx[w*16+g] = av_clip(noise_sf[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
+
         for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
             for (g = 0; g < sce->ics.num_swb; g++) {
                 int prevsc = sce->sf_idx[w*16+g];
+                if (sce->band_type[w*16+g] == NOISE_BT)
+                    continue;
                 if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
                     if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
                         sce->sf_idx[w*16+g]--;

libavcodec/aacenc.c

@@ -842,6 +842,9 @@ static const AVOption aacenc_options[] = {
         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
         {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
+    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
+        {"disable",  "Disable PNS", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
+        {"enable",   "Enable PNS (Proof of concept)",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
     {NULL}
 };
 

libavcodec/aacenc.h

@@ -42,6 +42,7 @@ typedef enum AACCoder {
 typedef struct AACEncOptions {
     int stereo_mode;
     int aac_coder;
+    int pns;
 } AACEncOptions;
 
 struct AACEncContext;