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https://github.com/xenia-project/FFmpeg.git
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764 lines
35 KiB
C
764 lines
35 KiB
C
/*
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* AAC encoder twoloop coder
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* Copyright (C) 2008-2009 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder twoloop coder
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* @author Konstantin Shishkov, Claudio Freire
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*/
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/**
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* This file contains a template for the twoloop coder function.
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* It needs to be provided, externally, as an already included declaration,
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* the following functions from aacenc_quantization/util.h. They're not included
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* explicitly here to make it possible to provide alternative implementations:
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* - quantize_band_cost
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* - abs_pow34_v
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* - find_max_val
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* - find_min_book
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* - find_form_factor
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*/
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#ifndef AVCODEC_AACCODER_TWOLOOP_H
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#define AVCODEC_AACCODER_TWOLOOP_H
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#include <float.h>
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#include "libavutil/mathematics.h"
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#include "mathops.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "aac.h"
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#include "aacenc.h"
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#include "aactab.h"
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#include "aacenctab.h"
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/** Frequency in Hz for lower limit of noise substitution **/
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#define NOISE_LOW_LIMIT 4000
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#define sclip(x) av_clip(x,60,218)
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/* Reflects the cost to change codebooks */
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static inline int ff_pns_bits(SingleChannelElement *sce, int w, int g)
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{
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return (!g || !sce->zeroes[w*16+g-1] || !sce->can_pns[w*16+g-1]) ? 9 : 5;
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}
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/**
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* two-loop quantizers search taken from ISO 13818-7 Appendix C
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*/
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static void search_for_quantizers_twoloop(AVCodecContext *avctx,
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AACEncContext *s,
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SingleChannelElement *sce,
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const float lambda)
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{
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int start = 0, i, w, w2, g, recomprd;
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int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
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/ ((avctx->flags & CODEC_FLAG_QSCALE) ? 2.0f : avctx->channels)
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* (lambda / 120.f);
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int refbits = destbits;
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int toomanybits, toofewbits;
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char nzs[128];
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uint8_t nextband[128];
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int maxsf[128], minsf[128];
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float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
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float maxvals[128], spread_thr_r[128];
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float min_spread_thr_r, max_spread_thr_r;
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/**
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* rdlambda controls the maximum tolerated distortion. Twoloop
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* will keep iterating until it fails to lower it or it reaches
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* ulimit * rdlambda. Keeping it low increases quality on difficult
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* signals, but lower it too much, and bits will be taken from weak
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* signals, creating "holes". A balance is necesary.
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* rdmax and rdmin specify the relative deviation from rdlambda
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* allowed for tonality compensation
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*/
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float rdlambda = av_clipf(2.0f * 120.f / lambda, 0.0625f, 16.0f);
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const float nzslope = 1.5f;
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float rdmin = 0.03125f;
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float rdmax = 1.0f;
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/**
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* sfoffs controls an offset of optmium allocation that will be
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* applied based on lambda. Keep it real and modest, the loop
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* will take care of the rest, this just accelerates convergence
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*/
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float sfoffs = av_clipf(log2f(120.0f / lambda) * 4.0f, -5, 10);
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int fflag, minscaler, maxscaler, nminscaler;
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int its = 0;
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int maxits = 30;
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int allz = 0;
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int tbits;
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int cutoff = 1024;
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int pns_start_pos;
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int prev;
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/**
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* zeroscale controls a multiplier of the threshold, if band energy
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* is below this, a zero is forced. Keep it lower than 1, unless
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* low lambda is used, because energy < threshold doesn't mean there's
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* no audible signal outright, it's just energy. Also make it rise
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* slower than rdlambda, as rdscale has due compensation with
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* noisy band depriorization below, whereas zeroing logic is rather dumb
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*/
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float zeroscale;
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if (lambda > 120.f) {
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zeroscale = av_clipf(powf(120.f / lambda, 0.25f), 0.0625f, 1.0f);
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} else {
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zeroscale = 1.f;
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}
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if (s->psy.bitres.alloc >= 0) {
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/**
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* Psy granted us extra bits to use, from the reservoire
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* adjust for lambda except what psy already did
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*/
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destbits = s->psy.bitres.alloc
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* (lambda / (avctx->global_quality ? avctx->global_quality : 120));
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}
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if (avctx->flags & CODEC_FLAG_QSCALE) {
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/**
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* Constant Q-scale doesn't compensate MS coding on its own
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* No need to be overly precise, this only controls RD
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* adjustment CB limits when going overboard
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*/
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if (s->options.mid_side && s->cur_type == TYPE_CPE)
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destbits *= 2;
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/**
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* When using a constant Q-scale, don't adjust bits, just use RD
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* Don't let it go overboard, though... 8x psy target is enough
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*/
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toomanybits = 5800;
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toofewbits = destbits / 16;
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/** Don't offset scalers, just RD */
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sfoffs = sce->ics.num_windows - 1;
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rdlambda = sqrtf(rdlambda);
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/** search further */
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maxits *= 2;
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} else {
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/* When using ABR, be strict, but a reasonable leeway is
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* critical to allow RC to smoothly track desired bitrate
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* without sudden quality drops that cause audible artifacts.
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* Symmetry is also desirable, to avoid systematic bias.
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*/
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toomanybits = destbits + destbits/8;
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toofewbits = destbits - destbits/8;
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sfoffs = 0;
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rdlambda = sqrtf(rdlambda);
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}
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/** and zero out above cutoff frequency */
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{
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int wlen = 1024 / sce->ics.num_windows;
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int bandwidth;
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/**
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* Scale, psy gives us constant quality, this LP only scales
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* bitrate by lambda, so we save bits on subjectively unimportant HF
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* rather than increase quantization noise. Adjust nominal bitrate
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* to effective bitrate according to encoding parameters,
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* AAC_CUTOFF_FROM_BITRATE is calibrated for effective bitrate.
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*/
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float rate_bandwidth_multiplier = 1.5f;
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int frame_bit_rate = (avctx->flags & CODEC_FLAG_QSCALE)
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? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
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: (avctx->bit_rate / avctx->channels);
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/** Compensate for extensions that increase efficiency */
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if (s->options.pns || s->options.intensity_stereo)
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frame_bit_rate *= 1.15f;
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if (avctx->cutoff > 0) {
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bandwidth = avctx->cutoff;
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} else {
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bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
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s->psy.cutoff = bandwidth;
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}
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cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
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pns_start_pos = NOISE_LOW_LIMIT * 2 * wlen / avctx->sample_rate;
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}
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/**
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* for values above this the decoder might end up in an endless loop
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* due to always having more bits than what can be encoded.
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*/
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destbits = FFMIN(destbits, 5800);
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toomanybits = FFMIN(toomanybits, 5800);
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toofewbits = FFMIN(toofewbits, 5800);
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/**
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* XXX: some heuristic to determine initial quantizers will reduce search time
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* determine zero bands and upper distortion limits
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*/
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min_spread_thr_r = -1;
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max_spread_thr_r = -1;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
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int nz = 0;
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float uplim = 0.0f, energy = 0.0f, spread = 0.0f;
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
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if (start >= cutoff || band->energy <= (band->threshold * zeroscale) || band->threshold == 0.0f) {
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sce->zeroes[(w+w2)*16+g] = 1;
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continue;
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}
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nz = 1;
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}
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if (!nz) {
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uplim = 0.0f;
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} else {
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nz = 0;
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
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if (band->energy <= (band->threshold * zeroscale) || band->threshold == 0.0f)
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continue;
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uplim += band->threshold;
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energy += band->energy;
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spread += band->spread;
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nz++;
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}
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}
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uplims[w*16+g] = uplim;
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energies[w*16+g] = energy;
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nzs[w*16+g] = nz;
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sce->zeroes[w*16+g] = !nz;
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allz |= nz;
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if (nz && sce->can_pns[w*16+g]) {
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spread_thr_r[w*16+g] = energy * nz / (uplim * spread);
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if (min_spread_thr_r < 0) {
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min_spread_thr_r = max_spread_thr_r = spread_thr_r[w*16+g];
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} else {
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min_spread_thr_r = FFMIN(min_spread_thr_r, spread_thr_r[w*16+g]);
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max_spread_thr_r = FFMAX(max_spread_thr_r, spread_thr_r[w*16+g]);
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}
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}
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}
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}
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/** Compute initial scalers */
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minscaler = 65535;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (g = 0; g < sce->ics.num_swb; g++) {
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if (sce->zeroes[w*16+g]) {
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sce->sf_idx[w*16+g] = SCALE_ONE_POS;
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continue;
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}
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/**
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* log2f-to-distortion ratio is, technically, 2 (1.5db = 4, but it's power vs level so it's 2).
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* But, as offsets are applied, low-frequency signals are too sensitive to the induced distortion,
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* so we make scaling more conservative by choosing a lower log2f-to-distortion ratio, and thus
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* more robust.
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*/
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sce->sf_idx[w*16+g] = av_clip(
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SCALE_ONE_POS
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+ 1.75*log2f(FFMAX(0.00125f,uplims[w*16+g]) / sce->ics.swb_sizes[g])
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+ sfoffs,
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60, SCALE_MAX_POS);
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minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
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}
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}
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/** Clip */
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minscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
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for (g = 0; g < sce->ics.num_swb; g++)
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if (!sce->zeroes[w*16+g])
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sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF - 1);
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if (!allz)
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return;
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abs_pow34_v(s->scoefs, sce->coeffs, 1024);
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ff_quantize_band_cost_cache_init(s);
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for (i = 0; i < sizeof(minsf) / sizeof(minsf[0]); ++i)
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minsf[i] = 0;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = w*128;
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for (g = 0; g < sce->ics.num_swb; g++) {
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const float *scaled = s->scoefs + start;
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int minsfidx;
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maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
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if (maxvals[w*16+g] > 0) {
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minsfidx = coef2minsf(maxvals[w*16+g]);
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
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minsf[(w+w2)*16+g] = minsfidx;
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}
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start += sce->ics.swb_sizes[g];
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}
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}
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/**
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* Scale uplims to match rate distortion to quality
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* bu applying noisy band depriorization and tonal band priorization.
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* Maxval-energy ratio gives us an idea of how noisy/tonal the band is.
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* If maxval^2 ~ energy, then that band is mostly noise, and we can relax
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* rate distortion requirements.
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*/
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memcpy(euplims, uplims, sizeof(euplims));
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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/** psy already priorizes transients to some extent */
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float de_psy_factor = (sce->ics.num_windows > 1) ? 8.0f / sce->ics.group_len[w] : 1.0f;
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start = w*128;
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for (g = 0; g < sce->ics.num_swb; g++) {
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if (nzs[g] > 0) {
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float cleanup_factor = ff_sqrf(av_clipf(start / (cutoff * 0.75f), 1.0f, 2.0f));
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float energy2uplim = find_form_factor(
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sce->ics.group_len[w], sce->ics.swb_sizes[g],
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uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
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sce->coeffs + start,
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nzslope * cleanup_factor);
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energy2uplim *= de_psy_factor;
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if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
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/** In ABR, we need to priorize less and let rate control do its thing */
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energy2uplim = sqrtf(energy2uplim);
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}
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energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
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uplims[w*16+g] *= av_clipf(rdlambda * energy2uplim, rdmin, rdmax)
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* sce->ics.group_len[w];
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energy2uplim = find_form_factor(
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sce->ics.group_len[w], sce->ics.swb_sizes[g],
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uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
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sce->coeffs + start,
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2.0f);
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energy2uplim *= de_psy_factor;
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if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
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/** In ABR, we need to priorize less and let rate control do its thing */
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energy2uplim = sqrtf(energy2uplim);
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}
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energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
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euplims[w*16+g] *= av_clipf(rdlambda * energy2uplim * sce->ics.group_len[w],
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0.5f, 1.0f);
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}
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start += sce->ics.swb_sizes[g];
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}
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}
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for (i = 0; i < sizeof(maxsf) / sizeof(maxsf[0]); ++i)
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maxsf[i] = SCALE_MAX_POS;
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//perform two-loop search
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//outer loop - improve quality
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do {
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//inner loop - quantize spectrum to fit into given number of bits
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int overdist;
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int qstep = its ? 1 : 32;
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do {
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int changed = 0;
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prev = -1;
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recomprd = 0;
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tbits = 0;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = w*128;
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for (g = 0; g < sce->ics.num_swb; g++) {
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const float *coefs = &sce->coeffs[start];
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const float *scaled = &s->scoefs[start];
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int bits = 0;
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int cb;
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float dist = 0.0f;
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float qenergy = 0.0f;
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if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
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start += sce->ics.swb_sizes[g];
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if (sce->can_pns[w*16+g]) {
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/** PNS isn't free */
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tbits += ff_pns_bits(sce, w, g);
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}
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continue;
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}
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cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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int b;
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float sqenergy;
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dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
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scaled + w2*128,
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sce->ics.swb_sizes[g],
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sce->sf_idx[w*16+g],
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cb,
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1.0f,
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INFINITY,
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&b, &sqenergy,
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0);
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bits += b;
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qenergy += sqenergy;
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}
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dists[w*16+g] = dist - bits;
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qenergies[w*16+g] = qenergy;
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if (prev != -1) {
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int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
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bits += ff_aac_scalefactor_bits[sfdiff];
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}
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tbits += bits;
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start += sce->ics.swb_sizes[g];
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prev = sce->sf_idx[w*16+g];
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}
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}
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if (tbits > toomanybits) {
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recomprd = 1;
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for (i = 0; i < 128; i++) {
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if (sce->sf_idx[i] < (SCALE_MAX_POS - SCALE_DIV_512)) {
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int maxsf_i = (tbits > 5800) ? SCALE_MAX_POS : maxsf[i];
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int new_sf = FFMIN(maxsf_i, sce->sf_idx[i] + qstep);
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if (new_sf != sce->sf_idx[i]) {
|
|
sce->sf_idx[i] = new_sf;
|
|
changed = 1;
|
|
}
|
|
}
|
|
}
|
|
} else if (tbits < toofewbits) {
|
|
recomprd = 1;
|
|
for (i = 0; i < 128; i++) {
|
|
if (sce->sf_idx[i] > SCALE_ONE_POS) {
|
|
int new_sf = FFMAX3(minsf[i], SCALE_ONE_POS, sce->sf_idx[i] - qstep);
|
|
if (new_sf != sce->sf_idx[i]) {
|
|
sce->sf_idx[i] = new_sf;
|
|
changed = 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
qstep >>= 1;
|
|
if (!qstep && tbits > toomanybits && sce->sf_idx[0] < 217 && changed)
|
|
qstep = 1;
|
|
} while (qstep);
|
|
|
|
overdist = 1;
|
|
fflag = tbits < toofewbits;
|
|
for (i = 0; i < 2 && (overdist || recomprd); ++i) {
|
|
if (recomprd) {
|
|
/** Must recompute distortion */
|
|
prev = -1;
|
|
tbits = 0;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
start = w*128;
|
|
for (g = 0; g < sce->ics.num_swb; g++) {
|
|
const float *coefs = sce->coeffs + start;
|
|
const float *scaled = s->scoefs + start;
|
|
int bits = 0;
|
|
int cb;
|
|
float dist = 0.0f;
|
|
float qenergy = 0.0f;
|
|
|
|
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
|
|
start += sce->ics.swb_sizes[g];
|
|
if (sce->can_pns[w*16+g]) {
|
|
/** PNS isn't free */
|
|
tbits += ff_pns_bits(sce, w, g);
|
|
}
|
|
continue;
|
|
}
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g],
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
dists[w*16+g] = dist - bits;
|
|
qenergies[w*16+g] = qenergy;
|
|
if (prev != -1) {
|
|
int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
|
|
bits += ff_aac_scalefactor_bits[sfdiff];
|
|
}
|
|
tbits += bits;
|
|
start += sce->ics.swb_sizes[g];
|
|
prev = sce->sf_idx[w*16+g];
|
|
}
|
|
}
|
|
}
|
|
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]) {
|
|
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
|
|
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++;
|
|
}
|
|
}
|
|
}
|
|
if (overdist) {
|
|
/* We have overdistorted bands, trade for zeroes (that can be noise)
|
|
* Zero the bands in the lowest 1.25% spread-energy-threshold ranking
|
|
*/
|
|
float minspread = max_spread_thr_r;
|
|
float maxspread = min_spread_thr_r;
|
|
float zspread;
|
|
int zeroable = 0;
|
|
int zeroed = 0;
|
|
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]) {
|
|
minspread = FFMIN(minspread, spread_thr_r[w*16+g]);
|
|
maxspread = FFMAX(maxspread, spread_thr_r[w*16+g]);
|
|
zeroable++;
|
|
}
|
|
}
|
|
}
|
|
zspread = (maxspread-minspread) * 0.0125f + minspread;
|
|
/* 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 = fflag = 1;
|
|
} else {
|
|
overdist = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
minscaler = SCALE_MAX_POS;
|
|
maxscaler = 0;
|
|
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]) {
|
|
minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
|
|
maxscaler = FFMAX(maxscaler, sce->sf_idx[w*16+g]);
|
|
}
|
|
}
|
|
}
|
|
|
|
minscaler = nminscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
|
|
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;
|
|
int edepth = depth+2;
|
|
float uplmax = its / (maxits*0.25f) + 1.0f;
|
|
uplmax *= (tbits > destbits) ? FFMIN(2.0f, tbits / (float)FFMAX(1,destbits)) : 1.0f;
|
|
start = w * 128;
|
|
for (g = 0; g < sce->ics.num_swb; g++) {
|
|
int prevsc = sce->sf_idx[w*16+g];
|
|
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]);
|
|
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] > FFMAX(mindeltasf, minsf[w*16+g])) {
|
|
/* 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 && 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);
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
dist = qenergy = 0.f;
|
|
bits = 0;
|
|
if (!cb) {
|
|
maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g]-1, maxsf[w*16+g]);
|
|
} 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;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g]-1,
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
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] < 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] < 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 && 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);
|
|
if (cb > 0) {
|
|
dist = qenergy = 0.f;
|
|
bits = 0;
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g]+1,
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
dist -= bits;
|
|
if (dist < FFMIN(euplims[w*16+g], uplims[w*16+g])) {
|
|
sce->sf_idx[w*16+g]++;
|
|
dists[w*16+g] = dist;
|
|
qenergies[w*16+g] = qenergy;
|
|
} else {
|
|
break;
|
|
}
|
|
} else {
|
|
maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
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]);
|
|
}
|
|
start += sce->ics.swb_sizes[g];
|
|
}
|
|
}
|
|
|
|
/** 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 = 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) {
|
|
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;
|
|
}
|
|
/** Check that there's no SF delta range violations */
|
|
if (!sce->zeroes[w*16+g]) {
|
|
if (prev != -1) {
|
|
av_unused int sfdiff = sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO;
|
|
av_assert1(sfdiff >= 0 && sfdiff <= 2*SCALE_MAX_DIFF);
|
|
} else if (sce->zeroes[0]) {
|
|
/** Set global gain to something useful */
|
|
sce->sf_idx[0] = sce->sf_idx[w*16+g];
|
|
}
|
|
prev = sce->sf_idx[w*16+g];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#endif /* AVCODEC_AACCODER_TWOLOOP_H */
|