Attached
Before pushing this, I'd like some feedback, especially about
the implementation of point 3. I'm not sure the AAC encoder
setting the cutoff in the encoder context like this is legal or desirable.
It does work quite well, and all attempts to do it otherwise were either
very invasive or repeated lots of complex code, so I opted for this
approach. But perhaps someone more familiar with the interfaces
and the contracts behind them could comment on this.
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 the encoder 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.
From e236ab0021eb28012ed05f46e7e520b5cbec413e Mon Sep 17 00:00:00 2001
From: Claudio Freire <klaussfre...@gmail.com>
Date: Sun, 29 Nov 2015 16:33:31 -0300
Subject: [PATCH] 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 the encoder 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.
---
libavcodec/aaccoder.c | 60 +++++++++++++------
libavcodec/aaccoder_twoloop.h | 136 +++++++++++++++++++++++++++++-------------
libavcodec/aacenc.c | 2 +-
libavcodec/aacenc_is.c | 11 +++-
libavcodec/aacenc_utils.h | 59 ++++++++++++++++++
libavcodec/aacpsy.c | 18 ++++--
tests/fate/aac.mak | 18 +++---
7 files changed, 224 insertions(+), 80 deletions(-)
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
index 2a66045..e0d6e50 100644
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -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];
+ unsigned char 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;
+ unsigned char 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];
}
}
diff --git a/libavcodec/aaccoder_twoloop.h b/libavcodec/aaccoder_twoloop.h
index d4290b4..3dc0d9a 100644
--- a/libavcodec/aaccoder_twoloop.h
+++ b/libavcodec/aaccoder_twoloop.h
@@ -76,6 +76,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
int refbits = destbits;
int toomanybits, toofewbits;
char nzs[128];
+ unsigned char nextband[128];
int maxsf[128];
float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
float maxvals[128], spread_thr_r[128];
@@ -102,7 +103,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
*/
float sfoffs = av_clipf(log2f(120.0f / lambda) * 4.0f, -5, 10);
- int fflag, minscaler, maxscaler, nminscaler, minrdsf;
+ int fflag, minscaler, maxscaler, nminscaler;
int its = 0;
int maxits = 30;
int allz = 0;
@@ -158,9 +159,13 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
/** search further */
maxits *= 2;
} else {
- /** When using ABR, be strict */
- toomanybits = destbits + destbits/16;
- toofewbits = destbits - destbits/4;
+ /* When using ABR, be strict, but a reasonable leeway is
+ * critical to allow RC to smoothly track desired bitrate
+ * without sudden quality drops that cause audible artifacts.
+ * Symmetry is also desirable, to avoid systematic bias.
+ */
+ toomanybits = destbits + destbits/8;
+ toofewbits = destbits - destbits/8;
sfoffs = 0;
rdlambda = sqrtf(rdlambda);
@@ -191,6 +196,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
bandwidth = avctx->cutoff;
} else {
bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+ avctx->cutoff = bandwidth;
}
cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
@@ -241,7 +247,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
nzs[w*16+g] = nz;
sce->zeroes[w*16+g] = !nz;
allz |= nz;
- if (nz) {
+ if (nz && sce->can_pns[w*16+g]) {
spread_thr_r[w*16+g] = energy * nz / (uplim * spread);
if (min_spread_thr_r < 0) {
min_spread_thr_r = max_spread_thr_r = spread_thr_r[w*16+g];
@@ -433,6 +439,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
} while (qstep);
overdist = 1;
+ fflag = tbits < toofewbits;
for (i = 0; i < 2 && (overdist || recomprd); ++i) {
if (recomprd) {
/** Must recompute distortion */
@@ -484,13 +491,13 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
}
}
}
- if (!i && s->options.pns && its > maxits/2) {
+ if (!i && s->options.pns && its > maxits/2 && tbits > toofewbits) {
float maxoverdist = 0.0f;
+ float ovrfactor = 1.f+(maxits-its)*16.f/maxits;
overdist = recomprd = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
- float ovrfactor = 2.f+(maxits-its)*16.f/maxits;
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
- if (!sce->zeroes[w*16+g] && dists[w*16+g] > uplims[w*16+g]*ovrfactor) {
+ if (!sce->zeroes[w*16+g] && sce->sf_idx[w*16+g] > SCALE_ONE_POS && dists[w*16+g] > uplims[w*16+g]*ovrfactor) {
float ovrdist = dists[w*16+g] / FFMAX(uplims[w*16+g],euplims[w*16+g]);
maxoverdist = FFMAX(maxoverdist, ovrdist);
overdist++;
@@ -506,7 +513,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
float zspread;
int zeroable = 0;
int zeroed = 0;
- int maxzeroed;
+ int maxzeroed, zloop;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
if (start >= pns_start_pos && !sce->zeroes[w*16+g] && sce->can_pns[w*16+g]) {
@@ -517,21 +524,41 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
}
}
zspread = (maxspread-minspread) * 0.0125f + minspread;
- zspread = FFMIN(maxoverdist, zspread);
- maxzeroed = zeroable * its / (2 * maxits);
- for (g = sce->ics.num_swb-1; g > 0 && zeroed < maxzeroed; g--) {
- if (sce->ics.swb_offset[g] < pns_start_pos)
- continue;
- for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
- if (!sce->zeroes[w*16+g] && sce->can_pns[w*16+g] && spread_thr_r[w*16+g] <= zspread) {
- sce->zeroes[w*16+g] = 1;
- sce->band_type[w*16+g] = 0;
- zeroed++;
+ /* Don't PNS everything even if allowed. It suppresses bit starvation signals from RC,
+ * and forced the hand of the later search_for_pns step.
+ * Instead, PNS a fraction of the spread_thr_r range depending on how starved for bits we are,
+ * and leave further PNSing to search_for_pns if worthwhile.
+ */
+ zspread = FFMIN3(min_spread_thr_r * 8.f, zspread,
+ ((toomanybits - tbits) * min_spread_thr_r + (tbits - toofewbits) * max_spread_thr_r) / (toomanybits - toofewbits + 1));
+ maxzeroed = FFMIN(zeroable, FFMAX(1, (zeroable * its + maxits - 1) / (2 * maxits)));
+ for (zloop = 0; zloop < 2; zloop++) {
+ /* Two passes: first distorted stuff - two birds in one shot and all that,
+ * then anything viable. Viable means not zero, but either CB=zero-able
+ * (too high SF), not SF <= 1 (that means we'd be operating at very high
+ * quality, we don't want PNS when doing VHQ), PNS allowed, and within
+ * the lowest ranking percentile.
+ */
+ float loopovrfactor = (zloop) ? 1.0f : ovrfactor;
+ int loopminsf = (zloop) ? (SCALE_ONE_POS - SCALE_DIV_512) : SCALE_ONE_POS;
+ int mcb;
+ for (g = sce->ics.num_swb-1; g > 0 && zeroed < maxzeroed; g--) {
+ if (sce->ics.swb_offset[g] < pns_start_pos)
+ continue;
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ if (!sce->zeroes[w*16+g] && sce->can_pns[w*16+g] && spread_thr_r[w*16+g] <= zspread
+ && sce->sf_idx[w*16+g] > loopminsf
+ && (dists[w*16+g] > loopovrfactor*uplims[w*16+g] || !(mcb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]))
+ || (mcb <= 1 && dists[w*16+g] > FFMIN(uplims[w*16+g], euplims[w*16+g]))) ) {
+ sce->zeroes[w*16+g] = 1;
+ sce->band_type[w*16+g] = 0;
+ zeroed++;
+ }
}
}
}
if (zeroed)
- recomprd = 1;
+ recomprd = fflag = 1;
} else {
overdist = 0;
}
@@ -549,9 +576,8 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
}
}
- fflag = 0;
minscaler = nminscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
- minrdsf = FFMAX3(60, minscaler - 1, maxscaler - SCALE_MAX_DIFF - 1);
+ prev = -1;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
/** Start with big steps, end up fine-tunning */
int depth = (its > maxits/2) ? ((its > maxits*2/3) ? 1 : 3) : 10;
@@ -561,19 +587,22 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
start = w * 128;
for (g = 0; g < sce->ics.num_swb; g++) {
int prevsc = sce->sf_idx[w*16+g];
- int minrdsfboost = (sce->ics.num_windows > 1) ? av_clip(g-4, -2, 0) : av_clip(g-16, -4, 0);
+ if (prev < 0 && !sce->zeroes[w*16+g])
+ prev = sce->sf_idx[0];
if (!sce->zeroes[w*16+g]) {
const float *coefs = sce->coeffs + start;
const float *scaled = s->scoefs + start;
int cmb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
- if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > minrdsf) {
+ int mindeltasf = FFMAX(0, prev - SCALE_MAX_DIFF);
+ int maxdeltasf = FFMIN(SCALE_MAX_POS - SCALE_DIV_512, prev + SCALE_MAX_DIFF);
+ if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > mindeltasf) {
/* Try to make sure there is some energy in every nonzero band
* NOTE: This algorithm must be forcibly imbalanced, pushing harder
* on holes or more distorted bands at first, otherwise there's
* no net gain (since the next iteration will offset all bands
* on the opposite direction to compensate for extra bits)
*/
- for (i = 0; i < edepth; ++i) {
+ for (i = 0; i < edepth && sce->sf_idx[w*16+g] > mindeltasf; ++i) {
int cb, bits;
float dist, qenergy;
int mb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1);
@@ -585,6 +614,12 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
} else if (i >= depth && dists[w*16+g] < euplims[w*16+g]) {
break;
}
+ /* !g is the DC band, it's important, since quantization error here
+ * applies to less than a cycle, it creates horrible intermodulation
+ * distortion if it doesn't stick to what psy requests
+ */
+ if (!g && sce->ics.num_windows > 1 && dists[w*16+g] >= euplims[w*16+g])
+ maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
float sqenergy;
@@ -603,19 +638,19 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
sce->sf_idx[w*16+g]--;
dists[w*16+g] = dist - bits;
qenergies[w*16+g] = qenergy;
- if (mb && (sce->sf_idx[w*16+g] < (minrdsf+minrdsfboost) || (
+ if (mb && (sce->sf_idx[w*16+g] < mindeltasf || (
(dists[w*16+g] < FFMIN(uplmax*uplims[w*16+g], euplims[w*16+g]))
&& (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
) )) {
break;
}
}
- } else if (tbits > toofewbits && sce->sf_idx[w*16+g] < maxscaler
+ } else if (tbits > toofewbits && sce->sf_idx[w*16+g] < FFMIN(maxdeltasf, maxsf[w*16+g])
&& (dists[w*16+g] < FFMIN(euplims[w*16+g], uplims[w*16+g]))
&& (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
) {
/** Um... over target. Save bits for more important stuff. */
- for (i = 0; i < depth; ++i) {
+ for (i = 0; i < depth && sce->sf_idx[w*16+g] < maxdeltasf; ++i) {
int cb, bits;
float dist, qenergy;
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]+1);
@@ -651,38 +686,53 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
}
}
}
+ prev = sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], mindeltasf, maxdeltasf);
+ if (sce->sf_idx[w*16+g] != prevsc)
+ fflag = 1;
+ nminscaler = FFMIN(nminscaler, sce->sf_idx[w*16+g]);
+ sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
}
- sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minrdsf, minscaler + SCALE_MAX_DIFF);
- sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], SCALE_MAX_POS - SCALE_DIV_512);
- if (sce->sf_idx[w*16+g] != prevsc)
- fflag = 1;
- nminscaler = FFMIN(nminscaler, sce->sf_idx[w*16+g]);
- sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
start += sce->ics.swb_sizes[g];
}
}
- if (nminscaler < minscaler || sce->ics.num_windows > 1) {
- /** SF difference limit violation risk. Must re-clamp. */
- minscaler = nminscaler;
- for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
- for (g = 0; g < sce->ics.num_swb; g++) {
- sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
+
+ /** SF difference limit violation risk. Must re-clamp. */
+ prev = -1;
+ 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]) {
+ int prevsf = sce->sf_idx[w*16+g];
+ if (prev < 0)
+ prev = prevsf;
+ sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], prev - SCALE_MAX_DIFF, prev + SCALE_MAX_DIFF);
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+ prev = sce->sf_idx[w*16+g];
+ if (!fflag && prevsf != sce->sf_idx[w*16+g])
+ fflag = 1;
}
}
}
+
its++;
} while (fflag && its < maxits);
+ /** Scout out next nonzero bands */
+ ff_init_nextband_map(sce, nextband);
+
prev = -1;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
/** Make sure proper codebooks are set */
- for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
+ for (g = 0; g < sce->ics.num_swb; g++) {
if (!sce->zeroes[w*16+g]) {
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
if (sce->band_type[w*16+g] <= 0) {
- sce->zeroes[w*16+g] = 1;
- sce->band_type[w*16+g] = 0;
+ if (!ff_sfdelta_can_remove_band(sce, nextband, prev, w*16+g)) {
+ /** Cannot zero out, make sure it's not attempted */
+ sce->band_type[w*16+g] = 1;
+ } else {
+ sce->zeroes[w*16+g] = 1;
+ sce->band_type[w*16+g] = 0;
+ }
}
} else {
sce->band_type[w*16+g] = 0;
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 0bd8891..971f8ab 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -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) {
diff --git a/libavcodec/aacenc_is.c b/libavcodec/aacenc_is.c
index 97be9b3..2e96c51 100644
--- a/libavcodec/aacenc_is.c
+++ b/libavcodec/aacenc_is.c
@@ -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;
+ unsigned char 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];
}
}
diff --git a/libavcodec/aacenc_utils.h b/libavcodec/aacenc_utils.h
index 40e1746..8b69171 100644
--- a/libavcodec/aacenc_utils.h
+++ b/libavcodec/aacenc_utils.h
@@ -191,6 +191,65 @@ 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, unsigned char *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(unsigned char *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 unsigned char *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 unsigned char *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__); \
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 40b3b41..b33c38f 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -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->avctx->cutoff ? ctx->avctx->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) {
diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak
index 2d41888..a46474b 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 = 1139
+fate-aac-aref-encode: CMP_TARGET = 586
fate-aac-aref-encode: SIZE_TOLERANCE = 2464
fate-aac-aref-encode: FUZZ = 6
@@ -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 = 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
@@ -173,7 +173,7 @@ 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 = 616
fate-aac-pns-encode: SIZE_TOLERANCE = 3560
fate-aac-pns-encode: FUZZ = 25
@@ -182,7 +182,7 @@ fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
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
@@ -191,7 +191,7 @@ 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
@@ -200,7 +200,7 @@ fate-aac-ms-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
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
@@ -209,7 +209,7 @@ fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
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
--
1.8.4.5
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