[FFmpeg-cvslog] aaccoder: tweak PNS implementation further
Rostislav Pehlivanov
git at videolan.org
Wed Sep 9 11:50:48 CEST 2015
ffmpeg | branch: master | Rostislav Pehlivanov <atomnuker at gmail.com> | Wed Sep 9 10:44:33 2015 +0100| [da64bd6a992c5aa3b07d45412b34d26302e67c82] | committer: Rostislav Pehlivanov
aaccoder: tweak PNS implementation further
This commit changes a few things about the noise substitution
logic:
- Brings back the quantization factor (reduced to 3) during
scalefactor index calculations.
- Rejects any zeroed bands. They should be inaudiable and it's
a waste transmitting the scalefactor indices for these.
- Uses swb_offsets instead of incrementing a 'start' with every
window group size.
- Rejects all PNS during short windows.
Overall improves quality. There was a plan to use the lfg system
to create the random numbers instead of using whatever the decoder
uses but for now this works fine. Entropy is far from important here.
Signed-off-by: Rostislav Pehlivanov <atomnuker at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=da64bd6a992c5aa3b07d45412b34d26302e67c82
---
libavcodec/aaccoder.c | 52 +++++++++++++++++++++++--------------------------
tests/fate/aac.mak | 2 +-
2 files changed, 25 insertions(+), 29 deletions(-)
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
index cf4b11b..dd2e62f 100644
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -51,9 +51,12 @@
/** Frequency in Hz for lower limit of noise substitution **/
#define NOISE_LOW_LIMIT 4000
+/** Pointless to substitute very high short lived inaudiable frequencies **/
+#define NOISE_HIGH_LIMIT 18120
+
/* 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 0.9673f
+#define NOISE_SPREAD_THRESHOLD 0.5073f
/* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to
* replace low energy non zero bands */
@@ -335,7 +338,7 @@ static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement
minscaler_i = FFMIN(minscaler_i, sce->sf_idx[w*16+g]);
bands++;
} else if (sce->band_type[w*16+g] == NOISE_BT) {
- sce->sf_idx[w*16+g] = av_clip(roundf(log2f(sce->pns_ener[w*16+g])*2), -100, 155);
+ sce->sf_idx[w*16+g] = av_clip(3+ceilf(log2f(sce->pns_ener[w*16+g])*2), -100, 155);
minscaler_n = FFMIN(minscaler_n, sce->sf_idx[w*16+g]);
bands++;
}
@@ -863,7 +866,7 @@ static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
{
FFPsyBand *band;
- int w, g, w2, i, start, count = 0;
+ int w, g, w2, i;
float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128];
float *NOR34 = &s->scoefs[3*128];
const float lambda = s->lambda;
@@ -871,20 +874,20 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);
const float spread_threshold = NOISE_SPREAD_THRESHOLD*(lambda/100.f);
+ if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
+ return;
+
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 noise_sfi, try_pns = 0;
+ int noise_sfi;
float dist1 = 0.0f, dist2 = 0.0f, noise_amp;
float pns_energy = 0.0f, energy_ratio, dist_thresh;
float sfb_energy = 0.0f, threshold = 0.0f, spread = 0.0f;
- float freq_boost = FFMAX(0.88f*start*freq_mult/NOISE_LOW_LIMIT, 1.0f);
- if (start*freq_mult < NOISE_LOW_LIMIT) {
- start += sce->ics.swb_sizes[g];
+ const int start = sce->ics.swb_offset[w*16+g];
+ const float freq = start*freq_mult;
+ const float freq_boost = FFMAX(0.88f*freq/NOISE_LOW_LIMIT, 1.0f);
+ if (freq < NOISE_LOW_LIMIT)
continue;
- } else {
- dist_thresh = FFMIN(0.008f*(NOISE_LOW_LIMIT/start*freq_mult), 1.11f);
- }
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
sfb_energy += band->energy;
@@ -892,18 +895,12 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
threshold += band->threshold;
}
- if (sce->zeroes[w*16+g]) {
- try_pns = 1;
- } else if (sfb_energy < threshold*freq_boost) {
- try_pns = 1;
- } else if (spread > spread_threshold) {
- try_pns = 0;
- } else if (sfb_energy < threshold*thr_mult*freq_boost) {
- try_pns = 1;
- }
+ /* Ramps down at ~8000Hz and loosens the dist threshold */
+ dist_thresh = FFMIN(2.5f*NOISE_LOW_LIMIT/freq, 1.27f);
- if (!try_pns || !sfb_energy) {
- start += sce->ics.swb_sizes[g];
+ if (sce->zeroes[w*16+g] || spread < spread_threshold ||
+ sfb_energy > threshold*thr_mult*freq_boost || !sfb_energy) {
+ sce->pns_ener[w*16+g] = sfb_energy;
continue;
}
@@ -911,16 +908,17 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
float band_energy, scale;
- band = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
+ const int start_c = sce->ics.swb_offset[(w+w2)*16+g];
+ band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
for (i = 0; i < sce->ics.swb_sizes[g]; i++)
PNS[i] = s->random_state = lcg_random(s->random_state);
band_energy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
scale = noise_amp/sqrtf(band_energy);
s->fdsp->vector_fmul_scalar(PNS, PNS, scale, sce->ics.swb_sizes[g]);
pns_energy += s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
- abs_pow34_v(NOR34, &sce->coeffs[start+(w+w2)*128], sce->ics.swb_sizes[g]);
+ abs_pow34_v(NOR34, &sce->coeffs[start_c], sce->ics.swb_sizes[g]);
abs_pow34_v(PNS34, PNS, sce->ics.swb_sizes[g]);
- dist1 += quantize_band_cost(s, &sce->coeffs[start + (w+w2)*128],
+ dist1 += quantize_band_cost(s, &sce->coeffs[start_c],
NOR34,
sce->ics.swb_sizes[g],
sce->sf_idx[(w+w2)*16+g],
@@ -935,7 +933,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
}
energy_ratio = sfb_energy/pns_energy; /* Compensates for quantization error */
sce->pns_ener[w*16+g] = energy_ratio*sfb_energy;
- if (energy_ratio > 0.80f && energy_ratio < 1.20f && dist1/dist2 > dist_thresh) {
+ if (energy_ratio > 0.85f && energy_ratio < 1.25f && dist1/dist2 > dist_thresh) {
sce->band_type[w*16+g] = NOISE_BT;
sce->zeroes[w*16+g] = 0;
if (sce->band_type[w*16+g-1] != NOISE_BT && /* Prevent holes */
@@ -943,9 +941,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
sce->band_type[w*16+g-1] = NOISE_BT;
sce->zeroes[w*16+g-1] = 0;
}
- count++;
}
- start += sce->ics.swb_sizes[g];
}
}
}
diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak
index 4391106..2f533e2 100644
--- a/tests/fate/aac.mak
+++ b/tests/fate/aac.mak
@@ -163,7 +163,7 @@ 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 = 637.66
+fate-aac-pns-encode: CMP_TARGET = 633.77
fate-aac-pns-encode: SIZE_TOLERANCE = 3560
fate-aac-pns-encode: FUZZ = 5
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