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-rw-r--r--libavcodec/aac.h4
-rw-r--r--libavcodec/aaccoder.c108
-rw-r--r--libavcodec/aacenc.c38
-rw-r--r--libavcodec/aacenc.h2
-rw-r--r--libavcodec/aacpsy.c30
-rw-r--r--libavcodec/psymodel.h1
-rw-r--r--tests/fate/aac.mak4
7 files changed, 145 insertions, 42 deletions
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index d62455d1ec..3e3e479986 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -50,6 +50,8 @@
#define TNS_MAX_ORDER 20
#define MAX_LTP_LONG_SFB 40
+#define CLIP_AVOIDANCE_FACTOR 0.95f
+
enum RawDataBlockType {
TYPE_SCE,
TYPE_CPE,
@@ -180,6 +182,8 @@ typedef struct IndividualChannelStream {
int predictor_initialized;
int predictor_reset_group;
uint8_t prediction_used[41];
+ uint8_t window_clipping[8]; ///< set if a certain window is near clipping
+ float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
} IndividualChannelStream;
/**
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
index 17b14d6381..eb583426ed 100644
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -79,6 +79,9 @@ static const uint8_t * const run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
+#define ROUND_STANDARD 0.4054f
+#define ROUND_TO_ZERO 0.1054f
+
/** Map to convert values from BandCodingPath index to a codebook index **/
static const uint8_t aac_cb_out_map[CB_TOT_ALL] = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
/** Inverse map to convert from codebooks to BandCodingPath indices **/
@@ -89,20 +92,20 @@ static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,
* @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)
+static av_always_inline int quant(float coef, const float Q, const float rounding)
{
float a = coef * Q;
- return sqrtf(a * sqrtf(a)) + 0.4054;
+ return sqrtf(a * sqrtf(a)) + rounding;
}
static void quantize_bands(int *out, const float *in, const float *scaled,
- int size, float Q34, int is_signed, int maxval)
+ int size, float Q34, int is_signed, int maxval, const float rounding)
{
int i;
double qc;
for (i = 0; i < size; i++) {
qc = scaled[i] * Q34;
- out[i] = (int)FFMIN(qc + 0.4054, (double)maxval);
+ out[i] = (int)FFMIN(qc + rounding, (double)maxval);
if (is_signed && in[i] < 0.0f) {
out[i] = -out[i];
}
@@ -134,7 +137,8 @@ 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_NOISE, int BT_STEREO)
+ int BT_PAIR, int BT_ESC, int BT_NOISE, int BT_STEREO,
+ const float ROUNDING)
{
const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
const float Q = ff_aac_pow2sf_tab [q_idx];
@@ -158,7 +162,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
abs_pow34_v(s->scoefs, in, size);
scaled = s->scoefs;
}
- quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb]);
+ quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb], ROUNDING);
if (BT_UNSIGNED) {
off = 0;
} else {
@@ -185,7 +189,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
di = t - CLIPPED_ESCAPE;
curbits += 21;
} else {
- int c = av_clip_uintp2(quant(t, Q), 13);
+ int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13);
di = t - c*cbrtf(c)*IQ;
curbits += av_log2(c)*2 - 4 + 1;
}
@@ -215,7 +219,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
if (BT_ESC) {
for (j = 0; j < 2; j++) {
if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
- int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q), 13);
+ int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13);
int len = av_log2(coef);
put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
@@ -240,7 +244,7 @@ static float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitC
return 0.0f;
}
-#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO) \
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \
static float quantize_and_encode_band_cost_ ## NAME( \
struct AACEncContext *s, \
PutBitContext *pb, const float *in, \
@@ -250,17 +254,19 @@ 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_NOISE, BT_STEREO); \
+ BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, \
+ ROUNDING); \
}
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD)
static float (*const quantize_and_encode_band_cost_arr[])(
struct AACEncContext *s,
@@ -286,28 +292,52 @@ static float (*const quantize_and_encode_band_cost_arr[])(
quantize_and_encode_band_cost_STEREO,
};
+static float (*const quantize_and_encode_band_cost_rtz_arr[])(
+ 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) = {
+ quantize_and_encode_band_cost_ZERO,
+ quantize_and_encode_band_cost_SQUAD,
+ quantize_and_encode_band_cost_SQUAD,
+ quantize_and_encode_band_cost_UQUAD,
+ quantize_and_encode_band_cost_UQUAD,
+ quantize_and_encode_band_cost_SPAIR,
+ quantize_and_encode_band_cost_SPAIR,
+ quantize_and_encode_band_cost_UPAIR,
+ quantize_and_encode_band_cost_UPAIR,
+ quantize_and_encode_band_cost_UPAIR,
+ quantize_and_encode_band_cost_UPAIR,
+ quantize_and_encode_band_cost_ESC_RTZ,
+ quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */
+ quantize_and_encode_band_cost_NOISE,
+ quantize_and_encode_band_cost_STEREO,
+ quantize_and_encode_band_cost_STEREO,
+};
+
#define quantize_and_encode_band_cost( \
s, pb, in, scaled, size, scale_idx, cb, \
- lambda, uplim, bits) \
- quantize_and_encode_band_cost_arr[cb]( \
+ lambda, uplim, bits, rtz) \
+ ((rtz) ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb]( \
s, pb, in, scaled, size, scale_idx, cb, \
lambda, uplim, bits)
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)
+ int *bits, int rtz)
{
return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
- cb, lambda, uplim, bits);
+ cb, lambda, uplim, bits, rtz);
}
static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
const float *in, int size, int scale_idx,
- int cb, const float lambda)
+ int cb, const float lambda, int rtz)
{
quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
- INFINITY, NULL);
+ INFINITY, NULL, rtz);
}
static float find_max_val(int group_len, int swb_size, const float *scaled) {
@@ -397,7 +427,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
rd += quantize_band_cost(s, sce->coeffs + start + w*128,
s->scoefs + start + w*128, size,
sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
- lambda / band->threshold, INFINITY, NULL);
+ lambda / band->threshold, INFINITY, NULL, 0);
}
cost_stay_here = path[swb][cb].cost + rd;
cost_get_here = minrd + rd + run_bits + 4;
@@ -527,9 +557,9 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
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],
+ sce->sf_idx[win*16+swb],
aac_cb_out_map[cb],
- 0, INFINITY, NULL);
+ 0, INFINITY, NULL, 0);
}
cost_stay_here = path[swb][cb].cost + bits;
cost_get_here = minbits + bits + run_bits + 4;
@@ -749,7 +779,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
- q + q0, cb, lambda / band->threshold, INFINITY, NULL);
+ q + q0, cb, lambda / band->threshold, INFINITY, NULL, 0);
}
minrd = FFMIN(minrd, dist);
@@ -895,7 +925,8 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
cb,
1.0f,
INFINITY,
- &b);
+ &b,
+ 0);
bits += b;
}
dists[w*16+g] = dist - bits;
@@ -1061,11 +1092,12 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
ESC_BT,
lambda,
INFINITY,
- &b);
+ &b,
+ 0);
dist -= b;
}
dist *= 1.0f / 512.0f / lambda;
- quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
+ quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512], ROUND_STANDARD);
if (quant_max >= 8191) { // too much, return to the previous quantizer
sce->sf_idx[w*16+g] = prev_scf;
break;
@@ -1242,19 +1274,19 @@ static void search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElemen
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
- lambda / band0->threshold, INFINITY, NULL);
+ lambda / band0->threshold, INFINITY, NULL, 0);
dist1 += quantize_band_cost(s, sce1->coeffs + start + (w+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);
+ lambda / band1->threshold, INFINITY, NULL, 0);
dist2 += quantize_band_cost(s, IS,
I34,
sce0->ics.swb_sizes[g],
is_sf_idx,
is_band_type,
- lambda / minthr, INFINITY, NULL);
+ lambda / minthr, INFINITY, NULL, 0);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
@@ -1315,25 +1347,25 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
- lambda / band0->threshold, INFINITY, NULL);
+ lambda / band0->threshold, INFINITY, NULL, 0);
dist1 += quantize_band_cost(s, sce1->coeffs + start + (w+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);
+ lambda / band1->threshold, INFINITY, NULL, 0);
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);
+ lambda / maxthr, INFINITY, NULL, 0);
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);
+ lambda / minthr, INFINITY, NULL, 0);
}
cpe->ms_mask[w*16+g] = dist2 < dist1;
}
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index a3c31de684..c3c72aff31 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -472,13 +472,33 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
- s->lambda);
+ s->lambda, sce->ics.window_clipping[w]);
start += sce->ics.swb_sizes[i];
}
}
}
/**
+ * Downscale spectral coefficients for near-clipping windows to avoid artifacts
+ */
+static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
+{
+ int start, i, j, w;
+
+ if (sce->ics.clip_avoidance_factor < 1.0f) {
+ for (w = 0; w < sce->ics.num_windows; w++) {
+ start = 0;
+ for (i = 0; i < sce->ics.max_sfb; i++) {
+ float *swb_coeffs = sce->coeffs + start + w*128;
+ for (j = 0; j < sce->ics.swb_sizes[i]; j++)
+ swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
+ start += sce->ics.swb_sizes[i];
+ }
+ }
+ }
+}
+
+/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
@@ -578,6 +598,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
int cur_channel = start_ch + ch;
+ float clip_avoidance_factor;
overlap = &samples[cur_channel][0];
samples2 = overlap + 1024;
la = samples2 + (448+64);
@@ -605,14 +626,29 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
ics->num_windows = wi[ch].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
+ clip_avoidance_factor = 0.0f;
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];
+ for (w = 0; w < ics->num_windows; w++) {
+ if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
+ ics->window_clipping[w] = 1;
+ clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
+ } else {
+ ics->window_clipping[w] = 0;
+ }
+ }
+ if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
+ ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
+ } else {
+ ics->clip_avoidance_factor = 1.0f;
+ }
apply_window_and_mdct(s, &cpe->ch[ch], overlap);
if (isnan(cpe->ch->coeffs[0])) {
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
return AVERROR(EINVAL);
}
+ avoid_clipping(s, &cpe->ch[ch]);
}
start_ch += chans;
}
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index 42104552eb..1f05aabd76 100644
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -54,7 +54,7 @@ typedef struct AACCoefficientsEncoder {
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);
+ int scale_idx, int cb, const float lambda, int rtz);
void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce, const float lambda);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index b16f6b94f0..a5474b9383 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -837,6 +837,7 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
int grouping = 0;
int uselongblock = 1;
int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
+ float clippings[AAC_NUM_BLOCKS_SHORT];
int i;
FFPsyWindowInfo wi = { { 0 } };
@@ -926,14 +927,35 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
lame_apply_block_type(pch, &wi, uselongblock);
+ /* Calculate input sample maximums and evaluate clipping risk */
+ if (audio) {
+ for (i = 0; i < AAC_NUM_BLOCKS_SHORT; i++) {
+ const float *wbuf = audio + i * AAC_BLOCK_SIZE_SHORT;
+ float max = 0;
+ int j;
+ for (j = 0; j < AAC_BLOCK_SIZE_SHORT; j++)
+ max = FFMAX(max, fabsf(wbuf[j]));
+ clippings[i] = max;
+ }
+ } else {
+ for (i = 0; i < 8; i++)
+ clippings[i] = 0;
+ }
+
wi.window_type[1] = prev_type;
if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
+ float clipping = 0.0f;
+
wi.num_windows = 1;
wi.grouping[0] = 1;
if (wi.window_type[0] == LONG_START_SEQUENCE)
wi.window_shape = 0;
else
wi.window_shape = 1;
+
+ for (i = 0; i < 8; i++)
+ clipping = FFMAX(clipping, clippings[i]);
+ wi.clipping[0] = clipping;
} else {
int lastgrp = 0;
@@ -944,6 +966,14 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
lastgrp = i;
wi.grouping[lastgrp]++;
}
+
+ for (i = 0; i < 8; i += wi.grouping[i]) {
+ int w;
+ float clipping = 0.0f;
+ for (w = 0; w < wi.grouping[i] && !clipping; w++)
+ clipping = FFMAX(clipping, clippings[i+w]);
+ wi.clipping[i] = clipping;
+ }
}
/* Determine grouping, based on the location of the first attack, and save for
diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
index 2e3ab911e4..e9be1f6fa5 100644
--- a/libavcodec/psymodel.h
+++ b/libavcodec/psymodel.h
@@ -66,6 +66,7 @@ typedef struct FFPsyWindowInfo {
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)
+ float clipping[8]; ///< maximum absolute normalized intensity in the given window for clip avoidance
int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
} FFPsyWindowInfo;
diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak
index 7ebec45e02..9b0895911e 100644
--- a/tests/fate/aac.mak
+++ b/tests/fate/aac.mak
@@ -146,7 +146,7 @@ fate-aac-aref-encode: CMD = enc_dec_pcm adts wav s16le $(REF) -strict -2 -c:a aa
fate-aac-aref-encode: CMP = stddev
fate-aac-aref-encode: REF = ./tests/data/asynth-44100-2.wav
fate-aac-aref-encode: CMP_SHIFT = -4096
-fate-aac-aref-encode: CMP_TARGET = 434
+fate-aac-aref-encode: CMP_TARGET = 594
fate-aac-aref-encode: SIZE_TOLERANCE = 2464
fate-aac-aref-encode: FUZZ = 5
@@ -155,7 +155,7 @@ fate-aac-ln-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
fate-aac-ln-encode: CMP = stddev
fate-aac-ln-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ln-encode: CMP_SHIFT = -4096
-fate-aac-ln-encode: CMP_TARGET = 65
+fate-aac-ln-encode: CMP_TARGET = 68
fate-aac-ln-encode: SIZE_TOLERANCE = 3560
FATE_AAC_LATM += fate-aac-latm_000000001180bc60