view ra144.c @ 11032:01bd040f8607 libavcodec

Unroll main loop so the edge==0 case is seperate. This allows many things to be simplified away. h264 decoder is overall 1% faster with a mbaff sample and 0.1% slower with the cathedral sample, probably because the slow loop filter code must be loaded into the code cache for each first MB of each row but isnt used for the following MBs.

/*
 * Real Audio 1.0 (14.4K)
 *
 * Copyright (c) 2008 Vitor Sessak
 * Copyright (c) 2003 Nick Kurshev
 *     Based on public domain decoder at http://www.honeypot.net/audio
 *
 * 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"
#include "get_bits.h"
#include "ra144.h"
#include "celp_filters.h"

#define NBLOCKS         4       ///< number of subblocks within a block
#define BLOCKSIZE       40      ///< subblock size in 16-bit words
#define BUFFERSIZE      146     ///< the size of the adaptive codebook


typedef struct {
    AVCodecContext *avctx;

    unsigned int     old_energy;        ///< previous frame energy

    unsigned int     lpc_tables[2][10];

    /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
     *  and lpc_coef[1] of the previous one. */
    unsigned int    *lpc_coef[2];

    unsigned int     lpc_refl_rms[2];

    /** The current subblock padded by the last 10 values of the previous one. */
    int16_t curr_sblock[50];

    /** Adaptive codebook, its size is two units bigger to avoid a
     *  buffer overflow. */
    uint16_t adapt_cb[146+2];
} RA144Context;

static av_cold int ra144_decode_init(AVCodecContext * avctx)
{
    RA144Context *ractx = avctx->priv_data;

    ractx->avctx = avctx;

    ractx->lpc_coef[0] = ractx->lpc_tables[0];
    ractx->lpc_coef[1] = ractx->lpc_tables[1];

    avctx->sample_fmt = SAMPLE_FMT_S16;
    return 0;
}

/**
 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
 * odd way to make the output identical to the binary decoder.
 */
static int t_sqrt(unsigned int x)
{
    int s = 2;
    while (x > 0xfff) {
        s++;
        x >>= 2;
    }

    return ff_sqrt(x << 20) << s;
}

/**
 * Evaluate the LPC filter coefficients from the reflection coefficients.
 * Does the inverse of the eval_refl() function.
 */
static void eval_coefs(int *coefs, const int *refl)
{
    int buffer[10];
    int *b1 = buffer;
    int *b2 = coefs;
    int i, j;

    for (i=0; i < 10; i++) {
        b1[i] = refl[i] << 4;

        for (j=0; j < i; j++)
            b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j];

        FFSWAP(int *, b1, b2);
    }

    for (i=0; i < 10; i++)
        coefs[i] >>= 4;
}

/**
 * Copy the last offset values of *source to *target. If those values are not
 * enough to fill the target buffer, fill it with another copy of those values.
 */
static void copy_and_dup(int16_t *target, const int16_t *source, int offset)
{
    source += BUFFERSIZE - offset;

    memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target));
    if (offset < BLOCKSIZE)
        memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target));
}

/** inverse root mean square */
static int irms(const int16_t *data)
{
    unsigned int i, sum = 0;

    for (i=0; i < BLOCKSIZE; i++)
        sum += data[i] * data[i];

    if (sum == 0)
        return 0; /* OOPS - division by zero */

    return 0x20000000 / (t_sqrt(sum) >> 8);
}

static void add_wav(int16_t *dest, int n, int skip_first, int *m,
                    const int16_t *s1, const int8_t *s2, const int8_t *s3)
{
    int i;
    int v[3];

    v[0] = 0;
    for (i=!skip_first; i<3; i++)
        v[i] = (gain_val_tab[n][i] * m[i]) >> gain_exp_tab[n];

    if (v[0]) {
        for (i=0; i < BLOCKSIZE; i++)
            dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12;
    } else {
        for (i=0; i < BLOCKSIZE; i++)
            dest[i] = (             s2[i]*v[1] + s3[i]*v[2]) >> 12;
    }
}

static unsigned int rescale_rms(unsigned int rms, unsigned int energy)
{
    return (rms * energy) >> 10;
}

static unsigned int rms(const int *data)
{
    int i;
    unsigned int res = 0x10000;
    int b = 10;

    for (i=0; i < 10; i++) {
        res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12;

        if (res == 0)
            return 0;

        while (res <= 0x3fff) {
            b++;
            res <<= 2;
        }
    }

    return t_sqrt(res) >> b;
}

static void do_output_subblock(RA144Context *ractx, const uint16_t  *lpc_coefs,
                               int gval, GetBitContext *gb)
{
    uint16_t buffer_a[40];
    uint16_t *block;
    int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
    int gain    = get_bits(gb, 8);
    int cb1_idx = get_bits(gb, 7);
    int cb2_idx = get_bits(gb, 7);
    int m[3];

    if (cba_idx) {
        cba_idx += BLOCKSIZE/2 - 1;
        copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx);
        m[0] = (irms(buffer_a) * gval) >> 12;
    } else {
        m[0] = 0;
    }

    m[1] = (cb1_base[cb1_idx] * gval) >> 8;
    m[2] = (cb2_base[cb2_idx] * gval) >> 8;

    memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
            (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb));

    block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;

    add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL,
            cb1_vects[cb1_idx], cb2_vects[cb2_idx]);

    memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
           10*sizeof(*ractx->curr_sblock));

    if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
                                    block, BLOCKSIZE, 10, 1, 0xfff))
        memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
}

static void int_to_int16(int16_t *out, const int *inp)
{
    int i;

    for (i=0; i < 30; i++)
        *out++ = *inp++;
}

/**
 * Evaluate the reflection coefficients from the filter coefficients.
 * Does the inverse of the eval_coefs() function.
 *
 * @return 1 if one of the reflection coefficients is greater than
 *         4095, 0 if not.
 */
static int eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
{
    int b, i, j;
    int buffer1[10];
    int buffer2[10];
    int *bp1 = buffer1;
    int *bp2 = buffer2;

    for (i=0; i < 10; i++)
        buffer2[i] = coefs[i];

    refl[9] = bp2[9];

    if ((unsigned) bp2[9] + 0x1000 > 0x1fff) {
        av_log(avctx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
        return 1;
    }

    for (i=8; i >= 0; i--) {
        b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12);

        if (!b)
            b = -2;

        for (j=0; j <= i; j++)
            bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12;

        if ((unsigned) bp1[i] + 0x1000 > 0x1fff)
            return 1;

        refl[i] = bp1[i];

        FFSWAP(int *, bp1, bp2);
    }
    return 0;
}

static int interp(RA144Context *ractx, int16_t *out, int a,
                  int copyold, int energy)
{
    int work[10];
    int b = NBLOCKS - a;
    int i;

    // Interpolate block coefficients from the this frame's forth block and
    // last frame's forth block.
    for (i=0; i<30; i++)
        out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;

    if (eval_refl(work, out, ractx->avctx)) {
        // The interpolated coefficients are unstable, copy either new or old
        // coefficients.
        int_to_int16(out, ractx->lpc_coef[copyold]);
        return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
    } else {
        return rescale_rms(rms(work), energy);
    }
}

/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
                              int *data_size, AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
    unsigned int refl_rms[4];    // RMS of the reflection coefficients
    uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block
    unsigned int lpc_refl[10];   // LPC reflection coefficients of the frame
    int i, j;
    int16_t *data = vdata;
    unsigned int energy;

    RA144Context *ractx = avctx->priv_data;
    GetBitContext gb;

    if (*data_size < 2*160)
        return -1;

    if(buf_size < 20) {
        av_log(avctx, AV_LOG_ERROR,
               "Frame too small (%d bytes). Truncated file?\n", buf_size);
        *data_size = 0;
        return buf_size;
    }
    init_get_bits(&gb, buf, 20 * 8);

    for (i=0; i<10; i++)
        lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])];

    eval_coefs(ractx->lpc_coef[0], lpc_refl);
    ractx->lpc_refl_rms[0] = rms(lpc_refl);

    energy = energy_tab[get_bits(&gb, 5)];

    refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
    refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy,
                    t_sqrt(energy*ractx->old_energy) >> 12);
    refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy);
    refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy);

    int_to_int16(block_coefs[3], ractx->lpc_coef[0]);

    for (i=0; i < 4; i++) {
        do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);

        for (j=0; j < BLOCKSIZE; j++)
            *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
    }

    ractx->old_energy = energy;
    ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];

    FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

    *data_size = 2*160;
    return 20;
}

AVCodec ra_144_decoder =
{
    "real_144",
    CODEC_TYPE_AUDIO,
    CODEC_ID_RA_144,
    sizeof(RA144Context),
    ra144_decode_init,
    NULL,
    NULL,
    ra144_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};