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/*

 * Bink Audio decoder

 * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)

 * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)

 *

 * 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

 * Bink Audio decoder

 *

 * Technical details here:

 *  http://wiki.multimedia.cx/index.php?title=Bink_Audio

 */

#include "libavutil/channel_layout.h"

#include "avcodec.h"

#define BITSTREAM_READER_LE

#include "get_bits.h"

#include "dct.h"

#include "rdft.h"

#include "fmtconvert.h"

#include "internal.h"

#include "wma.h"

#include "libavutil/intfloat.h"

static float quant_table[96];

#define MAX_CHANNELS 2

#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)

typedef struct {

    GetBitContext gb;

    int version_b;          ///< Bink version 'b'

    int first;

    int channels;

    int frame_len;          ///< transform size (samples)

    int overlap_len;        ///< overlap size (samples)

    int block_size;

    int num_bands;

    unsigned int *bands;

    float root;

    DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];

    float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block

    uint8_t *packet_buffer;

    union {

        RDFTContext rdft;

        DCTContext dct;

    } trans;

} BinkAudioContext;

static av_cold int decode_init(AVCodecContext *avctx)

{

    BinkAudioContext *s = avctx->priv_data;

    int sample_rate = avctx->sample_rate;

    int sample_rate_half;

    int i;

    int frame_len_bits;

    /* determine frame length */

    if (avctx->sample_rate < 22050) {

        frame_len_bits = 9;

    } else if (avctx->sample_rate < 44100) {

        frame_len_bits = 10;

    } else {

        frame_len_bits = 11;

    }

    if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {

        av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);

        return AVERROR_INVALIDDATA;

    }

    avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :

                                                   AV_CH_LAYOUT_STEREO;

    s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';

    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {

        // audio is already interleaved for the RDFT format variant

        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

        sample_rate  *= avctx->channels;

        s->channels = 1;

        if (!s->version_b)

            frame_len_bits += av_log2(avctx->channels);

    } else {

        s->channels = avctx->channels;

        avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    }

    s->frame_len     = 1 << frame_len_bits;

    s->overlap_len   = s->frame_len / 16;

    s->block_size    = (s->frame_len - s->overlap_len) * s->channels;

    sample_rate_half = (sample_rate + 1) / 2;

    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

        s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);

    else

        s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);

    for (i = 0; i < 96; i++) {

        /* constant is result of 0.066399999/log10(M_E) */

        quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;

    }

    /* calculate number of bands */

    for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)

        if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])

            break;

    s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));

    if (!s->bands)

        return AVERROR(ENOMEM);

    /* populate bands data */

    s->bands[0] = 2;

    for (i = 1; i < s->num_bands; i++)

        s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;

    s->bands[s->num_bands] = s->frame_len;

    s->first = 1;

    if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

        ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);

    else if (CONFIG_BINKAUDIO_DCT_DECODER)

        ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);

    else

        return -1;

    return 0;

}

static float get_float(GetBitContext *gb)

{

    int power = get_bits(gb, 5);

    float f = ldexpf(get_bits_long(gb, 23), power - 23);

    if (get_bits1(gb))

        f = -f;

    return f;

}

static const uint8_t rle_length_tab[16] = {

    2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64

};

/**

 * Decode Bink Audio block

 * @param[out] out Output buffer (must contain s->block_size elements)

 * @return 0 on success, negative error code on failure

 */

static int decode_block(BinkAudioContext *s, float **out, int use_dct)

{

    int ch, i, j, k;

    float q, quant[25];

    int width, coeff;

    GetBitContext *gb = &s->gb;

    if (use_dct)

        skip_bits(gb, 2);

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

        FFTSample *coeffs = out[ch];

        if (s->version_b) {

            if (get_bits_left(gb) < 64)

                return AVERROR_INVALIDDATA;

            coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;

            coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;

        } else {

            if (get_bits_left(gb) < 58)

                return AVERROR_INVALIDDATA;

            coeffs[0] = get_float(gb) * s->root;

            coeffs[1] = get_float(gb) * s->root;

        }

        if (get_bits_left(gb) < s->num_bands * 8)

            return AVERROR_INVALIDDATA;

        for (i = 0; i < s->num_bands; i++) {

            int value = get_bits(gb, 8);

            quant[i]  = quant_table[FFMIN(value, 95)];

        }

        k = 0;

        q = quant[0];

        // parse coefficients

        i = 2;

        while (i < s->frame_len) {

            if (s->version_b) {

                j = i + 16;

            } else {

                int v = get_bits1(gb);

                if (v) {

                    v = get_bits(gb, 4);

                    j = i + rle_length_tab[v] * 8;

                } else {

                    j = i + 8;

                }

            }

            j = FFMIN(j, s->frame_len);

            width = get_bits(gb, 4);

            if (width == 0) {

                memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));

                i = j;

                while (s->bands[k] < i)

                    q = quant[k++];

            } else {

                while (i < j) {

                    if (s->bands[k] == i)

                        q = quant[k++];

                    coeff = get_bits(gb, width);

                    if (coeff) {

                        int v;

                        v = get_bits1(gb);

                        if (v)

                            coeffs[i] = -q * coeff;

                        else

                            coeffs[i] =  q * coeff;

                    } else {

                        coeffs[i] = 0.0f;

                    }

                    i++;

                }

            }

        }

        if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {

            coeffs[0] /= 0.5;

            s->trans.dct.dct_calc(&s->trans.dct,  coeffs);

        }

        else if (CONFIG_BINKAUDIO_RDFT_DECODER)

            s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);

    }

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

        int j;

        int count = s->overlap_len * s->channels;

        if (!s->first) {

            j = ch;

            for (i = 0; i < s->overlap_len; i++, j += s->channels)

                out[ch][i] = (s->previous[ch][i] * (count - j) +

                                      out[ch][i] *          j) / count;

        }

        memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],

               s->overlap_len * sizeof(*s->previous[ch]));

    }

    s->first = 0;

    return 0;

}

static av_cold int decode_end(AVCodecContext *avctx)

{

    BinkAudioContext * s = avctx->priv_data;

    av_freep(&s->bands);

    av_freep(&s->packet_buffer);

    if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

        ff_rdft_end(&s->trans.rdft);

    else if (CONFIG_BINKAUDIO_DCT_DECODER)

        ff_dct_end(&s->trans.dct);

    return 0;

}

static void get_bits_align32(GetBitContext *s)

{

    int n = (-get_bits_count(s)) & 31;

    if (n) skip_bits(s, n);

}

static int decode_frame(AVCodecContext *avctx, void *data,

                        int *got_frame_ptr, AVPacket *avpkt)

{

    BinkAudioContext *s = avctx->priv_data;

    AVFrame *frame      = data;

    GetBitContext *gb = &s->gb;

    int ret, consumed = 0;

    if (!get_bits_left(gb)) {

        uint8_t *buf;

        /* handle end-of-stream */

        if (!avpkt->size) {

            *got_frame_ptr = 0;

            return 0;

        }

        if (avpkt->size < 4) {

            av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");

            return AVERROR_INVALIDDATA;

        }

        buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);

        if (!buf)

            return AVERROR(ENOMEM);

        s->packet_buffer = buf;

        memcpy(s->packet_buffer, avpkt->data, avpkt->size);

        init_get_bits(gb, s->packet_buffer, avpkt->size * 8);

        consumed = avpkt->size;

        /* skip reported size */

        skip_bits_long(gb, 32);

    }

    /* get output buffer */

    frame->nb_samples = s->frame_len;

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

        return ret;

    if (decode_block(s, (float **)frame->extended_data,

                     avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {

        av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");

        return AVERROR_INVALIDDATA;

    }

    get_bits_align32(gb);

    frame->nb_samples = s->block_size / avctx->channels;

    *got_frame_ptr    = 1;

    return consumed;

}

AVCodec ff_binkaudio_rdft_decoder = {

    .name           = "binkaudio_rdft",

    .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = AV_CODEC_ID_BINKAUDIO_RDFT,

    .priv_data_size = sizeof(BinkAudioContext),

    .init           = decode_init,

    .close          = decode_end,

    .decode         = decode_frame,

    .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,

};

AVCodec ff_binkaudio_dct_decoder = {

    .name           = "binkaudio_dct",

    .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = AV_CODEC_ID_BINKAUDIO_DCT,

    .priv_data_size = sizeof(BinkAudioContext),

    .init           = decode_init,

    .close          = decode_end,

    .decode         = decode_frame,

    .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,

};