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/*

 * 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

 * AAC encoder

 */

/***********************************

 *              TODOs:

 * add sane pulse detection

 * add temporal noise shaping

 ***********************************/

#include "libavutil/float_dsp.h"

#include "libavutil/opt.h"

#include "avcodec.h"

#include "put_bits.h"

#include "internal.h"

#include "mpeg4audio.h"

#include "kbdwin.h"

#include "sinewin.h"

#include "aac.h"

#include "aactab.h"

#include "aacenc.h"

#include "psymodel.h"

#define AAC_MAX_CHANNELS 6

#define ERROR_IF(cond, ...) \

    if (cond) { \

        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \

        return AVERROR(EINVAL); \

    }

float ff_aac_pow34sf_tab[428];

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

};

/** 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

};

/**

 * Table to remap channels from libavcodec's default order to AAC order.

 */

static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {

    { 0 },

    { 0, 1 },

    { 2, 0, 1 },

    { 2, 0, 1, 3 },

    { 2, 0, 1, 3, 4 },

    { 2, 0, 1, 4, 5, 3 },

};

/**

 * 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, s->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

    //Explicitly Mark SBR absent

    put_bits(&pb, 11, 0x2b7); //sync extension

    put_bits(&pb, 5,  AOT_SBR);

    put_bits(&pb, 1,  0);

    flush_put_bits(&pb);

}

#define WINDOW_FUNC(type) \

static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \

                                    SingleChannelElement *sce, \

                                    const float *audio)

WINDOW_FUNC(only_long)

{

    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;

    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

    float *out = sce->ret_buf;

    fdsp->vector_fmul        (out,        audio,        lwindow, 1024);

    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);

}

WINDOW_FUNC(long_start)

{

    const float *lwindow = sce->ics.use_kb_window[1] ? 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;

    float *out = sce->ret_buf;

    fdsp->vector_fmul(out, audio, lwindow, 1024);

    memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);

    fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);

    memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);

}

WINDOW_FUNC(long_stop)

{

    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[1] ? ff_aac_kbd_short_128 : ff_sine_128;

    float *out = sce->ret_buf;

    memset(out, 0, sizeof(out[0]) * 448);

    fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);

    memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);

    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);

}

WINDOW_FUNC(eight_short)

{

    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;

    const float *in = audio + 448;

    float *out = sce->ret_buf;

    int w;

    for (w = 0; w < 8; w++) {

        fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);

        out += 128;

        in  += 128;

        fdsp->vector_fmul_reverse(out, in, swindow, 128);

        out += 128;

    }

}

static void (*const apply_window[4])(AVFloatDSPContext *fdsp,

                                     SingleChannelElement *sce,

                                     const float *audio) = {

    [ONLY_LONG_SEQUENCE]   = apply_only_long_window,

    [LONG_START_SEQUENCE]  = apply_long_start_window,

    [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,

    [LONG_STOP_SEQUENCE]   = apply_long_stop_window

};

static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,

                                  float *audio)

{

    int i;

    float *output = sce->ret_buf;

    apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)

        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);

    else

        for (i = 0; i < 1024; i += 128)

            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);

    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);

}

/**

 * 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]);

}

/**

 * Produce integer coefficients from scalefactors provided by the model.

 */

static void adjust_frame_information(ChannelElement *cpe, int chans)

{

    int i, w, w2, g, ch;

    int start, 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++) {

                //apply M/S

                if (cpe->common_window && !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];

                    }

                }

                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 * ics0->num_windows ? 1 : 2;

    }

}

/**

 * 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])

        s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);

}

/**

 * 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;

                av_assert0(diff >= 0 && diff <= 120);

                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++)

                s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,

                                                   sce->ics.swb_sizes[i],

                                                   sce->sf_idx[w*16 + i],

                                                   sce->band_type[w*16 + i],

                                                   s->lambda);

            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(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 - 14);

    put_bits(&s->pb, 4, 0); //extension type - filler

    padbits = -put_bits_count(&s->pb) & 7;

    avpriv_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);

}

/*

 * Copy input samples.

 * Channels are reordered from libavcodec's default order to AAC order.

 */

static void copy_input_samples(AACEncContext *s, const AVFrame *frame)

{

    int ch;

    int end = 2048 + (frame ? frame->nb_samples : 0);

    const uint8_t *channel_map = aac_chan_maps[s->channels - 1];

    /* copy and remap input samples */

    for (ch = 0; ch < s->channels; ch++) {

        /* copy last 1024 samples of previous frame to the start of the current frame */

        memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));

        /* copy new samples and zero any remaining samples */

        if (frame) {

            memcpy(&s->planar_samples[ch][2048],

                   frame->extended_data[channel_map[ch]],

                   frame->nb_samples * sizeof(s->planar_samples[0][0]));

        }

        memset(&s->planar_samples[ch][end], 0,

               (3072 - end) * sizeof(s->planar_samples[0][0]));

    }

}

static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

                            const AVFrame *frame, int *got_packet_ptr)

{

    AACEncContext *s = avctx->priv_data;

    float **samples = s->planar_samples, *samples2, *la, *overlap;

    ChannelElement *cpe;

    int i, ch, w, g, chans, tag, start_ch, ret;

    int chan_el_counter[4];

    FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

    if (s->last_frame == 2)

        return 0;

    /* add current frame to queue */

    if (frame) {

        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)

            return ret;

    }

    copy_input_samples(s, frame);

    if (s->psypp)

        ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

    if (!avctx->frame_number)

        return 0;

    start_ch = 0;

    for (i = 0; i < s->chan_map[0]; i++) {

        FFPsyWindowInfo* wi = windows + start_ch;

        tag      = s->chan_map[i+1];

        chans    = tag == TYPE_CPE ? 2 : 1;

        cpe      = &s->cpe[i];

        for (ch = 0; ch < chans; ch++) {

            IndividualChannelStream *ics = &cpe->ch[ch].ics;

            int cur_channel = start_ch + ch;

            overlap  = &samples[cur_channel][0];

            samples2 = overlap + 1024;

            la       = samples2 + (448+64);

            if (!frame)

                la = NULL;

            if (tag == TYPE_LFE) {

                wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;

                wi[ch].window_shape   = 0;

                wi[ch].num_windows    = 1;

                wi[ch].grouping[0]    = 1;

                /* Only the lowest 12 coefficients are used in a LFE channel.

                 * The expression below results in only the bottom 8 coefficients

                 * being used for 11.025kHz to 16kHz sample rates.

                 */

                ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;

            } else {

                wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,

                                              ics->window_sequence[0]);

            }

            ics->window_sequence[1] = ics->window_sequence[0];

            ics->window_sequence[0] = wi[ch].window_type[0];

            ics->use_kb_window[1]   = ics->use_kb_window[0];

            ics->use_kb_window[0]   = wi[ch].window_shape;

            ics->num_windows        = wi[ch].num_windows;

            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];

            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];

            for (w = 0; w < ics->num_windows; w++)

                ics->group_len[w] = wi[ch].grouping[w];

            apply_window_and_mdct(s, &cpe->ch[ch], overlap);

        }

        start_ch += chans;

    }

    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)

        return ret;

    do {

        int frame_bits;

        init_put_bits(&s->pb, avpkt->data, avpkt->size);

        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))

            put_bitstream_info(s, LIBAVCODEC_IDENT);

        start_ch = 0;

        memset(chan_el_counter, 0, sizeof(chan_el_counter));

        for (i = 0; i < s->chan_map[0]; i++) {

            FFPsyWindowInfo* wi = windows + start_ch;

            const float *coeffs[2];

            tag      = s->chan_map[i+1];

            chans    = tag == TYPE_CPE ? 2 : 1;

            cpe      = &s->cpe[i];

            put_bits(&s->pb, 3, tag);

            put_bits(&s->pb, 4, chan_el_counter[tag]++);

            for (ch = 0; ch < chans; ch++)

                coeffs[ch] = cpe->ch[ch].coeffs;

            s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);

            for (ch = 0; ch < chans; ch++) {

                s->cur_channel = start_ch + ch;

                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);

            }

            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 (w = 0; w < wi[0].num_windows; w++) {

                    if (wi[0].grouping[w] != wi[1].grouping[w]) {

                        cpe->common_window = 0;

                        break;

                    }

                }

            }

            s->cur_channel = start_ch;

            if (s->options.stereo_mode && cpe->common_window) {

                if (s->options.stereo_mode > 0) {

                    IndividualChannelStream *ics = &cpe->ch[0].ics;

                    for (w = 0; w < ics->num_windows; w += ics->group_len[w])

                        for (g = 0;  g < ics->num_swb; g++)

                            cpe->ms_mask[w*16+g] = 1;

                } else if (s->coder->search_for_ms) {

                    s->coder->search_for_ms(s, cpe, s->lambda);

                }

            }

            adjust_frame_information(cpe, chans);

            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 (ch = 0; ch < chans; ch++) {

                s->cur_channel = start_ch + ch;

                encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);

            }

            start_ch += chans;

        }

        frame_bits = put_bits_count(&s->pb);

        if (frame_bits <= 6144 * s->channels - 3) {

            s->psy.bitres.bits = frame_bits / s->channels;

            break;

        }

        s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;

    } while (1);

    put_bits(&s->pb, 3, TYPE_END);

    flush_put_bits(&s->pb);

    avctx->frame_bits = put_bits_count(&s->pb);

    // rate control stuff

    if (!(avctx->flags & CODEC_FLAG_QSCALE)) {

        float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;

        s->lambda *= ratio;

        s->lambda = FFMIN(s->lambda, 65536.f);

    }

    if (!frame)

        s->last_frame++;

    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,

                       &avpkt->duration);

    avpkt->size = put_bits_count(&s->pb) >> 3;

    *got_packet_ptr = 1;

    return 0;

}

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);

    if (s->psypp)

        ff_psy_preprocess_end(s->psypp);

    av_freep(&s->buffer.samples);

    av_freep(&s->cpe);

    ff_af_queue_close(&s->afq);

    return 0;

}

static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)

{

    int ret = 0;

    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

    // 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_init_ff_sine_windows(10);

    ff_init_ff_sine_windows(7);

    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))

        return ret;

    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))

        return ret;

    return 0;

}

static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)

{

    int ch;

    FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);

    FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);

    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

    for(ch = 0; ch < s->channels; ch++)

        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

    return 0;

alloc_fail:

    return AVERROR(ENOMEM);

}

static av_cold int aac_encode_init(AVCodecContext *avctx)

{

    AACEncContext *s = avctx->priv_data;

    int i, ret = 0;

    const uint8_t *sizes[2];

    uint8_t grouping[AAC_MAX_CHANNELS];

    int lengths[2];

    avctx->frame_size = 1024;

    for (i = 0; i < 16; i++)

        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])

            break;

    s->channels = avctx->channels;

    ERROR_IF(i == 16,

             "Unsupported sample rate %d\n", avctx->sample_rate);

    ERROR_IF(s->channels > AAC_MAX_CHANNELS,

             "Unsupported number of channels: %d\n", s->channels);

    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,

             "Unsupported profile %d\n", avctx->profile);

    ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,

             "Too many bits per frame requested\n");

    s->samplerate_index = i;

    s->chan_map = aac_chan_configs[s->channels-1];

    if (ret = dsp_init(avctx, s))

        goto fail;

    if (ret = alloc_buffers(avctx, s))

        goto fail;

    avctx->extradata_size = 5;

    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];

    for (i = 0; i < s->chan_map[0]; i++)

        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;

    if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))

        goto fail;

    s->psypp = ff_psy_preprocess_init(avctx);

    s->coder = &ff_aac_coders[s->options.aac_coder];

    if (HAVE_MIPSDSPR1)

        ff_aac_coder_init_mips(s);

    s->lambda = avctx->global_quality ? avctx->global_quality : 120;

    ff_aac_tableinit();

    for (i = 0; i < 428; i++)

        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));

    avctx->delay = 1024;

    ff_af_queue_init(avctx, &s->afq);

    return 0;

fail:

    aac_encode_end(avctx);

    return ret;

}

#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM

static const AVOption aacenc_options[] = {

    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},

        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},

        {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

        {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

        {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

    {NULL}

};

static const AVClass aacenc_class = {

    "AAC encoder",

    av_default_item_name,

    aacenc_options,

    LIBAVUTIL_VERSION_INT,

};

/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build

 * failures */

static const int mpeg4audio_sample_rates[16] = {

    96000, 88200, 64000, 48000, 44100, 32000,

    24000, 22050, 16000, 12000, 11025, 8000, 7350

};

AVCodec ff_aac_encoder = {

    .name           = "aac",

    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = AV_CODEC_ID_AAC,

    .priv_data_size = sizeof(AACEncContext),

    .init           = aac_encode_init,

    .encode2        = aac_encode_frame,

    .close          = aac_encode_end,

    .supported_samplerates = mpeg4audio_sample_rates,

    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |

                      CODEC_CAP_EXPERIMENTAL,

    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,

                                                     AV_SAMPLE_FMT_NONE },

    .priv_class     = &aacenc_class,

};