PeerStreamer

/*

 * 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 "avcodec.h"

#include "put_bits.h"

#include "dsputil.h"

#include "mpeg4audio.h"

#include "aac.h"

#include "aactab.h"

#include "aacenc.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

};

/** 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

};

/**

 * 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;

    }

    if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {

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

        return -1;

    }

    if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {

        av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");

        return -1;

    }

    s->samplerate_index = i;

    dsputil_init(&s->dsp, avctx);

    ff_mdct_init(&s->mdct1024, 11, 0, 1.0);

    ff_mdct_init(&s->mdct128,   8, 0, 1.0);

    // 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_init_ff_sine_windows(10);

    ff_init_ff_sine_windows(7);

    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);

    s->coder = &ff_aac_coders[0];

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

#if !CONFIG_HARDCODED_TABLES

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

        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);

#endif /* CONFIG_HARDCODED_TABLES */

    if (avctx->channels > 5)

        av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "

               "The output will most likely be an illegal bitstream.\n");

    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]);

}

/**

 * Produce integer coefficients from scalefactors provided by the model.

 */

static void adjust_frame_information(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];

                    }

                }

                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;

    }

}

/**

 * 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;

                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++)

                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(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];

    FFPsyWindowInfo windows[avctx->channels];

    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, 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;

    }

    start_ch = 0;

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

        FFPsyWindowInfo* wi = windows + start_ch;

        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];

            s->cur_channel = start_ch + j;

            apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);

        }

        start_ch += chans;

    }

    do {

        int frame_bits;

        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 = windows + start_ch;

            tag      = chan_map[i+1];

            chans    = tag == TYPE_CPE ? 2 : 1;

            cpe      = &s->cpe[i];

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

                s->cur_channel = start_ch + j;

                s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], 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 (j = 0; j < wi[0].num_windows; j++) {

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

                        cpe->common_window = 0;

                        break;

                    }

                }

            }

            s->cur_channel = start_ch;

            if (cpe->common_window && s->coder->search_for_ms)

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

            adjust_frame_information(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;

        }

        frame_bits = put_bits_count(&s->pb);

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

            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 (!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",

    AVMEDIA_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 = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

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

};