annotate ra288.c @ 7891:13ee9bb85721 libavcodec

Zero 'idx' for each iteration of the coupling gain loop and as it is only used within this loop, we can move its declaration there too. This fixes bitstream desync when decoding streams containing CCEs. Based on a patch by Alex Converse (alex converse gmail com)

/*
 * 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"
#include "lpc.h"

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

    /** speech data history                                      (spec: SB).
     *  Its first 70 coefficients are updated only at backward filtering.
     */
    float sp_hist[111];

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

    /** log-gain history                                         (spec: SBLG).
     *  Its first 28 coefficients are updated only at backward filtering.
     */
    float gain_hist[38];

    /// recursive part of the gain autocorrelation               (spec: REXPLG)
    float gain_rec[11];
} RA288Context;

static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
    avctx->sample_fmt = SAMPLE_FMT_FLT;
    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 apply_window(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_hist + 70 + 36; // current block
    float *gain_block = ractx->gain_hist + 28;

    memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*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 -= 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 * (1./2048.);

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

    sum = FFMAX(sum, 1);

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

    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);
}

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   filter order
 * @param n       input length
 * @param non_rec number of non-recursive samples
 * @param out     filter output
 * @param hist    pointer to the input history of the filter
 * @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, 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];

    apply_window(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, temp1, ractx->sp_hist,
                     ractx->sp_rec, syn_window);

    if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1))
        apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);

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

    if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1))
        apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);

    memmove(ractx->gain_hist, ractx->gain_hist + 8,
                                 28*sizeof(*ractx->gain_hist));

    memmove(ractx->sp_hist  , ractx->sp_hist   + 40,
                                 70*sizeof(*ractx->sp_hist  ));
}

static int ra288_decode_frame(AVCodecContext * avctx, void *data,
                              int *data_size, const uint8_t * buf,
                              int buf_size)
{
    float *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*4)
        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++) = (1/4096.) * ractx->sp_hist[70 + 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)"),
};