1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
FFmpeg/libavcodec/aacenc_utils.h
Claudio Freire ca203e9985 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
2015-12-02 07:47:37 -03:00

269 lines
8.1 KiB
C

/*
* AAC encoder utilities
* Copyright (C) 2015 Rostislav Pehlivanov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC encoder utilities
* @author Rostislav Pehlivanov ( atomnuker gmail com )
*/
#ifndef AVCODEC_AACENC_UTILS_H
#define AVCODEC_AACENC_UTILS_H
#include "aac.h"
#include "aacenctab.h"
#include "aactab.h"
#define ROUND_STANDARD 0.4054f
#define ROUND_TO_ZERO 0.1054f
#define C_QUANT 0.4054f
static inline void abs_pow34_v(float *out, const float *in, const int size)
{
int i;
for (i = 0; i < size; i++) {
float a = fabsf(in[i]);
out[i] = sqrtf(a * sqrtf(a));
}
}
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static inline int quant(float coef, const float Q, const float rounding)
{
float a = coef * Q;
return sqrtf(a * sqrtf(a)) + rounding;
}
static inline void quantize_bands(int *out, const float *in, const float *scaled,
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 + rounding, (double)maxval);
if (is_signed && in[i] < 0.0f) {
out[i] = -out[i];
}
}
}
static inline float find_max_val(int group_len, int swb_size, const float *scaled)
{
float maxval = 0.0f;
int w2, i;
for (w2 = 0; w2 < group_len; w2++) {
for (i = 0; i < swb_size; i++) {
maxval = FFMAX(maxval, scaled[w2*128+i]);
}
}
return maxval;
}
static inline int find_min_book(float maxval, int sf)
{
float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
float Q34 = sqrtf(Q * sqrtf(Q));
int qmaxval, cb;
qmaxval = maxval * Q34 + C_QUANT;
if (qmaxval >= (FF_ARRAY_ELEMS(aac_maxval_cb)))
cb = 11;
else
cb = aac_maxval_cb[qmaxval];
return cb;
}
static inline float find_form_factor(int group_len, int swb_size, float thresh,
const float *scaled, float nzslope) {
const float iswb_size = 1.0f / swb_size;
const float iswb_sizem1 = 1.0f / (swb_size - 1);
const float ethresh = thresh;
float form = 0.0f, weight = 0.0f;
int w2, i;
for (w2 = 0; w2 < group_len; w2++) {
float e = 0.0f, e2 = 0.0f, var = 0.0f, maxval = 0.0f;
float nzl = 0;
for (i = 0; i < swb_size; i++) {
float s = fabsf(scaled[w2*128+i]);
maxval = FFMAX(maxval, s);
e += s;
e2 += s *= s;
/* We really don't want a hard non-zero-line count, since
* even below-threshold lines do add up towards band spectral power.
* So, fall steeply towards zero, but smoothly
*/
if (s >= ethresh) {
nzl += 1.0f;
} else {
nzl += powf(s / ethresh, nzslope);
}
}
if (e2 > thresh) {
float frm;
e *= iswb_size;
/** compute variance */
for (i = 0; i < swb_size; i++) {
float d = fabsf(scaled[w2*128+i]) - e;
var += d*d;
}
var = sqrtf(var * iswb_sizem1);
e2 *= iswb_size;
frm = e / FFMIN(e+4*var,maxval);
form += e2 * sqrtf(frm) / FFMAX(0.5f,nzl);
weight += e2;
}
}
if (weight > 0) {
return form / weight;
} else {
return 1.0f;
}
}
/** Return the minimum scalefactor where the quantized coef does not clip. */
static inline uint8_t coef2minsf(float coef)
{
return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
}
/** Return the maximum scalefactor where the quantized coef is not zero. */
static inline uint8_t coef2maxsf(float coef)
{
return av_clip_uint8(log2f(coef)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
}
/*
* Returns the closest possible index to an array of float values, given a value.
*/
static inline int quant_array_idx(const float val, const float *arr, const int num)
{
int i, index = 0;
float quant_min_err = INFINITY;
for (i = 0; i < num; i++) {
float error = (val - arr[i])*(val - arr[i]);
if (error < quant_min_err) {
quant_min_err = error;
index = i;
}
}
return index;
}
/**
* approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
*/
static av_always_inline float bval2bmax(float b)
{
return 0.001f + 0.0035f * (b*b*b) / (15.5f*15.5f*15.5f);
}
/*
* linear congruential pseudorandom number generator, copied from the decoder
*/
static inline int lcg_random(unsigned previous_val)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
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__); \
return AVERROR(EINVAL); \
}
#define WARN_IF(cond, ...) \
if (cond) { \
av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
}
#endif /* AVCODEC_AACENC_UTILS_H */