view aacenc.c @ 10061:09f2db2d7c90 libavcodec

Fix bug caused by difference in stride and picture width. When a frame is allocated using libschroedinger routines, the frame data size does not match the actual frame size if the width is not a multiple of 16. So we cannot do a straightforward memcpy of the frame returned by libschroedinger into the FFmpeg picture as the stride differs from the width. Fix this bug by allocating for the libschroedinger frame with the dimensions in AVCodecContext within libavcodec and passing the frame to libschroedinger. patch by Anuradha Suraparaju, anuradha rd.bbc.co uk

/*
 * 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 libavcodec/aacenc.c
 * 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;
    }
    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_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);
    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->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;
                    }
                }
            }
            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",
    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"),
};