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ffmpeg.wasm-core
Commit ca203e99 authored by Claudio Freire's avatar Claudio Freire
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AAC encoder: improve SF range utilization 

This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.

Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.

1. Increase SF range utilization.

The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.

This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.

2. PNS tweaks

The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.

3. Account for lowpass cutoff during PSY analysis

The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).

This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.

4. Tweaks to RC lambda tracking loop in relation to PNS

Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.

This tweak makes PNS much less aggressive, though it can still
use some further tweaks.

Also update MIPS specializations and adjust fuzz

Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
parent ec83efd4
Expand all Hide whitespace changes
Inline Side-by-side
Showing with 393 additions and 177 deletions
libavcodec/aaccoder.c
View file @ ca203e99
......@@ -54,7 +54,7 @@
/* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread
* beyond which no PNS is used (since the SFBs contain tone rather than noise) */
#define NOISE_SPREAD_THRESHOLD 0.5073f
#define NOISE_SPREAD_THRESHOLD 0.9f
/* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to
* replace low energy non zero bands */
......@@ -591,6 +591,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
int bandwidth, cutoff;
float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128];
float *NOR34 = &s->scoefs[3*128];
uint8_t nextband[128];
const float lambda = s->lambda;
const float freq_mult = avctx->sample_rate*0.5f/wlen;
const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);
......@@ -604,6 +605,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
/** Keep this in sync with twoloop's cutoff selection */
float rate_bandwidth_multiplier = 1.5f;
int prev = -1000, prev_sf = -1;
int frame_bit_rate = (avctx->flags & CODEC_FLAG_QSCALE)
? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
: (avctx->bit_rate / avctx->channels);
......@@ -619,6 +621,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
ff_init_nextband_map(sce, nextband);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
int wstart = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
......@@ -655,16 +658,27 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
*
* At this stage, point 2 is relaxed for zeroed bands near the noise threshold (hole avoidance is more important)
*/
if (((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.5f/freq_boost)) || spread < spread_threshold ||
if ((!sce->zeroes[w*16+g] && !ff_sfdelta_can_remove_band(sce, nextband, prev_sf, w*16+g)) ||
((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.0f/freq_boost)) || spread < spread_threshold ||
(!sce->zeroes[w*16+g] && sce->band_alt[w*16+g] && sfb_energy > threshold*thr_mult*freq_boost) ||
min_energy < pns_transient_energy_r * max_energy ) {
sce->pns_ener[w*16+g] = sfb_energy;
if (!sce->zeroes[w*16+g])
prev_sf = sce->sf_idx[w*16+g];
continue;
}
pns_tgt_energy = sfb_energy*FFMIN(1.0f, spread*spread);
noise_sfi = av_clip(roundf(log2f(pns_tgt_energy)*2), -100, 155); /* Quantize */
noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */
if (prev != -1000) {
int noise_sfdiff = noise_sfi - prev + SCALE_DIFF_ZERO;
if (noise_sfdiff < 0 || noise_sfdiff > 2*SCALE_MAX_DIFF) {
if (!sce->zeroes[w*16+g])
prev_sf = sce->sf_idx[w*16+g];
continue;
}
}
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
float band_energy, scale, pns_senergy;
const int start_c = (w+w2)*128+sce->ics.swb_offset[g];
......@@ -697,7 +711,10 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
if (sce->zeroes[w*16+g] || !sce->band_alt[w*16+g] || (energy_ratio > 0.85f && energy_ratio < 1.25f && dist2 < dist1)) {
sce->band_type[w*16+g] = NOISE_BT;
sce->zeroes[w*16+g] = 0;
prev = noise_sfi;
}
if (!sce->zeroes[w*16+g])
prev_sf = sce->sf_idx[w*16+g];
}
}
}
......@@ -775,7 +792,8 @@ static void mark_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelEleme
static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
{
int start = 0, i, w, w2, g, sid_sf_boost;
int start = 0, i, w, w2, g, sid_sf_boost, prev_mid, prev_side;
uint8_t nextband0[128], nextband1[128];
float M[128], S[128];
float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
const float lambda = s->lambda;
......@@ -784,21 +802,19 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
SingleChannelElement *sce1 = &cpe->ch[1];
if (!cpe->common_window)
return;
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
int min_sf_idx_mid = SCALE_MAX_POS;
int min_sf_idx_side = SCALE_MAX_POS;
for (g = 0; g < sce0->ics.num_swb; g++) {
if (!sce0->zeroes[w*16+g] && sce0->band_type[w*16+g] < RESERVED_BT)
min_sf_idx_mid = FFMIN(min_sf_idx_mid, sce0->sf_idx[w*16+g]);
if (!sce1->zeroes[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
min_sf_idx_side = FFMIN(min_sf_idx_side, sce1->sf_idx[w*16+g]);
}
/** Scout out next nonzero bands */
ff_init_nextband_map(sce0, nextband0);
ff_init_nextband_map(sce1, nextband1);
prev_mid = sce0->sf_idx[0];
prev_side = sce1->sf_idx[0];
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
start = 0;
for (g = 0; g < sce0->ics.num_swb; g++) {
float bmax = bval2bmax(g * 17.0f / sce0->ics.num_swb) / 0.0045f;
cpe->ms_mask[w*16+g] = 0;
if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
if (!sce0->zeroes[w*16+g] && !sce1->zeroes[w*16+g]) {
float Mmax = 0.0f, Smax = 0.0f;
/* Must compute mid/side SF and book for the whole window group */
......@@ -825,16 +841,18 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
int midcb, sidcb;
minidx = FFMIN(sce0->sf_idx[w*16+g], sce1->sf_idx[w*16+g]);
mididx = av_clip(minidx, min_sf_idx_mid, min_sf_idx_mid + SCALE_MAX_DIFF);
sididx = av_clip(minidx - sid_sf_boost * 3, min_sf_idx_side, min_sf_idx_side + SCALE_MAX_DIFF);
midcb = find_min_book(Mmax, mididx);
sidcb = find_min_book(Smax, sididx);
if ((mididx > minidx) || (sididx > minidx)) {
mididx = av_clip(minidx, 0, SCALE_MAX_POS - SCALE_DIV_512);
sididx = av_clip(minidx - sid_sf_boost * 3, 0, SCALE_MAX_POS - SCALE_DIV_512);
if (!cpe->is_mask[w*16+g] && sce0->band_type[w*16+g] != NOISE_BT && sce1->band_type[w*16+g] != NOISE_BT
&& ( !ff_sfdelta_can_replace(sce0, nextband0, prev_mid, mididx, w*16+g)
|| !ff_sfdelta_can_replace(sce1, nextband1, prev_side, sididx, w*16+g))) {
/* scalefactor range violation, bad stuff, will decrease quality unacceptably */
continue;
}
midcb = find_min_book(Mmax, mididx);
sidcb = find_min_book(Smax, sididx);
/* No CB can be zero */
midcb = FFMAX(1,midcb);
sidcb = FFMAX(1,sidcb);
......@@ -900,6 +918,10 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
}
}
}
if (!sce0->zeroes[w*16+g] && sce0->band_type[w*16+g] < RESERVED_BT)
prev_mid = sce0->sf_idx[w*16+g];
if (!sce1->zeroes[w*16+g] && !cpe->is_mask[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
prev_side = sce1->sf_idx[w*16+g];
start += sce0->ics.swb_sizes[g];
}
}
......
libavcodec/aaccoder_twoloop.h
View file @ ca203e99
This diff is collapsed.
libavcodec/aacenc.c
View file @ ca203e99
......@@ -793,7 +793,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
s->lambda = FFMIN(s->lambda * ratio, 65536.f);
/* Keep iterating if we must reduce and lambda is in the sky */
if ((s->lambda < 300.f || ratio > 0.9f) && (s->lambda > 10.f || ratio < 1.1f)) {
if (ratio > 0.9f && ratio < 1.1f) {
break;
} else {
if (is_mode || ms_mode || tns_mode || pred_mode) {
......
libavcodec/aacenc_is.c
View file @ ca203e99
......@@ -99,18 +99,23 @@ void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElemen
{
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
int start = 0, count = 0, w, w2, g, i;
int start = 0, count = 0, w, w2, g, i, prev_sf1 = -1;
const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
uint8_t nextband1[128];
if (!cpe->common_window)
return;
/** Scout out next nonzero bands */
ff_init_nextband_map(sce1, nextband1);
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
start = 0;
for (g = 0; g < sce0->ics.num_swb; g++) {
if (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g]) {
cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g] &&
ff_sfdelta_can_remove_band(sce1, nextband1, prev_sf1, w*16+g)) {
float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f, ener01p = 0.0f;
struct AACISError ph_err1, ph_err2, *erf;
if (sce0->band_type[w*16+g] == NOISE_BT ||
......@@ -142,6 +147,8 @@ void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElemen
count++;
}
}
if (!sce1->zeroes[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
prev_sf1 = sce1->sf_idx[w*16+g];
start += sce0->ics.swb_sizes[g];
}
}
......
libavcodec/aacenc_utils.h
View file @ ca203e99
......@@ -191,6 +191,69 @@ static inline int lcg_random(unsigned previous_val)
return v.s;
}
/*
* Compute a nextband map to be used with SF delta constraint utilities.
* The nextband array should contain 128 elements, and positions that don't
* map to valid, nonzero bands of the form w*16+g (with w being the initial
* window of the window group, only) are left indetermined.
*/
static inline void ff_init_nextband_map(const SingleChannelElement *sce, uint8_t *nextband)
{
unsigned char prevband = 0;
int w, g;
/** Just a safe default */
for (g = 0; g < 128; g++)
nextband[g] = g;
/** Now really navigate the nonzero band chain */
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->band_type[w*16+g] < RESERVED_BT)
prevband = nextband[prevband] = w*16+g;
}
}
nextband[prevband] = prevband; /* terminate */
}
/*
* Updates nextband to reflect a removed band (equivalent to
* calling ff_init_nextband_map after marking a band as zero)
*/
static inline void ff_nextband_remove(uint8_t *nextband, int prevband, int band)
{
nextband[prevband] = nextband[band];
}
/*
* Checks whether the specified band could be removed without inducing
* scalefactor delta that violates SF delta encoding constraints.
* prev_sf has to be the scalefactor of the previous nonzero, nonspecial
* band, in encoding order, or negative if there was no such band.
*/
static inline int ff_sfdelta_can_remove_band(const SingleChannelElement *sce,
const uint8_t *nextband, int prev_sf, int band)
{
return prev_sf >= 0
&& sce->sf_idx[nextband[band]] >= (prev_sf - SCALE_MAX_DIFF)
&& sce->sf_idx[nextband[band]] <= (prev_sf + SCALE_MAX_DIFF);
}
/*
* Checks whether the specified band's scalefactor could be replaced
* with another one without violating SF delta encoding constraints.
* prev_sf has to be the scalefactor of the previous nonzero, nonsepcial
* band, in encoding order, or negative if there was no such band.
*/
static inline int ff_sfdelta_can_replace(const SingleChannelElement *sce,
const uint8_t *nextband, int prev_sf, int new_sf, int band)
{
return new_sf >= (prev_sf - SCALE_MAX_DIFF)
&& new_sf <= (prev_sf + SCALE_MAX_DIFF)
&& sce->sf_idx[nextband[band]] >= (new_sf - SCALE_MAX_DIFF)
&& sce->sf_idx[nextband[band]] <= (new_sf + SCALE_MAX_DIFF);
}
#define ERROR_IF(cond, ...) \
if (cond) { \
av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
......
libavcodec/aacpsy.c
View file @ ca203e99
......@@ -305,7 +305,7 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
float prev, minscale, minath, minsnr, pe_min;
int chan_bitrate = ctx->avctx->bit_rate / ((ctx->avctx->flags & CODEC_FLAG_QSCALE) ? 2.0f : ctx->avctx->channels);
const int bandwidth = ctx->avctx->cutoff ? ctx->avctx->cutoff : AAC_CUTOFF(ctx->avctx);
const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx);
const float num_bark = calc_bark((float)bandwidth);
ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
......@@ -595,26 +595,30 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
#ifndef calc_thr_3gpp
static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
const uint8_t *band_sizes, const float *coefs)
const uint8_t *band_sizes, const float *coefs, const int cutoff)
{
int i, w, g;
int start = 0;
int start = 0, wstart = 0;
for (w = 0; w < wi->num_windows*16; w += 16) {
wstart = 0;
for (g = 0; g < num_bands; g++) {
AacPsyBand *band = &pch->band[w+g];
float form_factor = 0.0f;
float Temp;
band->energy = 0.0f;
for (i = 0; i < band_sizes[g]; i++) {
band->energy += coefs[start+i] * coefs[start+i];
form_factor += sqrtf(fabs(coefs[start+i]));
if (wstart < cutoff) {
for (i = 0; i < band_sizes[g]; i++) {
band->energy += coefs[start+i] * coefs[start+i];
form_factor += sqrtf(fabs(coefs[start+i]));
}
}
Temp = band->energy > 0 ? sqrtf((float)band_sizes[g] / band->energy) : 0;
band->thr = band->energy * 0.001258925f;
band->nz_lines = form_factor * sqrtf(Temp);
start += band_sizes[g];
wstart += band_sizes[g];
}
}
}
......@@ -655,9 +659,11 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8];
AacPsyCoeffs *coeffs = pctx->psy_coef[wi->num_windows == 8];
const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx);
const int cutoff = bandwidth * 2048 / wi->num_windows / ctx->avctx->sample_rate;
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs);
calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs, cutoff);
//modify thresholds and energies - spread, threshold in quiet, pre-echo control
for (w = 0; w < wi->num_windows*16; w += 16) {
......
libavcodec/mips/aaccoder_mips.c
View file @ ca203e99
This diff is collapsed.
libavcodec/mips/aacpsy_mips.h
View file @ ca203e99
......@@ -61,58 +61,62 @@
#if HAVE_INLINE_ASM && HAVE_MIPSFPU && ( PSY_LAME_FIR_LEN == 21 )
static void calc_thr_3gpp_mips(const FFPsyWindowInfo *wi, const int num_bands,
AacPsyChannel *pch, const uint8_t *band_sizes,
const float *coefs)
const float *coefs, const int cutoff)
{
int i, w, g;
int start = 0;
int start = 0, wstart = 0;
for (w = 0; w < wi->num_windows*16; w += 16) {
wstart = 0;
for (g = 0; g < num_bands; g++) {
AacPsyBand *band = &pch->band[w+g];
float form_factor = 0.0f;
float Temp;
band->energy = 0.0f;
for (i = 0; i < band_sizes[g]; i+=4) {
float a, b, c, d;
float ax, bx, cx, dx;
float *cf = (float *)&coefs[start+i];
__asm__ volatile (
"lwc1 %[a], 0(%[cf]) \n\t"
"lwc1 %[b], 4(%[cf]) \n\t"
"lwc1 %[c], 8(%[cf]) \n\t"
"lwc1 %[d], 12(%[cf]) \n\t"
"abs.s %[a], %[a] \n\t"
"abs.s %[b], %[b] \n\t"
"abs.s %[c], %[c] \n\t"
"abs.s %[d], %[d] \n\t"
"sqrt.s %[ax], %[a] \n\t"
"sqrt.s %[bx], %[b] \n\t"
"sqrt.s %[cx], %[c] \n\t"
"sqrt.s %[dx], %[d] \n\t"
"madd.s %[e], %[e], %[a], %[a] \n\t"
"madd.s %[e], %[e], %[b], %[b] \n\t"
"madd.s %[e], %[e], %[c], %[c] \n\t"
"madd.s %[e], %[e], %[d], %[d] \n\t"
"add.s %[f], %[f], %[ax] \n\t"
"add.s %[f], %[f], %[bx] \n\t"
"add.s %[f], %[f], %[cx] \n\t"
"add.s %[f], %[f], %[dx] \n\t"
: [a]"=&f"(a), [b]"=&f"(b),
[c]"=&f"(c), [d]"=&f"(d),
[e]"+f"(band->energy), [f]"+f"(form_factor),
[ax]"=&f"(ax), [bx]"=&f"(bx),
[cx]"=&f"(cx), [dx]"=&f"(dx)
: [cf]"r"(cf)
: "memory"
);
if (wstart < cutoff) {
for (i = 0; i < band_sizes[g]; i+=4) {
float a, b, c, d;
float ax, bx, cx, dx;
float *cf = (float *)&coefs[start+i];
__asm__ volatile (
"lwc1 %[a], 0(%[cf]) \n\t"
"lwc1 %[b], 4(%[cf]) \n\t"
"lwc1 %[c], 8(%[cf]) \n\t"
"lwc1 %[d], 12(%[cf]) \n\t"
"abs.s %[a], %[a] \n\t"
"abs.s %[b], %[b] \n\t"
"abs.s %[c], %[c] \n\t"
"abs.s %[d], %[d] \n\t"
"sqrt.s %[ax], %[a] \n\t"
"sqrt.s %[bx], %[b] \n\t"
"sqrt.s %[cx], %[c] \n\t"
"sqrt.s %[dx], %[d] \n\t"
"madd.s %[e], %[e], %[a], %[a] \n\t"
"madd.s %[e], %[e], %[b], %[b] \n\t"
"madd.s %[e], %[e], %[c], %[c] \n\t"
"madd.s %[e], %[e], %[d], %[d] \n\t"
"add.s %[f], %[f], %[ax] \n\t"
"add.s %[f], %[f], %[bx] \n\t"
"add.s %[f], %[f], %[cx] \n\t"
"add.s %[f], %[f], %[dx] \n\t"
: [a]"=&f"(a), [b]"=&f"(b),
[c]"=&f"(c), [d]"=&f"(d),
[e]"+f"(band->energy), [f]"+f"(form_factor),
[ax]"=&f"(ax), [bx]"=&f"(bx),
[cx]"=&f"(cx), [dx]"=&f"(dx)
: [cf]"r"(cf)
: "memory"
);
}
}
Temp = sqrtf((float)band_sizes[g] / band->energy);
band->thr = band->energy * 0.001258925f;
band->nz_lines = form_factor * sqrtf(Temp);
start += band_sizes[g];
wstart += band_sizes[g];
}
}
}
......
libavcodec/psymodel.c
View file @ ca203e99
......@@ -39,6 +39,7 @@ av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
ctx->group = av_mallocz_array(sizeof(ctx->group[0]), num_groups);
ctx->bands = av_malloc_array (sizeof(ctx->bands[0]), num_lens);
ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]), num_lens);
ctx->cutoff = avctx->cutoff;
if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
ff_psy_end(ctx);
......
libavcodec/psymodel.h
View file @ ca203e99
......@@ -93,6 +93,7 @@ typedef struct FFPsyContext {
FFPsyChannel *ch; ///< single channel information
FFPsyChannelGroup *group; ///< channel group information
int num_groups; ///< number of channel groups
int cutoff; ///< lowpass frequency cutoff for analysis
uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
int *num_bands; ///< number of scalefactor bands for possible frame sizes
......
tests/fate/aac.mak
View file @ ca203e99
......@@ -146,16 +146,16 @@ 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 = 1139
fate-aac-aref-encode: CMP_TARGET = 670
fate-aac-aref-encode: SIZE_TOLERANCE = 2464
fate-aac-aref-encode: FUZZ = 6
fate-aac-aref-encode: FUZZ = 89
FATE_AAC_ENCODE += fate-aac-ln-encode
fate-aac-ln-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 512k
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 = 80
fate-aac-ln-encode: CMP_TARGET = 50
fate-aac-ln-encode: SIZE_TOLERANCE = 3560
fate-aac-ln-encode: FUZZ = 30
......@@ -164,7 +164,7 @@ fate-aac-ln-encode-128k: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audi
fate-aac-ln-encode-128k: CMP = stddev
fate-aac-ln-encode-128k: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ln-encode-128k: CMP_SHIFT = -4096
fate-aac-ln-encode-128k: CMP_TARGET = 745
fate-aac-ln-encode-128k: CMP_TARGET = 798
fate-aac-ln-encode-128k: SIZE_TOLERANCE = 3560
fate-aac-ln-encode-128k: FUZZ = 5
......@@ -173,16 +173,16 @@ fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
fate-aac-pns-encode: CMP = stddev
fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-pns-encode: CMP_SHIFT = -4096
fate-aac-pns-encode: CMP_TARGET = 695
fate-aac-pns-encode: CMP_TARGET = 663
fate-aac-pns-encode: SIZE_TOLERANCE = 3560
fate-aac-pns-encode: FUZZ = 25
fate-aac-pns-encode: FUZZ = 72
FATE_AAC_ENCODE += fate-aac-tns-encode
fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_tns 1 -aac_is 0 -aac_pns 0 -aac_ms 0 -b:a 128k -cutoff 22050
fate-aac-tns-encode: CMP = stddev
fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-tns-encode: CMP_SHIFT = -4096
fate-aac-tns-encode: CMP_TARGET = 766
fate-aac-tns-encode: CMP_TARGET = 857
fate-aac-tns-encode: FUZZ = 6
fate-aac-tns-encode: SIZE_TOLERANCE = 3560
......@@ -191,25 +191,25 @@ fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
fate-aac-is-encode: CMP = stddev
fate-aac-is-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-is-encode: CMP_SHIFT = -4096
fate-aac-is-encode: CMP_TARGET = 584
fate-aac-is-encode: CMP_TARGET = 725
fate-aac-is-encode: SIZE_TOLERANCE = 3560
fate-aac-is-encode: FUZZ = 1
fate-aac-is-encode: FUZZ = 5
FATE_AAC_ENCODE += fate-aac-ms-encode
fate-aac-ms-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_pns 0 -aac_is 0 -aac_ms 1 -aac_tns 0 -b:a 128k -cutoff 22050
fate-aac-ms-encode: CMP = stddev
fate-aac-ms-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ms-encode: CMP_SHIFT = -4096
fate-aac-ms-encode: CMP_TARGET = 615
fate-aac-ms-encode: CMP_TARGET = 682
fate-aac-ms-encode: SIZE_TOLERANCE = 3560
fate-aac-ms-encode: FUZZ = 10
fate-aac-ms-encode: FUZZ = 15
FATE_AAC_ENCODE += fate-aac-ltp-encode
fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -profile:a aac_ltp -aac_pns 0 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 36k -fflags +bitexact -flags +bitexact
fate-aac-ltp-encode: CMP = stddev
fate-aac-ltp-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ltp-encode: CMP_SHIFT = -4096
fate-aac-ltp-encode: CMP_TARGET = 1120
fate-aac-ltp-encode: CMP_TARGET = 1284
fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
fate-aac-ltp-encode: FUZZ = 17
......@@ -218,7 +218,7 @@ fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-r
fate-aac-pred-encode: CMP = stddev
fate-aac-pred-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-pred-encode: CMP_SHIFT = -4096
fate-aac-pred-encode: CMP_TARGET = 790
fate-aac-pred-encode: CMP_TARGET = 835
fate-aac-pred-encode: FUZZ = 12
fate-aac-pred-encode: SIZE_TOLERANCE = 3560
......
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