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ffmpeg / libavcodec / binkaudio.c @ 72415b2a

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/*
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 * Bink Audio decoder
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 * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
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 * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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/**
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 * @file libavcodec/binkaudio.c
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 * Bink Audio decoder
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 *
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 * Technical details here:
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 *  http://wiki.multimedia.cx/index.php?title=Bink_Audio
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 */
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#include "avcodec.h"
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#define ALT_BITSTREAM_READER_LE
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#include "get_bits.h"
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#include "dsputil.h"
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#include "fft.h"
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extern const uint16_t ff_wma_critical_freqs[25];
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#define MAX_CHANNELS 2
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#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
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typedef struct {
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    AVCodecContext *avctx;
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    GetBitContext gb;
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    DSPContext dsp;
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    int first;
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    int channels;
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    int frame_len;          ///< transform size (samples)
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    int overlap_len;        ///< overlap size (samples)
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    int block_size;
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    int num_bands;
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    unsigned int *bands;
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    float root;
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    DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
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    DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
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    float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
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    union {
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        RDFTContext rdft;
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        DCTContext dct;
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    } trans;
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} BinkAudioContext;
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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    BinkAudioContext *s = avctx->priv_data;
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    int sample_rate = avctx->sample_rate;
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    int sample_rate_half;
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    int i;
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    int frame_len_bits;
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    s->avctx = avctx;
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    dsputil_init(&s->dsp, avctx);
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    /* determine frame length */
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    if (avctx->sample_rate < 22050) {
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        frame_len_bits = 9;
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    } else if (avctx->sample_rate < 44100) {
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        frame_len_bits = 10;
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    } else {
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        frame_len_bits = 11;
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    }
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    s->frame_len = 1 << frame_len_bits;
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    if (s->channels > MAX_CHANNELS) {
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        av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
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        return -1;
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    }
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    if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
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        // audio is already interleaved for the RDFT format variant
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        sample_rate  *= avctx->channels;
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        s->frame_len *= avctx->channels;
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        s->channels = 1;
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        if (avctx->channels == 2)
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            frame_len_bits++;
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    } else {
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        s->channels = avctx->channels;
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    }
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    s->overlap_len   = s->frame_len / 16;
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    s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
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    sample_rate_half = (sample_rate + 1) / 2;
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    s->root          = 2.0 / sqrt(s->frame_len);
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    /* calculate number of bands */
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    for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
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        if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
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            break;
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    s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
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    if (!s->bands)
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        return AVERROR(ENOMEM);
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    /* populate bands data */
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    s->bands[0] = 1;
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    for (i = 1; i < s->num_bands; i++)
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        s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;
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    s->bands[s->num_bands] = s->frame_len / 2;
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    s->first = 1;
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    avctx->sample_fmt = SAMPLE_FMT_S16;
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    for (i = 0; i < s->channels; i++)
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        s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
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    if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
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        ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
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    else if (CONFIG_BINKAUDIO_DCT_DECODER)
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        ff_dct_init(&s->trans.dct, frame_len_bits, 1);
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    else
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        return -1;
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    return 0;
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}
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static float get_float(GetBitContext *gb)
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{
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    int power = get_bits(gb, 5);
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    float f = ldexpf(get_bits_long(gb, 23), power - 23);
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    if (get_bits1(gb))
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        f = -f;
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    return f;
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}
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static const uint8_t rle_length_tab[16] = {
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    2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
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};
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/**
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 * Decode Bink Audio block
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 * @param[out] out Output buffer (must contain s->block_size elements)
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 */
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static void decode_block(BinkAudioContext *s, short *out, int use_dct)
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{
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    int ch, i, j, k;
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    float q, quant[25];
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    int width, coeff;
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    GetBitContext *gb = &s->gb;
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    if (use_dct)
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        skip_bits(gb, 2);
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    for (ch = 0; ch < s->channels; ch++) {
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        FFTSample *coeffs = s->coeffs_ptr[ch];
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        q = 0.0f;
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        coeffs[0] = get_float(gb) * s->root;
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        coeffs[1] = get_float(gb) * s->root;
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        for (i = 0; i < s->num_bands; i++) {
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            /* constant is result of 0.066399999/log10(M_E) */
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            int value = get_bits(gb, 8);
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            quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
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        }
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        // find band (k)
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        for (k = 0; s->bands[k] < 1; k++) {
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            q = quant[k];
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        }
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        // parse coefficients
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        i = 2;
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        while (i < s->frame_len) {
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            if (get_bits1(gb)) {
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                j = i + rle_length_tab[get_bits(gb, 4)] * 8;
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            } else {
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                j = i + 8;
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            }
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            j = FFMIN(j, s->frame_len);
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            width = get_bits(gb, 4);
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            if (width == 0) {
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                memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
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                i = j;
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                while (s->bands[k] * 2 < i)
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                    q = quant[k++];
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            } else {
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                while (i < j) {
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                    if (s->bands[k] * 2 == i)
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                        q = quant[k++];
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                    coeff = get_bits(gb, width);
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                    if (coeff) {
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                        if (get_bits1(gb))
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                            coeffs[i] = -q * coeff;
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                        else
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                            coeffs[i] =  q * coeff;
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                    } else {
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                        coeffs[i] = 0.0f;
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                    }
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                    i++;
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                }
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            }
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        }
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        if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
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            coeffs[0] /= 0.5;
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            ff_dct_calc (&s->trans.dct,  coeffs);
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            s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
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        }
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        else if (CONFIG_BINKAUDIO_RDFT_DECODER)
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            ff_rdft_calc(&s->trans.rdft, coeffs);
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    }
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    if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
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        for (i = 0; i < s->channels; i++)
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            for (j = 0; j < s->frame_len; j++)
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                s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0);
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    }
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    s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
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    if (!s->first) {
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        int count = s->overlap_len * s->channels;
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        int shift = av_log2(count);
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        for (i = 0; i < count; i++) {
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            out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
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        }
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    }
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    memcpy(s->previous, out + s->block_size,
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           s->overlap_len * s->channels * sizeof(*out));
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    s->first = 0;
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}
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static av_cold int decode_end(AVCodecContext *avctx)
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{
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    BinkAudioContext * s = avctx->priv_data;
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    av_freep(&s->bands);
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    if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
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        ff_rdft_end(&s->trans.rdft);
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    else if (CONFIG_BINKAUDIO_DCT_DECODER)
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        ff_dct_end(&s->trans.dct);
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    return 0;
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}
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static void get_bits_align32(GetBitContext *s)
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{
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    int n = (-get_bits_count(s)) & 31;
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    if (n) skip_bits(s, n);
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}
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static int decode_frame(AVCodecContext *avctx,
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                        void *data, int *data_size,
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                        AVPacket *avpkt)
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{
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    BinkAudioContext *s = avctx->priv_data;
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    const uint8_t *buf  = avpkt->data;
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    int buf_size        = avpkt->size;
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    short *samples      = data;
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    short *samples_end  = (short*)((uint8_t*)data + *data_size);
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    int reported_size;
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    GetBitContext *gb = &s->gb;
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    init_get_bits(gb, buf, buf_size * 8);
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    reported_size = get_bits_long(gb, 32);
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    while (get_bits_count(gb) / 8 < buf_size &&
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           samples + s->block_size <= samples_end) {
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        decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
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        samples += s->block_size;
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        get_bits_align32(gb);
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    }
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    *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
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    return buf_size;
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}
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AVCodec binkaudio_rdft_decoder = {
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    "binkaudio_rdft",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_BINKAUDIO_RDFT,
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    sizeof(BinkAudioContext),
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    decode_init,
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    NULL,
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    decode_end,
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    decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
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};
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AVCodec binkaudio_dct_decoder = {
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    "binkaudio_dct",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_BINKAUDIO_DCT,
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    sizeof(BinkAudioContext),
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    decode_init,
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    NULL,
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    decode_end,
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    decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
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};