view ra288.c @ 7744:7477cbdacb20 libavcodec

Fix lossless jpeg encoder to comply to spec and store full redundant residuals, Note this does not change RGB32 as we need to check this against some decoder that supports it.

/*
 * RealAudio 2.0 (28.8K)
 * Copyright (c) 2003 the ffmpeg project
 *
 * 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 "avcodec.h"
#define ALT_BITSTREAM_READER_LE
#include "bitstream.h"
#include "ra288.h"

typedef struct {
    float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)
    float gain_lpc[10];    ///< LPC coefficients for gain (spec: GB)

    float sp_hist[111];    ///< Speech data history (spec: SB)

    /** Speech part of the gain autocorrelation (spec: REXP) */
    float sp_rec[37];

    float gain_hist[38];   ///< Log-gain history (spec: SBLG)

    /** Recursive part of the gain autocorrelation (spec: REXPLG) */
    float gain_rec[11];

    float sp_block[41];    ///< Speech data of four blocks (spec: STTMP)
    float gain_block[10];  ///< Gain data of four blocks (spec: GSTATE)
} RA288Context;

static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
    avctx->sample_fmt = SAMPLE_FMT_S16;
    return 0;
}

static inline float scalar_product_float(const float * v1, const float * v2,
                                         int size)
{
    float res = 0.;

    while (size--)
        res += *v1++ * *v2++;

    return res;
}

static void colmult(float *tgt, const float *m1, const float *m2, int n)
{
    while (n--)
        *tgt++ = *m1++ * *m2++;
}

static void decode(RA288Context *ractx, float gain, int cb_coef)
{
    int i, j;
    double sumsum;
    float sum, buffer[5];
    float *block = ractx->sp_block + 36; // Current block

    memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block));

    for (i=0; i < 5; i++) {
        block[i] = 0.;
        for (j=0; j < 36; j++)
            block[i] -= block[i-1-j]*ractx->sp_lpc[j];
    }

    /* block 46 of G.728 spec */
    sum = 32.;
    for (i=0; i < 10; i++)
        sum -= ractx->gain_block[9-i] * ractx->gain_lpc[i];

    /* block 47 of G.728 spec */
    sum = av_clipf(sum, 0, 60);

    /* block 48 of G.728 spec */
    sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */

    for (i=0; i < 5; i++)
        buffer[i] = codetable[cb_coef][i] * sumsum;

    sum = scalar_product_float(buffer, buffer, 5) / 5;

    sum = FFMAX(sum, 1);

    /* shift and store */
    memmove(ractx->gain_block, ractx->gain_block + 1,
            9 * sizeof(*ractx->gain_block));

    ractx->gain_block[9] = 10 * log10(sum) - 32;

    for (i=1; i < 5; i++)
        for (j=i-1; j >= 0; j--)
            buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j];

    /* output */
    for (i=0; i < 5; i++)
        block[i] = av_clipf(block[i] + buffer[i], -4095, 4095);
}

/**
 * Converts autocorrelation coefficients to LPC coefficients using the
 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
 *
 * @return 0 if success, -1 if fail
 */
static int eval_lpc_coeffs(const float *in, float *tgt, int n)
{
    int i, j;
    double f0, f1, f2;

    if (in[n] == 0)
        return -1;

    if ((f0 = *in) <= 0)
        return -1;

    in--; // To avoid a -1 subtraction in the inner loop

    for (i=1; i <= n; i++) {
        f1 = in[i+1];

        for (j=0; j < i - 1; j++)
            f1 += in[i-j]*tgt[j];

        tgt[i-1] = f2 = -f1/f0;
        for (j=0; j < i >> 1; j++) {
            float temp = tgt[j] + tgt[i-j-2]*f2;
            tgt[i-j-2] += tgt[j]*f2;
            tgt[j] = temp;
        }
        if ((f0 += f1*f2) < 0)
            return -1;
    }

    return 0;
}

static void convolve(float *tgt, const float *src, int len, int n)
{
    for (; n >= 0; n--)
        tgt[n] = scalar_product_float(src, src - n, len);

}

/**
 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
 *
 * @param order   the order of the filter
 * @param n       the length of the input
 * @param non_rec the number of non-recursive samples
 * @param out     the filter output
 * @param in      pointer to the input of the filter
 * @param hist    pointer to the input history of the filter. It is updated by
 *                this function.
 * @param out     pointer to the non-recursive part of the output
 * @param out2    pointer to the recursive part of the output
 * @param window  pointer to the windowing function table
 */
static void do_hybrid_window(int order, int n, int non_rec, const float *in,
                             float *out, float *hist, float *out2,
                             const float *window)
{
    int i;
    float buffer1[order + 1];
    float buffer2[order + 1];
    float work[order + n + non_rec];

    /* update history */
    memmove(hist                  , hist + n, (order + non_rec)*sizeof(*hist));
    memcpy (hist + order + non_rec, in      , n                *sizeof(*hist));

    colmult(work, window, hist, order + n + non_rec);

    convolve(buffer1, work + order    , n      , order);
    convolve(buffer2, work + order + n, non_rec, order);

    for (i=0; i <= order; i++) {
        out2[i] = out2[i] * 0.5625 + buffer1[i];
        out [i] = out2[i]          + buffer2[i];
    }

    /* Multiply by the white noise correcting factor (WNCF) */
    *out *= 257./256.;
}

/**
 * Backward synthesis filter. Find the LPC coefficients from past speech data.
 */
static void backward_filter(RA288Context *ractx)
{
    float temp1[37]; // RTMP in the spec
    float temp2[11]; // GPTPMP in the spec

    do_hybrid_window(36, 40, 35, ractx->sp_block+1, temp1, ractx->sp_hist,
                     ractx->sp_rec, syn_window);

    if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36))
        colmult(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);

    do_hybrid_window(10, 8, 20, ractx->gain_block+2, temp2, ractx->gain_hist,
                     ractx->gain_rec, gain_window);

    if (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10))
        colmult(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
}

static int ra288_decode_frame(AVCodecContext * avctx, void *data,
                              int *data_size, const uint8_t * buf,
                              int buf_size)
{
    int16_t *out = data;
    int i, j;
    RA288Context *ractx = avctx->priv_data;
    GetBitContext gb;

    if (buf_size < avctx->block_align) {
        av_log(avctx, AV_LOG_ERROR,
               "Error! Input buffer is too small [%d<%d]\n",
               buf_size, avctx->block_align);
        return 0;
    }

    if (*data_size < 32*5*2)
        return -1;

    init_get_bits(&gb, buf, avctx->block_align * 8);

    for (i=0; i < 32; i++) {
        float gain = amptable[get_bits(&gb, 3)];
        int cb_coef = get_bits(&gb, 6 + (i&1));

        decode(ractx, gain, cb_coef);

        for (j=0; j < 5; j++)
            *(out++) = 8 * ractx->sp_block[36 + j];

        if ((i & 7) == 3)
            backward_filter(ractx);
    }

    *data_size = (char *)out - (char *)data;
    return avctx->block_align;
}

AVCodec ra_288_decoder =
{
    "real_288",
    CODEC_TYPE_AUDIO,
    CODEC_ID_RA_288,
    sizeof(RA288Context),
    ra288_decode_init,
    NULL,
    NULL,
    ra288_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};