PeerStreamer

/*

 * Bink Audio decoder

 * Copyright (c) 2007-2010 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 "avcodec.h"

#define ALT_BITSTREAM_READER_LE

#include "get_bits.h"

#include "dsputil.h"

#include "fft.h"

#include "fmtconvert.h"

extern const uint16_t ff_wma_critical_freqs[25];

#define MAX_CHANNELS 2

#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)

typedef struct {

    AVCodecContext *avctx;

    GetBitContext gb;

    DSPContext dsp;

    FmtConvertContext fmt_conv;

    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(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];

    DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block

    float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave

    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;

    s->avctx = avctx;

    dsputil_init(&s->dsp, avctx);

    ff_fmt_convert_init(&s->fmt_conv, avctx);

    /* 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;

    }

    s->frame_len = 1 << frame_len_bits;

    if (avctx->channels > MAX_CHANNELS) {

        av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);

        return -1;

    }

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

        // audio is already interleaved for the RDFT format variant

        sample_rate  *= avctx->channels;

        s->frame_len *= avctx->channels;

        s->channels = 1;

        if (avctx->channels == 2)

            frame_len_bits++;

    } else {

        s->channels = avctx->channels;

    }

    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;

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

    /* 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] = 1;

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

        s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;

    s->bands[s->num_bands] = s->frame_len / 2;

    s->first = 1;

    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

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

        s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;

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

 */

static void decode_block(BinkAudioContext *s, short *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 = s->coeffs_ptr[ch];

        q = 0.0f;

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

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

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

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

            int value = get_bits(gb, 8);

            quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;

        }

        // find band (k)

        for (k = 0; s->bands[k] < 1; k++) {

            q = quant[k];

        }

        // parse coefficients

        i = 2;

        while (i < s->frame_len) {

            if (get_bits1(gb)) {

                j = i + rle_length_tab[get_bits(gb, 4)] * 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] * 2 < i)

                    q = quant[k++];

            } else {

                while (i < j) {

                    if (s->bands[k] * 2 == i)

                        q = quant[k++];

                    coeff = get_bits(gb, width);

                    if (coeff) {

                        if (get_bits1(gb))

                            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;

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

            s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);

        }

        else if (CONFIG_BINKAUDIO_RDFT_DECODER)

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

    }

    s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,

                                          s->frame_len, s->channels);

    if (!s->first) {

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

        int shift = av_log2(count);

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

            out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;

        }

    }

    memcpy(s->previous, out + s->block_size,

           s->overlap_len * s->channels * sizeof(*out));

    s->first = 0;

}

static av_cold int decode_end(AVCodecContext *avctx)

{

    BinkAudioContext * s = avctx->priv_data;

    av_freep(&s->bands);

    if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == 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 *data_size,

                        AVPacket *avpkt)

{

    BinkAudioContext *s = avctx->priv_data;

    const uint8_t *buf  = avpkt->data;

    int buf_size        = avpkt->size;

    short *samples      = data;

    short *samples_end  = (short*)((uint8_t*)data + *data_size);

    int reported_size;

    GetBitContext *gb = &s->gb;

    init_get_bits(gb, buf, buf_size * 8);

    reported_size = get_bits_long(gb, 32);

    while (get_bits_count(gb) / 8 < buf_size &&

           samples + s->block_size <= samples_end) {

        decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);

        samples += s->block_size;

        get_bits_align32(gb);

    }

    *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);

    return buf_size;

}

AVCodec ff_binkaudio_rdft_decoder = {

    "binkaudio_rdft",

    AVMEDIA_TYPE_AUDIO,

    CODEC_ID_BINKAUDIO_RDFT,

    sizeof(BinkAudioContext),

    decode_init,

    NULL,

    decode_end,

    decode_frame,

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

};

AVCodec ff_binkaudio_dct_decoder = {

    "binkaudio_dct",

    AVMEDIA_TYPE_AUDIO,

    CODEC_ID_BINKAUDIO_DCT,

    sizeof(BinkAudioContext),

    decode_init,

    NULL,

    decode_end,

    decode_frame,

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

};