2015-07-29 06:44:27 +02:00
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/*
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* AAC encoder utilities
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* Copyright (C) 2015 Rostislav Pehlivanov
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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 utilities
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* @author Rostislav Pehlivanov ( atomnuker gmail com )
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*/
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#ifndef AVCODEC_AACENC_UTILS_H
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#define AVCODEC_AACENC_UTILS_H
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#include "aac.h"
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#include "aac_tablegen_decl.h"
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#include "aacenctab.h"
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#define ROUND_STANDARD 0.4054f
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#define ROUND_TO_ZERO 0.1054f
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#define C_QUANT 0.4054f
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static inline void abs_pow34_v(float *out, const float *in, const int size)
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{
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int i;
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for (i = 0; i < size; i++) {
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float a = fabsf(in[i]);
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out[i] = sqrtf(a * sqrtf(a));
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}
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}
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/**
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* Quantize one coefficient.
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* @return absolute value of the quantized coefficient
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* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
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*/
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static inline int quant(float coef, const float Q, const float rounding)
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{
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float a = coef * Q;
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return sqrtf(a * sqrtf(a)) + rounding;
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}
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static inline void quantize_bands(int *out, const float *in, const float *scaled,
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int size, float Q34, int is_signed, int maxval,
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const float rounding)
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{
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int i;
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double qc;
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for (i = 0; i < size; i++) {
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qc = scaled[i] * Q34;
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out[i] = (int)FFMIN(qc + rounding, (double)maxval);
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if (is_signed && in[i] < 0.0f) {
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out[i] = -out[i];
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}
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}
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}
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static inline float find_max_val(int group_len, int swb_size, const float *scaled)
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{
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float maxval = 0.0f;
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int w2, i;
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for (w2 = 0; w2 < group_len; w2++) {
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for (i = 0; i < swb_size; i++) {
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maxval = FFMAX(maxval, scaled[w2*128+i]);
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}
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}
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return maxval;
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}
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static inline int find_min_book(float maxval, int sf)
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{
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float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
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float Q34 = sqrtf(Q * sqrtf(Q));
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int qmaxval, cb;
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qmaxval = maxval * Q34 + C_QUANT;
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2015-09-21 08:41:26 +02:00
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if (qmaxval >= (FF_ARRAY_ELEMS(aac_maxval_cb)))
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cb = 11;
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else
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cb = aac_maxval_cb[qmaxval];
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2015-07-29 06:44:27 +02:00
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return cb;
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}
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2015-10-12 02:12:43 +02:00
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static inline float find_form_factor(int group_len, int swb_size, float thresh, const float *scaled, float nzslope) {
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AAC encoder: Extensive improvements
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
2015-10-11 22:29:50 +02:00
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const float iswb_size = 1.0f / swb_size;
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const float iswb_sizem1 = 1.0f / (swb_size - 1);
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const float ethresh = thresh;
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float form = 0.0f, weight = 0.0f;
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int w2, i;
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for (w2 = 0; w2 < group_len; w2++) {
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float e = 0.0f, e2 = 0.0f, var = 0.0f, maxval = 0.0f;
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float nzl = 0;
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for (i = 0; i < swb_size; i++) {
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float s = fabsf(scaled[w2*128+i]);
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maxval = FFMAX(maxval, s);
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e += s;
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e2 += s *= s;
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/* We really don't want a hard non-zero-line count, since
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* even below-threshold lines do add up towards band spectral power.
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* So, fall steeply towards zero, but smoothly
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*/
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if (s >= ethresh) {
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nzl += 1.0f;
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} else {
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nzl += powf(s / ethresh, nzslope);
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}
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}
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if (e2 > thresh) {
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float frm;
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e *= iswb_size;
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/** compute variance */
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for (i = 0; i < swb_size; i++) {
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float d = fabsf(scaled[w2*128+i]) - e;
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var += d*d;
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}
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var = sqrtf(var * iswb_sizem1);
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e2 *= iswb_size;
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frm = e / FFMIN(e+4*var,maxval);
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form += e2 * sqrtf(frm) / FFMAX(0.5f,nzl);
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weight += e2;
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}
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}
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if (weight > 0) {
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return form / weight;
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} else {
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return 1.0f;
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}
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}
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2015-07-29 06:44:27 +02:00
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/** Return the minimum scalefactor where the quantized coef does not clip. */
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static inline uint8_t coef2minsf(float coef)
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{
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return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
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}
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/** Return the maximum scalefactor where the quantized coef is not zero. */
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static inline uint8_t coef2maxsf(float coef)
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{
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return av_clip_uint8(log2f(coef)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
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}
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/*
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* Returns the closest possible index to an array of float values, given a value.
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*/
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static inline int quant_array_idx(const float val, const float *arr, const int num)
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{
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int i, index = 0;
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float quant_min_err = INFINITY;
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for (i = 0; i < num; i++) {
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float error = (val - arr[i])*(val - arr[i]);
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if (error < quant_min_err) {
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quant_min_err = error;
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index = i;
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}
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}
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return index;
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}
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AAC encoder: Extensive improvements
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
2015-10-11 22:29:50 +02:00
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/**
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* approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
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*/
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static av_always_inline float bval2bmax(float b)
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{
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return 0.001f + 0.0035f * (b*b*b) / (15.5f*15.5f*15.5f);
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}
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2015-09-06 16:06:34 +02:00
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/*
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* linear congruential pseudorandom number generator, copied from the decoder
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*/
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static inline int lcg_random(unsigned previous_val)
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{
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union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
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return v.s;
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}
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2015-07-29 06:44:27 +02:00
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#define ERROR_IF(cond, ...) \
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if (cond) { \
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av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
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return AVERROR(EINVAL); \
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}
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#define WARN_IF(cond, ...) \
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if (cond) { \
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av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
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}
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2015-10-12 17:50:10 +02:00
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#define AAC_OPT_SET(e_opt, p_opt, bypass, name) \
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ERROR_IF ((e_opt)->name == 1 && (p_opt)->name == OPT_BANNED, \
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"Profile %i does not allow %s\n", avctx->profile, #name); \
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ERROR_IF ((e_opt)->name == 0 && (p_opt)->name == OPT_REQUIRED, \
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"Option %s is a requirement for this profile (%i)\n", \
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#name, avctx->profile); \
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if ((e_opt)->name == 1 && (p_opt)->name == OPT_NEEDS_MAIN && \
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avctx->profile == FF_PROFILE_AAC_LOW) { \
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WARN_IF(1, "Profile %i does not allow for %s, setting profile to " \
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"\"aac_main\"(%i)\n", avctx->profile, #name, \
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FF_PROFILE_AAC_MAIN); \
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avctx->profile = FF_PROFILE_AAC_MAIN; \
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p_opt = &aacenc_profiles[FF_PROFILE_AAC_MAIN].opts; \
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} \
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if ((e_opt)->name == 1 && (p_opt)->name == OPT_NEEDS_LTP && \
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avctx->profile == FF_PROFILE_AAC_LOW) { \
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WARN_IF(1, "Profile %i does not allow for %s, setting profile to " \
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"\"aac_ltp\"(%i)\n", avctx->profile, #name, \
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FF_PROFILE_AAC_LTP); \
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avctx->profile = FF_PROFILE_AAC_LTP; \
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p_opt = &aacenc_profiles[FF_PROFILE_AAC_LTP].opts; \
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} \
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if ((e_opt)->name == OPT_AUTO) { \
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if ((p_opt)->name == OPT_BANNED) { \
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(e_opt)->name = 0; \
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} else if ((p_opt)->name == OPT_NEEDS_LTP) { \
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(e_opt)->name = 0; \
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} else if ((p_opt)->name == OPT_NEEDS_MAIN) { \
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(e_opt)->name = 0; \
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} else if ((p_opt)->name == OPT_REQUIRED) { \
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(e_opt)->name = 1; \
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} else if (bypass) { \
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(e_opt)->name = (e_opt)->name; \
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} else { \
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(e_opt)->name = (p_opt)->name; \
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} \
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} \
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av_log(avctx, AV_LOG_VERBOSE, "Option %s set to %i\n", #name, (e_opt)->name);
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2015-07-29 06:44:27 +02:00
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#endif /* AVCODEC_AACENC_UTILS_H */
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