[FFmpeg-devel] [RFC] AAC Encoder, now more optimal

Kostya kostya.shishkov
Fri Sep 5 15:13:58 CEST 2008


After some time (I'd like to have more free time to spend on it though),
I want to expose my new AAC encoder.

It is slower than realtime since it uses search for optimal quantizers
for given quality.

Known weak points:
* no bitrate management (it uses fixed quality for now)
* no M/S detection
* quantization is not optimal yet

I haven't implemented those because it will slow encoder even more, so
debug session will include an hour rest (and in case of M/S detection
I don't know how to implement it) - 16sec sample coding takes 42 secs
on my PPC G4-1.42GHz and 39 secs on Core2 1.8MHz.

Any comments, suggestions, speedup tricks are extremely welcomed
(especially the latter).
-------------- next part --------------
/*
 * AAC encoder
 * Copyright (C) 2008 Konstantin Shishkov
 *
 * 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 aacenc.c
 * AAC encoder
 */

/***********************************
 *              TODOs:
 * speedup quantizer selection
 * add sane pulse detection
 * add temporal noise shaping
 ***********************************/

#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "mpeg4audio.h"

#include "aac.h"
#include "aactab.h"

#include "psymodel.h"

static const uint8_t swb_size_1024_96[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
};

static const uint8_t swb_size_1024_64[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
};

static const uint8_t swb_size_1024_48[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
    96
};

static const uint8_t swb_size_1024_32[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
};

static const uint8_t swb_size_1024_24[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
};

static const uint8_t swb_size_1024_16[] = {
    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
};

static const uint8_t swb_size_1024_8[] = {
    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};

static const uint8_t *swb_size_1024[] = {
    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
};

static const uint8_t swb_size_128_96[] = {
    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
};

static const uint8_t swb_size_128_48[] = {
    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
};

static const uint8_t swb_size_128_24[] = {
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
};

static const uint8_t swb_size_128_16[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
};

static const uint8_t swb_size_128_8[] = {
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};

static const uint8_t *swb_size_128[] = {
    /* the last entry on the following row is swb_size_128_64 but is a
       duplicate of swb_size_128_96 */
    swb_size_128_96, swb_size_128_96, swb_size_128_96,
    swb_size_128_48, swb_size_128_48, swb_size_128_48,
    swb_size_128_24, swb_size_128_24, swb_size_128_16,
    swb_size_128_16, swb_size_128_16, swb_size_128_8
};

/** spectral coefficients codebook information */
static const struct {
    int16_t maxval;         ///< maximum possible value
     int8_t range;          ///< value used in vector calculation
} aac_cb_info[] = {
    {    0, -1 }, // zero codebook
    {    1,  3 },
    {    1,  3 },
    {    2,  3 },
    {    2,  3 },
    {    4,  9 },
    {    4,  9 },
    {    7,  8 },
    {    7,  8 },
    {   12, 13 },
    {   12, 13 },
    { 8191, 17 },
    {   -1, -1 }, // reserved
    {   -1, -1 }, // perceptual noise substitution
    {   -1, -1 }, // intensity out-of-phase
    {   -1, -1 }, // intensity in-phase
};

/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};

/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};

static const uint8_t* run_value_bits[2] = {
    run_value_bits_long, run_value_bits_short
};

/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
 {1, TYPE_SCE},                               // 1 channel  - single channel element
 {1, TYPE_CPE},                               // 2 channels - channel pair
 {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};

/**
 * structure used in optimal codebook search
 */
typedef struct BandCodingPath {
    int prev_idx; ///< pointer to the previous path point
    int codebook; ///< codebook for coding band run
    int bits;     ///< number of bit needed to code given number of bands
} BandCodingPath;

/**
 * AAC encoder context
 */
typedef struct {
    PutBitContext pb;
    MDCTContext mdct1024;                        ///< long (1024 samples) frame transform context
    MDCTContext mdct128;                         ///< short (128 samples) frame transform context
    DSPContext  dsp;
    DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
    int16_t* samples;                            ///< saved preprocessed input

    int samplerate_index;                        ///< MPEG-4 samplerate index

    ChannelElement *cpe;                         ///< channel elements
    FFPsyContext psy;
    struct FFPsyPreprocessContext* psypp;
    int cur_channel;
    int last_frame;
} AACEncContext;

/**
 * Make AAC audio config object.
 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
 */
static void put_audio_specific_config(AVCodecContext *avctx)
{
    PutBitContext pb;
    AACEncContext *s = avctx->priv_data;

    init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
    put_bits(&pb, 5, 2); //object type - AAC-LC
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
    put_bits(&pb, 4, avctx->channels);
    //GASpecificConfig
    put_bits(&pb, 1, 0); //frame length - 1024 samples
    put_bits(&pb, 1, 0); //does not depend on core coder
    put_bits(&pb, 1, 0); //is not extension
    flush_put_bits(&pb);
}

static av_cold int aac_encode_init(AVCodecContext *avctx)
{
    AACEncContext *s = avctx->priv_data;
    int i;
    const uint8_t *sizes[2];
    int lengths[2];

    avctx->frame_size = 1024;

    for(i = 0; i < 16; i++)
        if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
            break;
    if(i == 16){
        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
        return -1;
    }
    if(avctx->channels > 6){
        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
        return -1;
    }
    s->samplerate_index = i;

    dsputil_init(&s->dsp, avctx);
    ff_mdct_init(&s->mdct1024, 11, 0);
    ff_mdct_init(&s->mdct128,   8, 0);
    // window init
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
    ff_sine_window_init(ff_sine_1024, 1024);
    ff_sine_window_init(ff_sine_128, 128);

    s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
    s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
    avctx->extradata = av_malloc(2);
    avctx->extradata_size = 2;
    put_audio_specific_config(avctx);

    sizes[0] = swb_size_1024[i];
    sizes[1] = swb_size_128[i];
    lengths[0] = ff_aac_num_swb_1024[i];
    lengths[1] = ff_aac_num_swb_128[i];
    ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
    s->psypp = ff_psy_preprocess_init(avctx);

#ifndef CONFIG_HARDCODED_TABLES
    for (i = 0; i < 316; i++)
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */

    return 0;
}

static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
                                  SingleChannelElement *sce, short *audio, int channel)
{
    int i, j, k;
    const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
    const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
    const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
        memcpy(s->output, sce->saved, sizeof(float)*1024);
        if(sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE){
            memset(s->output, 0, sizeof(s->output[0]) * 448);
            for(i = 448; i < 576; i++)
                s->output[i] = sce->saved[i] * pwindow[i - 448];
            for(i = 576; i < 704; i++)
                s->output[i] = sce->saved[i];
        }
        if(sce->ics.window_sequence[0] != LONG_START_SEQUENCE){
            j = channel;
            for (i = 0; i < 1024; i++, j += avctx->channels){
                s->output[i+1024]         = audio[j] * lwindow[1024 - i - 1];
                sce->saved[i] = audio[j] * lwindow[i];
            }
        }else{
            j = channel;
            for(i = 0; i < 448; i++, j += avctx->channels)
                s->output[i+1024]         = audio[j];
            for(i = 448; i < 576; i++, j += avctx->channels)
                s->output[i+1024]         = audio[j] * swindow[576 - i - 1];
            memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
            j = channel;
            for(i = 0; i < 1024; i++, j += avctx->channels)
                sce->saved[i] = audio[j];
        }
        ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
    }else{
        j = channel;
        for (k = 0; k < 1024; k += 128) {
            for(i = 448 + k; i < 448 + k + 256; i++)
                s->output[i - 448 - k] = (i < 1024)
                                         ? sce->saved[i]
                                         : audio[channel + (i-1024)*avctx->channels];
            s->dsp.vector_fmul        (s->output,     k ?  swindow : pwindow, 128);
            s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
            ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
        }
        j = channel;
        for(i = 0; i < 1024; i++, j += avctx->channels)
            sce->saved[i] = audio[j];
    }
}

/**
 * Encode ics_info element.
 * @see Table 4.6 (syntax of ics_info)
 */
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
    int w;

    put_bits(&s->pb, 1, 0);                // ics_reserved bit
    put_bits(&s->pb, 2, info->window_sequence[0]);
    put_bits(&s->pb, 1, info->use_kb_window[0]);
    if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
        put_bits(&s->pb, 6, info->max_sfb);
        put_bits(&s->pb, 1, 0);            // no prediction
    }else{
        put_bits(&s->pb, 4, info->max_sfb);
        for(w = 1; w < 8; w++){
            put_bits(&s->pb, 1, !info->group_len[w]);
        }
    }
}

/**
 * Encode MS data.
 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
 */
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
    int i, w;

    put_bits(pb, 2, cpe->ms_mode);
    if(cpe->ms_mode == 1){
        for(w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]){
            for(i = 0; i < cpe->ch[0].ics.max_sfb; i++)
                put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
        }
    }
}

/**
 * Quantize one coefficient.
 * @return absolute value of the quantized coefficient
 * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
 */
static av_always_inline int quant(float coef, const float Q)
{
    return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
}

static inline float get_approximate_quant_error(const float *q, const int *c, int size, int scale_idx)
{
    int i;
    float coef, unquant, sum = 0.0f;
    const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
    for(i = 0; i < size; i++){
        coef = fabsf(q[i]);
        unquant = (c[i] * cbrt(c[i])) * IQ;
        sum += (coef - unquant) * (coef - unquant);
    }
    return sum;
}

/**
 * Convert coefficients to integers.
 * @fixme make it RD-optimal
 * @return sum of coefficient absolute values
 */
static inline int quantize_band(const float *in, int *out, int size, int scale_idx)
{
    int i, sign, sum = 0;
    const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
    for(i = 0; i < size; i++){
        sign = in[i] > 0.0;
        out[i] = quant(in[i], Q);
        sum += out[i];
        if(sign) out[i] = -out[i];
    }
    return sum;
}

static inline int get_approximate_bits(const int *in, int size)
{
    int i, bits = 0;
    for(i = 0; i < size; i += 2){
        int j, idx = 0;
        for(j = 0; j < 2; j++){
            int t = FFABS(in[i+j]);
            if(t)
                bits++;
            if(t > 16)
                bits += av_log2(t)*2 + 4 - 1;
            idx = idx*17 + FFMIN(t, 16);
        }
        bits += ff_aac_spectral_bits[ESC_BT-1][idx];
    }
    return bits;
}

/**
 * Calculate the number of bits needed to code all coefficient signs in current band.
 */
static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
                                    int group_len, int start, int size)
{
    int bits = 0;
    int i, w;
    for(w = 0; w < group_len; w++){
        for(i = 0; i < size; i++){
            if(sce->icoefs[start + i])
                bits++;
        }
        start += 128;
    }
    return bits;
}

/**
 * Calculate the number of bits needed to code given band with given codebook.
 *
 * @param s         encoder context
 * @param sce       channel element
 * @param group_len window group length
 * @param start     scalefactor band position in spectral coefficients
 * @param size      scalefactor band size
 * @param cb        codebook number
 */
static int calculate_band_bits(AACEncContext *s, SingleChannelElement *sce,
                               int group_len, int start, int size, int cb)
{
    int i, j, w;
    int bits = 0, dim, idx;
    int range = aac_cb_info[cb].range;

    if(range == -1) return 0;
    cb--;
    dim = cb < FIRST_PAIR_BT ? 4 : 2;

    if(cb == ESC_BT){
        for(w = 0; w < group_len; w++){
        int coef_abs[2];
            for(i = 0; i < size; i += 2){
               idx = 0;
               for(j = 0; j < 2; j++){
                    coef_abs[j] = FFABS(sce->icoefs[start+i+j]);
                    idx = idx*17 + FFMIN(coef_abs[j], 16);
                    if(coef_abs[j] > 15){
                        bits += av_log2(coef_abs[j])*2 - 4 + 1;
                    }
                }
                bits += ff_aac_spectral_bits[cb][idx];
            }
            start += 128;
        }
    }else if(IS_CODEBOOK_UNSIGNED(cb)){
        for(w = 0; w < group_len; w++){
            for(i = 0; i < size; i += dim){
                idx = FFABS(sce->icoefs[start+i]);
                for(j = 1; j < dim; j++){
                    idx = idx * range + FFABS(sce->icoefs[start+i+j]);
                }
                bits += ff_aac_spectral_bits[cb][idx];
            }
            start += 128;
        }
    }else{
        for(w = 0; w < group_len; w++){
            for(i = 0; i < size; i += dim){
                idx = sce->icoefs[start+i];
                for(j = 1; j < dim; j++)
                    idx = idx * range + sce->icoefs[start+i+j];
                //it turned out that all signed codebooks use the same offset for index coding
                idx += 40;
                bits += ff_aac_spectral_bits[cb][idx];
            }
            start += 128;
        }
    }
    return bits;
}

/**
 * Encode band info for single window group bands.
 */
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len)
{
    BandCodingPath path[64];
    int band_bits[64][12];
    int w, swb, cb, start, start2, size;
    int i, j;
    const int max_sfb = sce->ics.max_sfb;
    const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
    const int run_esc = (1 << run_bits) - 1;
    int bits, sbits, idx, count;
    int stack[64], stack_len;

    start = win*128;
    for(swb = 0; swb < max_sfb; swb++){
        int maxval = 0;
        start2 = start;
        size = sce->ics.swb_sizes[swb];
        if(sce->zeroes[win*16 + swb]){
            maxval = 0;
        }else{
            for(w = 0; w < group_len; w++){
                for(i = start2; i < start2 + size; i++){
                    maxval = FFMAX(maxval, FFABS(sce->icoefs[i]));
                }
                start2 += 128;
            }
        }
        sbits = calculate_band_sign_bits(s, sce, group_len, start, size);
        for(cb = 0; cb < 12; cb++){
            if(aac_cb_info[cb].maxval < maxval){
                band_bits[swb][cb] = INT_MAX;
            }else{
                band_bits[swb][cb] = calculate_band_bits(s, sce, group_len, start, size, cb);
                if(IS_CODEBOOK_UNSIGNED(cb-1)){
                    band_bits[swb][cb] += sbits;
                }
            }
        }
        start += sce->ics.swb_sizes[swb];
    }
    path[0].bits = 0;
    for(i = 1; i <= max_sfb; i++)
        path[i].bits = INT_MAX;
    for(i = 0; i < max_sfb; i++){
        for(cb = 0; cb < 12; cb++){
            int sum = 0;
            for(j = 1; j <= max_sfb - i; j++){
                if(band_bits[i+j-1][cb] == INT_MAX)
                    break;
                sum += band_bits[i+j-1][cb];
                bits = sum + path[i].bits + run_value_bits[sce->ics.num_windows == 8][j];
                if(bits < path[i+j].bits){
                    path[i+j].bits     = bits;
                    path[i+j].codebook = cb;
                    path[i+j].prev_idx = i;
                }
            }
        }
    }
    assert(path[max_sfb].bits != INT_MAX);

    //convert resulting path from backward-linked list
    stack_len = 0;
    idx = max_sfb;
    while(idx > 0){
        stack[stack_len++] = idx;
        idx = path[idx].prev_idx;
    }

    //perform actual band info encoding
    start = 0;
    for(i = stack_len - 1; i >= 0; i--){
        put_bits(&s->pb, 4, path[stack[i]].codebook);
        count = stack[i] - path[stack[i]].prev_idx;
        memset(sce->zeroes + win*16 + start, !path[stack[i]].codebook, count);
        //XXX: memset when band_type is also uint8_t
        for(j = 0; j < count; j++){
            sce->band_type[win*16 + start] =  path[stack[i]].codebook;
            start++;
        }
        while(count >= run_esc){
            put_bits(&s->pb, run_bits, run_esc);
            count -= run_esc;
        }
        put_bits(&s->pb, run_bits, count);
    }
}

/**
 * Produce integer coefficients from scalefactors provided by the model.
 */
static void quantize_coeffs(AACEncContext *apc, ChannelElement *cpe, int chans)
{
    int i, w, w2, g, ch;
    int start, sum, maxsfb, cmaxsfb;

    for(ch = 0; ch < chans; ch++){
        IndividualChannelStream *ics = &cpe->ch[ch].ics;
        start = 0;
        maxsfb = 0;
        cpe->ch[ch].pulse.num_pulse = 0;
        for(w = 0; w < ics->num_windows*16; w += 16){
            for(g = 0; g < ics->num_swb; g++){
                sum = 0;
                //apply M/S
                if(!ch && cpe->ms_mask[w + g]){
                    for(i = 0; i < ics->swb_sizes[g]; i++){
                        cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
                        cpe->ch[1].coeffs[start+i] =  cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
                    }
                }
                if(!cpe->ch[ch].zeroes[w + g])
                    sum = quantize_band(cpe->ch[ch].coeffs + start,
                                        cpe->ch[ch].icoefs + start,
                                        ics->swb_sizes[g],
                                        cpe->ch[ch].sf_idx[w + g]);
                else
                    memset(cpe->ch[ch].icoefs + start, 0, ics->swb_sizes[g] * sizeof(cpe->ch[0].icoefs[0]));
                cpe->ch[ch].zeroes[w + g] = !sum;
                start += ics->swb_sizes[g];
            }
            for(cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--);
            maxsfb = FFMAX(maxsfb, cmaxsfb);
        }
        ics->max_sfb = maxsfb;

        //adjust zero bands for window groups
        for(w = 0; w < ics->num_windows; w += ics->group_len[w]){
            for(g = 0; g < ics->max_sfb; g++){
                i = 1;
                for(w2 = w; w2 < w + ics->group_len[w]; w2++){
                    if(!cpe->ch[ch].zeroes[w2*16 + g]){
                        i = 0;
                        break;
                    }
                }
                cpe->ch[ch].zeroes[w*16 + g] = i;
            }
        }
    }

    if(chans > 1 && cpe->common_window){
        IndividualChannelStream *ics0 = &cpe->ch[0].ics;
        IndividualChannelStream *ics1 = &cpe->ch[1].ics;
        int msc = 0;
        ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
        ics1->max_sfb = ics0->max_sfb;
        for(w = 0; w < ics0->num_windows*16; w += 16)
            for(i = 0; i < ics0->max_sfb; i++)
                if(cpe->ms_mask[w+i]) msc++;
        if(msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0;
        else cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
    }
}

typedef struct TrellisPath {
    float cost;
    int prev;
    int min_val;
    int max_val;
} TrellisPath;

static void search_for_quantizers(AACEncContext *s, ChannelElement *cpe, int channels)
{
    int q, ch, w, w2, g, start = 0;
    int i;
    int qcoeffs[128];
    int idx;
    TrellisPath paths[256*121];
    int bandaddr[121];
    const float lambda = 5e-7f;
    int minq;
    float mincost;

    for(ch = 0; ch < channels; ch++){
        SingleChannelElement *sce = &cpe->ch[ch];
        for(i = 0; i < 256; i++){
            paths[i].cost = 0.0f;
            paths[i].prev = -1;
            paths[i].min_val = i;
            paths[i].max_val = i;
        }
        for(i = 256; i < 256*121; i++){
            paths[i].cost = INFINITY;
            paths[i].prev = -2;
            paths[i].min_val = INT_MAX;
            paths[i].max_val = 0;
        }
        idx = 256;
        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;
                float qmin, qmax;
                int nz = 0;

                bandaddr[idx >> 8] = w*16+g;
                qmin = INT_MAX;
                qmax = 0.0f;
                for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
                    FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
                    if(band->energy <= band->threshold || band->threshold == 0.0f){
                        sce->zeroes[(w+w2)*16+g] = 1;
                        continue;
                    }
                    sce->zeroes[(w+w2)*16+g] = 0;
                    nz = 1;
                    for(i = 0; i < sce->ics.swb_sizes[g]; i++){
                        float t = fabsf(coefs[w2*128+i]);
                        if(t > 0.0f) qmin = fminf(qmin, t);
                        qmax = fmaxf(qmax, t);
                    }
                }
                if(nz){
                    int minscale, maxscale;
                    //minimum scalefactor index is when mininum nonzero coefficient after quantizing is not clipped
                    minscale = av_clip_uint8(log2(qmin)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
                    //maximum scalefactor index is when maximum coefficient after quantizing is still not zero
                    maxscale = av_clip_uint8(log2(qmax)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
                    for(q = minscale; q < maxscale; q++){
                        float dist = 0.0f;
                        int bits = 0, sum = 0;
                        for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
                            FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
                            if(sce->zeroes[(w+w2)*16+g])
                                continue;
                            sum  += quantize_band(coefs + w2*128, qcoeffs, sce->ics.swb_sizes[g], q);
                            dist += get_approximate_quant_error(coefs + w2*128, qcoeffs, sce->ics.swb_sizes[g], q) / band->threshold;
                            bits += get_approximate_bits(qcoeffs, sce->ics.swb_sizes[g]);
                        }
                        for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
                            float cost;
                            int minv, maxv;
                            if(isinf(paths[idx - 256 + i].cost))
                                continue;
                            cost = paths[idx - 256 + i].cost + dist * lambda + bits
                                   + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
                            minv = FFMIN(paths[idx - 256 + i].min_val, q);
                            maxv = FFMAX(paths[idx - 256 + i].max_val, q);
                            if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
                                paths[idx + q].cost = cost;
                                paths[idx + q].prev = idx - 256 + i;
                                paths[idx + q].min_val = minv;
                                paths[idx + q].max_val = maxv;
                            }
                        }
                    }
                }else{
                    for(q = 0; q < 256; q++){
                        if(!isinf(paths[idx - 256 + q].cost)){
                            paths[idx + q].cost = paths[idx - 256 + q].cost + 1;
                            paths[idx + q].prev = idx - 256 + q;
                            paths[idx + q].min_val = FFMIN(paths[idx - 256 + q].min_val, q);
                            paths[idx + q].max_val = FFMAX(paths[idx - 256 + q].max_val, q);
                            continue;
                        }
                        for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
                            float cost;
                            int minv, maxv;
                            if(isinf(paths[idx - 256 + i].cost))
                                continue;
                            cost = paths[idx - 256 + i].cost + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
                            minv = FFMIN(paths[idx - 256 + i].min_val, q);
                            maxv = FFMAX(paths[idx - 256 + i].max_val, q);
                            if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
                                paths[idx + q].cost = cost;
                                paths[idx + q].prev = idx - 256 + i;
                                paths[idx + q].min_val = minv;
                                paths[idx + q].max_val = maxv;
                            }
                        }
                    }
                }
                sce->zeroes[w*16+g] = !nz;
                start += sce->ics.swb_sizes[g];
                idx += 256;
            }
        }
        idx -= 256;
        mincost = paths[idx].cost;
        minq = idx;
        for(i = 1; i < 256; i++){
            if(paths[idx + i].cost < mincost){
                mincost = paths[idx + i].cost;
                minq = idx + i;
            }
        }
        while(minq >= 256){
            sce->sf_idx[bandaddr[minq>>8]] = minq & 0xFF;
            minq = paths[minq].prev;
        }
        //set the same quantizers inside window groups
        for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
            for(g = 0;  g < sce->ics.num_swb; g++)
                for(w2 = 1; w2 < sce->ics.group_len[w]; w2++)
                    sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
    }
}

/**
 * Encode the coefficients of one scalefactor band with selected codebook.
 */
static void encode_band_coeffs(AACEncContext *s, SingleChannelElement *sce,
                               int start, int size, int cb)
{
    const uint8_t  *bits  = ff_aac_spectral_bits [cb - 1];
    const uint16_t *codes = ff_aac_spectral_codes[cb - 1];
    const int range = aac_cb_info[cb].range;
    const int dim = (cb < FIRST_PAIR_BT) ? 4 : 2;
    int i, j, idx;

    //do not encode zero or special codebooks
    if(range == -1) return;

    if(cb == ESC_BT){
        int coef_abs[2];
        for(i = start; i < start + size; i += 2){
            idx = 0;
            for(j = 0; j < 2; j++){
                coef_abs[j] = FFABS(sce->icoefs[i+j]);
                idx = idx*17 + FFMIN(coef_abs[j], 16);
            }
            put_bits(&s->pb, bits[idx], codes[idx]);
            //output signs
            for(j = 0; j < 2; j++)
                if(sce->icoefs[i+j])
                    put_bits(&s->pb, 1, sce->icoefs[i+j] < 0);
            //output escape values
            for(j = 0; j < 2; j++){
                if(coef_abs[j] > 15){
                    int len = av_log2(coef_abs[j]);

                    put_bits(&s->pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
                    put_bits(&s->pb, len, coef_abs[j] & ((1 << len) - 1));
                }
            }
        }
    }else if(IS_CODEBOOK_UNSIGNED(cb)){
        for(i = start; i < start + size; i += dim){
            idx = FFABS(sce->icoefs[i]);
            for(j = 1; j < dim; j++)
                idx = idx * range + FFABS(sce->icoefs[i+j]);
            put_bits(&s->pb, bits[idx], codes[idx]);
            //output signs
            for(j = 0; j < dim; j++)
                if(sce->icoefs[i+j])
                    put_bits(&s->pb, 1, sce->icoefs[i+j] < 0);
        }
    }else{
        for(i = start; i < start + size; i += dim){
            idx = sce->icoefs[i];
            for(j = 1; j < dim; j++)
                idx = idx * range + sce->icoefs[i+j];
            //it turned out that all signed codebooks use the same offset for index coding
            idx += 40;
            put_bits(&s->pb, bits[idx], codes[idx]);
        }
    }
}

/**
 * Encode scalefactor band coding type.
 */
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
    int w;

    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
        encode_window_bands_info(s, sce, w, sce->ics.group_len[w]);
    }
}

/**
 * Encode scalefactors.
 */
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
{
    int off = sce->sf_idx[0], diff;
    int i, w;

    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
        for(i = 0; i < sce->ics.max_sfb; i++){
            if(!sce->zeroes[w*16 + i]){
                diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
                if(diff < 0 || diff > 120) av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
                off = sce->sf_idx[w*16 + i];
                put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
            }
        }
    }
}

/**
 * Encode pulse data.
 */
static void encode_pulses(AACEncContext *s, Pulse *pulse)
{
    int i;

    put_bits(&s->pb, 1, !!pulse->num_pulse);
    if(!pulse->num_pulse) return;

    put_bits(&s->pb, 2, pulse->num_pulse - 1);
    put_bits(&s->pb, 6, pulse->start);
    for(i = 0; i < pulse->num_pulse; i++){
        put_bits(&s->pb, 5, pulse->pos[i]);
        put_bits(&s->pb, 4, pulse->amp[i]);
    }
}

/**
 * Encode spectral coefficients processed by psychoacoustic model.
 */
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
    int start, i, w, w2;

    for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
        start = 0;
        for(i = 0; i < sce->ics.max_sfb; i++){
            if(sce->zeroes[w*16 + i]){
                start += sce->ics.swb_sizes[i];
                continue;
            }
            for(w2 = w; w2 < w + sce->ics.group_len[w]; w2++){
                encode_band_coeffs(s, sce, start + w2*128,
                                   sce->ics.swb_sizes[i],
                                   sce->band_type[w*16 + i]);
            }
            start += sce->ics.swb_sizes[i];
        }
    }
}

/**
 * Encode one channel of audio data.
 */
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
{
    put_bits(&s->pb, 8, sce->sf_idx[0]);
    if(!common_window) put_ics_info(s, &sce->ics);
    encode_band_info(s, sce);
    encode_scale_factors(avctx, s, sce);
    encode_pulses(s, &sce->pulse);
    put_bits(&s->pb, 1, 0); //tns
    put_bits(&s->pb, 1, 0); //ssr
    encode_spectral_coeffs(s, sce);
    return 0;
}

/**
 * Write some auxiliary information about the created AAC file.
 */
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
{
    int i, namelen, padbits;

    namelen = strlen(name) + 2;
    put_bits(&s->pb, 3, TYPE_FIL);
    put_bits(&s->pb, 4, FFMIN(namelen, 15));
    if(namelen >= 15)
        put_bits(&s->pb, 8, namelen - 16);
    put_bits(&s->pb, 4, 0); //extension type - filler
    padbits = 8 - (put_bits_count(&s->pb) & 7);
    align_put_bits(&s->pb);
    for(i = 0; i < namelen - 2; i++)
        put_bits(&s->pb, 8, name[i]);
    put_bits(&s->pb, 12 - padbits, 0);
}

static int aac_encode_frame(AVCodecContext *avctx,
                            uint8_t *frame, int buf_size, void *data)
{
    AACEncContext *s = avctx->priv_data;
    int16_t *samples = s->samples, *samples2, *la;
    ChannelElement *cpe;
    int i, j, chans, tag, start_ch;
    const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
    int chan_el_counter[4];

    if(s->last_frame)
        return 0;
    if(data){
        if(!s->psypp){
            memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
        }else{
            start_ch = 0;
            samples2 = s->samples + 1024 * avctx->channels;
            for(i = 0; i < chan_map[0]; i++){
                tag = chan_map[i+1];
                chans = tag == TYPE_CPE ? 2 : 1;
                ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch + i, chans);
                start_ch += chans;
            }
        }
    }
    if(!avctx->frame_number){
        memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
        return 0;
    }

    init_put_bits(&s->pb, frame, buf_size*8);
    if((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)){
        put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
    }
    start_ch = 0;
    memset(chan_el_counter, 0, sizeof(chan_el_counter));
    for(i = 0; i < chan_map[0]; i++){
        FFPsyWindowInfo wi[2];
        tag = chan_map[i+1];
        chans = tag == TYPE_CPE ? 2 : 1;
        cpe = &s->cpe[i];
        samples2 = samples + start_ch;
        la = samples2 + 1024 * avctx->channels + start_ch;
        if(!data) la = NULL;
        for(j = 0; j < chans; j++){
            IndividualChannelStream *ics = &cpe->ch[j].ics;
            int k;
            wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
            ics->window_sequence[1] = ics->window_sequence[0];
            ics->window_sequence[0] = wi[j].window_type[0];
            ics->use_kb_window[1]   = ics->use_kb_window[0];
            ics->use_kb_window[0]   = wi[j].window_shape;
            ics->num_windows        = wi[j].num_windows;
            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
            ics->num_swb            = s->psy.num_bands[ics->num_windows == 8];
            for(k = 0; k < ics->num_windows; k++)
                ics->group_len[k] = wi[j].grouping[k];

            apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
        }
        cpe->common_window = 0;
        if(chans > 1
            && wi[0].window_type[0] == wi[1].window_type[0]
            && wi[0].window_shape   == wi[1].window_shape){

            cpe->common_window = 1;
            for(j = 0; j < wi[0].num_windows; j++){
                if(wi[0].grouping[j] != wi[1].grouping[j]){
                    cpe->common_window = 0;
                    break;
                }
            }
        }
        search_for_quantizers(s, cpe, chans);
        quantize_coeffs(s, cpe, chans);
        put_bits(&s->pb, 3, tag);
        put_bits(&s->pb, 4, chan_el_counter[tag]++);
        if(chans == 2){
            put_bits(&s->pb, 1, cpe->common_window);
            if(cpe->common_window){
                put_ics_info(s, &cpe->ch[0].ics);
                encode_ms_info(&s->pb, cpe);
            }
        }
        for(j = 0; j < chans; j++){
            s->cur_channel = start_ch + j;
            ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
            encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
        }
        start_ch += chans;
    }

    put_bits(&s->pb, 3, TYPE_END);
    flush_put_bits(&s->pb);
    avctx->frame_bits = put_bits_count(&s->pb);

    if(!data)
        s->last_frame = 1;
    memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
    return put_bits_count(&s->pb)>>3;
}

static av_cold int aac_encode_end(AVCodecContext *avctx)
{
    AACEncContext *s = avctx->priv_data;

    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
    ff_psy_end(&s->psy);
    ff_psy_preprocess_end(s->psypp);
    av_freep(&s->samples);
    av_freep(&s->cpe);
    return 0;
}

AVCodec aac_encoder = {
    "aac",
    CODEC_TYPE_AUDIO,
    CODEC_ID_AAC,
    sizeof(AACEncContext),
    aac_encode_init,
    aac_encode_frame,
    aac_encode_end,
    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};
-------------- next part --------------
/*
 * AAC encoder psychoacoustic model
 * Copyright (C) 2008 Konstantin Shishkov
 *
 * 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 aacpsy.c
 * AAC encoder psychoacoustic model
 */

#include "avcodec.h"
#include "aactab.h"
#include "psymodel.h"

/***********************************
 *              TODOs:
 * thresholds linearization after their modifications for attaining given bitrate
 * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
 * control quality for quality-based output
 **********************************/

/**
 * constants for 3GPP AAC psychoacoustic model
 * @{
 */
#define PSY_3GPP_SPREAD_LOW  1.5f // spreading factor for ascending threshold spreading  (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI   3.0f // spreading factor for descending threshold spreading (30 dB/Bark)

#define PSY_3GPP_RPEMIN      0.01f
#define PSY_3GPP_RPELEV      2.0f
/**
 * @}
 */

/**
 * information for single band used by 3GPP TS26.403-inspired psychoacoustic model
 */
typedef struct Psy3gppBand{
    float energy;    ///< band energy
    float ffac;      ///< form factor
    float thr;       ///< energy threshold
    float min_snr;   ///< minimal SNR
    float thr_quiet; ///< threshold in quiet
}Psy3gppBand;

/**
 * single/pair channel context for psychoacoustic model
 */
typedef struct Psy3gppChannel{
    Psy3gppBand band[128];               ///< bands information
    Psy3gppBand prev_band[128];          ///< bands information from the previous frame

    float       win_energy;              ///< sliding average of channel energy
    float       iir_state[2];            ///< hi-pass IIR filter state
    uint8_t     next_grouping;           ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
    enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
}Psy3gppChannel;

/**
 * psychoacoustic model frame type-dependent coefficients
 */
typedef struct Psy3gppCoeffs{
    float ath       [64]; ///< absolute threshold of hearing per bands
    float barks     [64]; ///< Bark value for each spectral band in long frame
    float spread_low[64]; ///< spreading factor for low-to-high threshold spreading in long frame
    float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
}Psy3gppCoeffs;

/**
 * 3GPP TS26.403-inspired psychoacoustic model specific data
 */
typedef struct Psy3gppContext{
    Psy3gppCoeffs psy_coef[2];
    Psy3gppChannel *ch;
}Psy3gppContext;

/**
 * Calculate Bark value for given line.
 */
static av_cold float calc_bark(float f)
{
    return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}

#define ATH_ADD 4
/**
 * Calculate ATH value for given frequency.
 * Borrowed from Lame.
 */
static av_cold float ath(float f, float add)
{
    f /= 1000.0f;
    return   3.64 * pow(f, -0.8)
            - 6.8  * exp(-0.6  * (f - 3.4) * (f - 3.4))
            + 6.0  * exp(-0.15 * (f - 8.7) * (f - 8.7))
            + (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
}

static av_cold int psy_3gpp_init(FFPsyContext *ctx){
    Psy3gppContext *pctx;
    float barks[1024];
    int i, j, g, start;
    float prev, minscale, minath;

    ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
    pctx = (Psy3gppContext*) ctx->model_priv_data;

    for(i = 0; i < 1024; i++)
        barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
    minath = ath(3410, ATH_ADD);
    for(j = 0; j < 2; j++){
        Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
        i = 0;
        prev = 0.0;
        for(g = 0; g < ctx->num_bands[j]; g++){
            i += ctx->bands[j][g];
            coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
            prev = barks[i - 1];
        }
        for(g = 0; g < ctx->num_bands[j] - 1; g++){
            coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
            coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
        }
        start = 0;
        for(g = 0; g < ctx->num_bands[j]; g++){
            minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
            for(i = 1; i < ctx->bands[j][g]; i++){
                minscale = fminf(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
            }
            coeffs->ath[g] = minscale - minath;
            start += ctx->bands[j][g];
        }
    }

    pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
    return 0;
}

/**
 * IIR filter used in block switching decision
 */
static float iir_filter(int in, float state[2])
{
    float ret;

    ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
    state[0] = in;
    state[1] = ret;
    return ret;
}

/**
 * window grouping information stored as bits (0 - new group, 1 - group continues)
 */
static const uint8_t window_grouping[9] = {
    0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
};

/**
 * Tell encoder which window types to use.
 * @see 3GPP TS26.403 5.4.1 "Blockswitching"
 */
static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
                                       const int16_t *audio, const int16_t *la,
                                       int channel, int prev_type)
{
    int i, j;
    int br = ctx->avctx->bit_rate / ctx->avctx->channels;
    int attack_ratio = br <= 16000 ? 18 : 10;
    Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
    Psy3gppChannel *pch = &pctx->ch[channel];
    uint8_t grouping = 0;
    FFPsyWindowInfo wi;

    memset(&wi, 0, sizeof(wi));
    if(la){
        float s[8], v;
        int switch_to_eight = 0;
        float sum = 0.0, sum2 = 0.0;
        int attack_n = 0;
        for(i = 0; i < 8; i++){
            for(j = 0; j < 128; j++){
                v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
                sum += v*v;
            }
            s[i] = sum;
            sum2 += sum;
        }
        for(i = 0; i < 8; i++){
            if(s[i] > pch->win_energy * attack_ratio){
                attack_n = i + 1;
                switch_to_eight = 1;
                break;
            }
        }
        pch->win_energy = pch->win_energy*7/8 + sum2/64;

        wi.window_type[1] = prev_type;
        switch(prev_type){
        case ONLY_LONG_SEQUENCE:
            wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
            break;
        case LONG_START_SEQUENCE:
            wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
            grouping = pch->next_grouping;
            break;
        case LONG_STOP_SEQUENCE:
            wi.window_type[0] = ONLY_LONG_SEQUENCE;
            break;
        case EIGHT_SHORT_SEQUENCE:
            wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
            grouping = switch_to_eight ? pch->next_grouping : 0;
            break;
        }
        pch->next_grouping = window_grouping[attack_n];
    }else{
        for(i = 0; i < 3; i++)
            wi.window_type[i] = prev_type;
        grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
    }

    wi.window_shape   = 1;
    if(wi.window_type[0] != EIGHT_SHORT_SEQUENCE){
        wi.num_windows = 1;
        wi.grouping[0] = 1;
    }else{
        int lastgrp = 0;
        wi.num_windows = 8;
        for(i = 0; i < 8; i++){
            if(!((grouping >> i) & 1))
                lastgrp = i;
            wi.grouping[lastgrp]++;
        }
    }

    return wi;
}

/**
 * Calculate band thresholds as suggested in 3GPP TS26.403
 */
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
                             FFPsyWindowInfo *wi)
{
    Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
    Psy3gppChannel *pch = &pctx->ch[channel];
    int start = 0;
    int i, w, g;
    const int num_bands = ctx->num_bands[wi->num_windows == 8];
    const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
    Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];

    //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
    for(w = 0; w < wi->num_windows*16; w += 16){
        for(g = 0; g < num_bands; g++){
            Psy3gppBand *band = &pch->band[w+g];
            for(i = 0; i < band_sizes[g]; i++)
                band->energy += coefs[start+i] * coefs[start+i];
            band->energy *= 1.0f / (512*512);
            band->thr = band->energy * 0.001258925f;
            start += band_sizes[g];

            ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
        }
    }
    //modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
    for(w = 0; w < wi->num_windows*16; w += 16){
        Psy3gppBand *band = &pch->band[w];
        for(g = 1; g < num_bands; g++){
            band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
        }
        for(g = num_bands - 2; g >= 0; g--){
            band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g+1]);
        }
        for(g = 0; g < num_bands; g++){
            band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
            if(wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE){
                band[g].thr_quiet = fmaxf(PSY_3GPP_RPEMIN*band[g].thr_quiet,
                                          fminf(band[g].thr_quiet,
                                          PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
            }
            band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);

            ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
        }
    }
    memcpy(pch->prev_band, pch->band, sizeof(pch->band));
}

static av_cold void psy_3gpp_end(FFPsyContext *apc)
{
    Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
    av_freep(&pctx->ch);
    av_freep(&apc->model_priv_data);
}


const FFPsyModel ff_aac_psy_model =
{
    .name    = "3GPP TS 26.403-inspired model",
    .init    = psy_3gpp_init,
    .window  = psy_3gpp_window,
    .analyze = psy_3gpp_analyze,
    .end     = psy_3gpp_end,
};
-------------- next part --------------
/*
 * audio encoder psychoacoustic model
 * Copyright (C) 2008 Konstantin Shishkov
 *
 * 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
 */

#ifndef AVCODEC_PSYMODEL_H
#define AVCODEC_PSYMODEL_H

#include "avcodec.h"

/** maximum possible number of bands */
#define PSY_MAX_BANDS 128

/**
 * single band psychoacoustic information
 */
typedef struct FFPsyBand{
    int   bits;
    float energy;
    float threshold;
    float distortion;
    float perceptual_weight;
}FFPsyBand;

/**
 * windowing related information
 */
typedef struct FFPsyWindowInfo{
    int window_type[3];               ///< window type (short/long/transitional, etc.) - current, previous and next
    int window_shape;                 ///< window shape (sine/KBD/whatever)
    int num_windows;                  ///< number of windows in a frame
    int grouping[8];                  ///< window grouping (for e.g. AAC)
    int *window_sizes;                ///< sequence of window sizes inside one frame (for eg. WMA)
}FFPsyWindowInfo;

/**
 * context used by psychoacoustic model
 */
typedef struct FFPsyContext{
    AVCodecContext *avctx;            ///< encoder context
    const struct FFPsyModel *model;   ///< encoder-specific model functions

    FFPsyBand *psy_bands;             ///< frame bands information

    uint8_t **bands;                  ///< scalefactor band sizes for possible frame sizes
    int     *num_bands;               ///< number of scalefactor bands for possible frame sizes
    int num_lens;                     ///< number of scalefactor band sets

    void* model_priv_data;            ///< psychoacoustic model implementation private data
}FFPsyContext;

/**
 * codec-specific psychoacoustic model implementation
 */
typedef struct FFPsyModel {
    const char *name;
    int  (*init)   (FFPsyContext *apc);
    FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
    void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi);
    void (*end)    (FFPsyContext *apc);
}FFPsyModel;

/**
 * Initialize psychoacoustic model.
 *
 * @param ctx        model context
 * @param avctx      codec context
 * @param num_lens   number of possible frame lengths
 * @param bands      scalefactor band lengths for all frame lengths
 * @param num_bands  number of scalefactor bands for all frame lengths
 *
 * @return zero if successful, a negative value if not
 */
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
                        int num_lens,
                        uint8_t **bands, int* num_bands);

/**
 * Suggest window sequence for channel.
 *
 * @param ctx       model context
 * @param audio     samples for the current frame
 * @param la        lookahead samples (NULL when unavailable)
 * @param channel   number of channel element to analyze
 * @param prev_type previous window type
 *
 * @return suggested window information in a structure
 */
FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
                                      const int16_t *audio, const int16_t *la,
                                      int channel, int prev_type);


/**
 * Perform psychoacoustic analysis and set band info (threshold, energy).
 *
 * @param ctx     model context
 * @param channel audio channel number
 * @param coeffs  pointer to the transformed coefficients
 * @param wi      window information
 */
void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs,
                          FFPsyWindowInfo *wi);

/**
 * Cleanup model context at the end.
 *
 * @param ctx model context
 */
av_cold void ff_psy_end(FFPsyContext *ctx);


/**************************************************************************
 *                       Audio preprocessing stuff.                       *
 *       This should be moved into some audio filter eventually.          *
 **************************************************************************/
struct FFPsyPreprocessContext;

/**
 * psychoacoustic model audio preprocessing initialization
 */
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);

/**
 * Preprocess several channel in audio frame in order to compress it better.
 *
 * @param ctx      preprocessing context
 * @param audio    samples to preprocess
 * @param dest     place to put filtered samples
 * @param tag      channel number
 * @param channels number of channel to preprocess (some additional work may be done on stereo pair)
 */
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
                       const int16_t *audio, int16_t *dest,
                       int tag, int channels);

/**
 * Cleanup audio preprocessing module.
 */
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);

#endif /* AVCODEC_PSYMODEL_H */
-------------- next part --------------
/*
 * audio encoder psychoacoustic model
 * Copyright (C) 2008 Konstantin Shishkov
 *
 * 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
 */

#include "avcodec.h"
#include "psymodel.h"
#include "iirfilter.h"

extern const FFPsyModel ff_aac_psy_model;

av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
                        int num_lens,
                        uint8_t **bands, int* num_bands)
{
    ctx->avctx = avctx;
    ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
    ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
    ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
    memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
    memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
    switch(ctx->avctx->codec_id){
    case CODEC_ID_AAC:
        ctx->model = &ff_aac_psy_model;
        break;
    }
    if(ctx->model->init)
        return ctx->model->init(ctx);
    return 0;
}

FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
                                      const int16_t *audio, const int16_t *la,
                                      int channel, int prev_type)
{
    return ctx->model->window(ctx, audio, la, channel, prev_type);
}

void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
                          const float *coeffs, FFPsyWindowInfo *wi)
{
    ctx->model->analyze(ctx, channel, coeffs, wi);
}

av_cold void ff_psy_end(FFPsyContext *ctx)
{
    if(ctx->model->end)
        ctx->model->end(ctx);
    av_freep(&ctx->bands);
    av_freep(&ctx->num_bands);
    av_freep(&ctx->psy_bands);
}

typedef struct FFPsyPreprocessContext{
    AVCodecContext *avctx;
    float stereo_att;
    struct FFIIRFilterCoeffs *fcoeffs;
    struct FFIIRFilterState **fstate;
}FFPsyPreprocessContext;

#define FILT_ORDER 4

av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
{
    FFPsyPreprocessContext *ctx;
    int i;
    ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
    ctx->avctx = avctx;
    ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
                                           FILT_ORDER, 0.25, 0.0, 0.0);
    if(ctx->fcoeffs){
        ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
        for(i = 0; i < avctx->channels; i++)
            ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
    }
    return ctx;
}

void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
                       const int16_t *audio, int16_t *dest,
                       int tag, int channels)
{
    int ch, i;
    if(ctx->fstate){
        for(ch = 0; ch < channels; ch++){
            ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
                          audio + ch, ctx->avctx->channels,
                          dest  + ch, ctx->avctx->channels);
        }
    }else{
        for(ch = 0; ch < channels; ch++){
            for(i = 0; i < ctx->avctx->frame_size; i++)
                dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
        }
    }
}

av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
{
    int i;
    ff_iir_filter_free_coeffs(ctx->fcoeffs);
    for(i = 0; i < ctx->avctx->channels; i++){
        ff_iir_filter_free_state(ctx->fstate[i]);
    }
    av_freep(&ctx->fstate);
}




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