/*

 * APAC audio decoder

 *

 * 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 "libavutil/audio_fifo.h"

#include "libavutil/mem.h"

#include "avcodec.h"

#include "codec_internal.h"

#include "decode.h"

#include "get_bits.h"

typedef struct ChContext {

    int have_code;

    int last_sample;

    int last_delta;

    int bit_length;

    int block_length;

    uint8_t block[32 * 2];

    AVAudioFifo *samples;

} ChContext;

typedef struct APACContext {

    GetBitContext gb;

    int skip;

    int cur_ch;

    ChContext ch[2];

    uint8_t *bitstream;

    int64_t max_framesize;

    int bitstream_size;

    int bitstream_index;

} APACContext;

static av_cold int apac_close(AVCodecContext *avctx)

{

    APACContext *s = avctx->priv_data;

    av_freep(&s->bitstream);

    s->bitstream_size = 0;

    for (int ch = 0; ch < 2; ch++) {

        ChContext *c = &s->ch[ch];

        av_audio_fifo_free(c->samples);

    }

    return 0;

}

static av_cold int apac_init(AVCodecContext *avctx)

{

    APACContext *s = avctx->priv_data;

    if (avctx->bits_per_coded_sample > 8)

        avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

    else

        avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

    if (avctx->ch_layout.nb_channels < 1 ||

        avctx->ch_layout.nb_channels > 2 ||

        avctx->bits_per_coded_sample < 8 ||

        avctx->bits_per_coded_sample > 16

    )

        return AVERROR_INVALIDDATA;

    for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {

        ChContext *c = &s->ch[ch];

        c->bit_length = avctx->bits_per_coded_sample;

        c->block_length = 8;

        c->have_code = 0;

        c->samples = av_audio_fifo_alloc(avctx->sample_fmt, 1, 1024);

        if (!c->samples)

            return AVERROR(ENOMEM);

    }

    s->max_framesize = 1024;

    s->bitstream = av_realloc_f(s->bitstream, s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));

    if (!s->bitstream)

        return AVERROR(ENOMEM);

    return 0;

}

static int get_code(ChContext *c, GetBitContext *gb)

{

    if (get_bits1(gb)) {

        int code = get_bits(gb, 2);

        switch (code) {

        case 0:

            c->bit_length--;

            break;

        case 1:

            c->bit_length++;

            break;

        case 2:

            c->bit_length = get_bits(gb, 5);

            break;

        case 3:

            c->block_length = get_bits(gb, 4);

            return 1;

        }

    }

    return 0;

}

static int apac_decode(AVCodecContext *avctx, AVFrame *frame,

                       int *got_frame_ptr, AVPacket *pkt)

{

    APACContext *s = avctx->priv_data;

    GetBitContext *gb = &s->gb;

    int ret, n, buf_size, input_buf_size;

    uint8_t *buf;

    int nb_samples;

    if (!pkt->size && s->bitstream_size <= 0) {

        *got_frame_ptr = 0;

        return 0;

    }

    buf_size = pkt->size;

    input_buf_size = buf_size;

    if (s->bitstream_index > 0 && s->bitstream_size > 0) {

        memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);

        s->bitstream_index = 0;

    }

    if (s->bitstream_index + s->bitstream_size + buf_size > s->max_framesize) {

        s->bitstream = av_realloc_f(s->bitstream, s->bitstream_index +

                                    s->bitstream_size +

                                    buf_size + AV_INPUT_BUFFER_PADDING_SIZE,

                                    sizeof(*s->bitstream));

        if (!s->bitstream)

            return AVERROR(ENOMEM);

        s->max_framesize = s->bitstream_index + s->bitstream_size + buf_size;

    }

    if (pkt->data)

        memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);

    buf                = &s->bitstream[s->bitstream_index];

    buf_size          += s->bitstream_size;

    s->bitstream_size  = buf_size;

    memset(buf + buf_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);

    frame->nb_samples = s->bitstream_size * 16 * 8;

    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

        return ret;

    if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)

        return ret;

    skip_bits(gb, s->skip);

    s->skip = 0;

    while (get_bits_left(gb) > 0) {

        for (int ch = s->cur_ch; ch < avctx->ch_layout.nb_channels; ch++) {

            ChContext *c = &s->ch[ch];

            int16_t *dst16 = (int16_t *)c->block;

            uint8_t *dst8 = (uint8_t *)c->block;

            void *samples[4];

            samples[0] = &c->block[0];

            if (get_bits_left(gb) < 16 && pkt->size) {

                s->cur_ch = ch;

                goto end;

            }

            if (!c->have_code && get_code(c, gb))

                get_code(c, gb);

            c->have_code = 0;

            if (c->block_length <= 0)

                continue;

            if (c->bit_length < 0 ||

                c->bit_length > 17) {

                c->bit_length = avctx->bits_per_coded_sample;

                s->bitstream_index = 0;

                s->bitstream_size  = 0;

                return AVERROR_INVALIDDATA;

            }

            if (get_bits_left(gb) < c->block_length * c->bit_length) {

                if (pkt->size) {

                    c->have_code = 1;

                    s->cur_ch = ch;

                    goto end;

                } else {

                    break;

                }

            }

            for (int i = 0; i < c->block_length; i++) {

                int val = get_bits_long(gb, c->bit_length);

                unsigned delta = (val & 1) ? ~(val >> 1) : (val >> 1);

                int sample;

                delta += c->last_delta;

                sample = c->last_sample + delta;

                c->last_delta = delta;

                c->last_sample = sample;

                switch (avctx->sample_fmt) {

                case AV_SAMPLE_FMT_S16P:

                    dst16[i] = sample;

                    break;

                case AV_SAMPLE_FMT_U8P:

                    dst8[i] = sample;

                    break;

                }

            }

            av_audio_fifo_write(c->samples, samples, c->block_length);

        }

        s->cur_ch = 0;

    }

end:

    nb_samples = frame->nb_samples;

    for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)

        nb_samples = FFMIN(av_audio_fifo_size(s->ch[ch].samples), nb_samples);

    frame->nb_samples = nb_samples;

    for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {

        void *samples[1] = { frame->extended_data[ch] };

        av_audio_fifo_read(s->ch[ch].samples, samples, nb_samples);

    }

    s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);

    n = get_bits_count(gb) / 8;

    if (nb_samples > 0 || pkt->size)

        *got_frame_ptr = 1;

    if (s->bitstream_size > 0) {

        s->bitstream_index += n;

        s->bitstream_size  -= n;

        return input_buf_size;

    }

    return n;

}

const FFCodec ff_apac_decoder = {

    .p.name           = "apac",

    CODEC_LONG_NAME("Marian's A-pac audio"),

    .p.type           = AVMEDIA_TYPE_AUDIO,

    .p.id             = AV_CODEC_ID_APAC,

    .priv_data_size   = sizeof(APACContext),

    .init             = apac_init,

    FF_CODEC_DECODE_CB(apac_decode),

    .close            = apac_close,

    .p.capabilities   = AV_CODEC_CAP_DELAY |

                        AV_CODEC_CAP_DR1,

    .caps_internal    = FF_CODEC_CAP_INIT_CLEANUP,

    CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P),

};