view qcelpdec.c @ 8228:416ffc3907bf libavcodec

Remove ineffectual hack that attempts to build ppc/check_altivec.o without AltiVec flags. The flags are set by configure and used to compile all files anyway. Setting extra AltiVec options here just duplicates them for the files for which they are set.

/*
 * QCELP decoder
 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
 *
 * 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 qcelpdec.c
 * QCELP decoder
 * @author Reynaldo H. Verdejo Pinochet
 * @remark FFmpeg merging spearheaded by Kenan Gillet
 */

#include <stddef.h>

#include "avcodec.h"
#include "bitstream.h"

#include "qcelp.h"
#include "qcelpdata.h"

#include "celp_math.h"
#include "celp_filters.h"

#undef NDEBUG
#include <assert.h>

static void weighted_vector_sumf(float *out, const float *in_a,
                                 const float *in_b, float weight_coeff_a,
                                 float weight_coeff_b, int length)
{
    int i;

    for(i=0; i<length; i++)
        out[i] = weight_coeff_a * in_a[i]
               + weight_coeff_b * in_b[i];
}

/**
 * Initialize the speech codec according to the specification.
 *
 * TIA/EIA/IS-733 2.4.9
 */
static av_cold int qcelp_decode_init(AVCodecContext *avctx)
{
    QCELPContext *q = avctx->priv_data;
    int i;

    avctx->sample_fmt = SAMPLE_FMT_FLT;

    for (i = 0; i < 10; i++)
        q->prev_lspf[i] = (i + 1) / 11.;

    return 0;
}

/**
 * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
 * transmission codes of any bitrate and checks for badly received packets.
 *
 * @param q the context
 * @param lspf line spectral pair frequencies
 *
 * @return 0 on success, -1 if the packet is badly received
 *
 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
 */
static int decode_lspf(QCELPContext *q, float *lspf)
{
    int i;
    float tmp_lspf;

    if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
    {
        float smooth;
        const float *predictors = (q->prev_bitrate != RATE_OCTAVE &&
                                   q->prev_bitrate != I_F_Q ? q->prev_lspf
                                                            : q->predictor_lspf);

        if(q->bitrate == RATE_OCTAVE)
        {
            q->octave_count++;

            for(i=0; i<10; i++)
            {
                q->predictor_lspf[i] =
                             lspf[i] = (q->lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
                                                   : -QCELP_LSP_SPREAD_FACTOR)
                                     + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
                                     + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
            }
            smooth = (q->octave_count < 10 ? .875 : 0.1);
        }else
        {
            float erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;

            assert(q->bitrate == I_F_Q);

            if(q->erasure_count > 1)
                erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);

            for(i=0; i<10; i++)
            {
                q->predictor_lspf[i] =
                             lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
                                     + erasure_coeff * predictors[i];
            }
            smooth = 0.125;
        }

        // Check the stability of the LSP frequencies.
        lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
        for(i=1; i<10; i++)
            lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));

        lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
        for(i=9; i>0; i--)
            lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));

        // Low-pass filter the LSP frequencies.
        weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
    }else
    {
        q->octave_count = 0;

        tmp_lspf = 0.;
        for(i=0; i<5 ; i++)
        {
            lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->lspv[i]][0] * 0.0001;
            lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->lspv[i]][1] * 0.0001;
        }

        // Check for badly received packets.
        if(q->bitrate == RATE_QUARTER)
        {
            if(lspf[9] <= .70 || lspf[9] >=  .97)
                return -1;
            for(i=3; i<10; i++)
                if(fabs(lspf[i] - lspf[i-2]) < .08)
                    return -1;
        }else
        {
            if(lspf[9] <= .66 || lspf[9] >= .985)
                return -1;
            for(i=4; i<10; i++)
                if (fabs(lspf[i] - lspf[i-4]) < .0931)
                    return -1;
        }
    }
    return 0;
}

/**
 * If the received packet is Rate 1/4 a further sanity check is made of the
 * codebook gain.
 *
 * @param cbgain the unpacked cbgain array
 * @return -1 if the sanity check fails, 0 otherwise
 *
 * TIA/EIA/IS-733 2.4.8.7.3
 */
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
{
    int i, prev_diff=0;

    for(i=1; i<5; i++)
    {
        int diff = cbgain[i] - cbgain[i-1];
        if(FFABS(diff) > 10)
            return -1;
        else if(FFABS(diff - prev_diff) > 12)
            return -1;
        prev_diff = diff;
    }
    return 0;
}

/**
 * Computes the scaled codebook vector Cdn From INDEX and GAIN
 * for all rates.
 *
 * The specification lacks some information here.
 *
 * TIA/EIA/IS-733 has an omission on the codebook index determination
 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
 * you have to subtract the decoded index parameter from the given scaled
 * codebook vector index 'n' to get the desired circular codebook index, but
 * it does not mention that you have to clamp 'n' to [0-9] in order to get
 * RI-compliant results.
 *
 * The reason for this mistake seems to be the fact they forgot to mention you
 * have to do these calculations per codebook subframe and adjust given
 * equation values accordingly.
 *
 * @param q the context
 * @param gain array holding the 4 pitch subframe gain values
 * @param cdn_vector array for the generated scaled codebook vector
 */
static void compute_svector(const QCELPContext *q, const float *gain,
                            float *cdn_vector)
{
    int      i, j, k;
    uint16_t cbseed, cindex;
    float    *rnd, tmp_gain, fir_filter_value;

    switch(q->bitrate)
    {
        case RATE_FULL:
            for(i=0; i<16; i++)
            {
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
                cindex = -q->cindex[i];
                for(j=0; j<10; j++)
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
            }
        break;
        case RATE_HALF:
            for(i=0; i<4; i++)
            {
                tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
                cindex = -q->cindex[i];
                for (j = 0; j < 40; j++)
                *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
            }
        break;
        case RATE_QUARTER:
            cbseed = (0x0003 & q->lspv[4])<<14 |
                     (0x003F & q->lspv[3])<< 8 |
                     (0x0060 & q->lspv[2])<< 1 |
                     (0x0007 & q->lspv[1])<< 3 |
                     (0x0038 & q->lspv[0])>> 3 ;
            rnd = q->rnd_fir_filter_mem + 20;
            for(i=0; i<8; i++)
            {
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
                for(k=0; k<20; k++)
                {
                    cbseed = 521 * cbseed + 259;
                    *rnd = (int16_t)cbseed;

                    // FIR filter
                    fir_filter_value = 0.0;
                    for(j=0; j<10; j++)
                        fir_filter_value += qcelp_rnd_fir_coefs[j ]
                                          * (rnd[-j ] + rnd[-20+j]);

                    fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
                    *cdn_vector++ = tmp_gain * fir_filter_value;
                    rnd++;
                }
            }
            memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
        break;
        case RATE_OCTAVE:
            cbseed = q->first16bits;
            for(i=0; i<8; i++)
            {
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
                for(j=0; j<20; j++)
                {
                    cbseed = 521 * cbseed + 259;
                    *cdn_vector++ = tmp_gain * (int16_t)cbseed;
                }
            }
        break;
        case I_F_Q:
            cbseed = -44; // random codebook index
            for(i=0; i<4; i++)
            {
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
                for(j=0; j<40; j++)
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
            }
        break;
    }
}

/**
 * Apply generic gain control.
 *
 * @param v_out output vector
 * @param v_in gain-controlled vector
 * @param v_ref vector to control gain of
 *
 * FIXME: If v_ref is a zero vector, it energy is zero
 *        and the behavior of the gain control is
 *        undefined in the specs.
 *
 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
 */
static void apply_gain_ctrl(float *v_out, const float *v_ref,
                            const float *v_in)
{
    int   i, j, len;
    float scalefactor;

    for(i=0, j=0; i<4; i++)
    {
        scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
        if(scalefactor)
            scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
                        / scalefactor);
        else
            av_log_missing_feature(NULL, "Zero energy for gain control", 1);
        for(len=j+40; j<len; j++)
            v_out[j] = scalefactor * v_in[j];
    }
}

/**
 * Apply filter in pitch-subframe steps.
 *
 * @param memory buffer for the previous state of the filter
 *        - must be able to contain 303 elements
 *        - the 143 first elements are from the previous state
 *        - the next 160 are for output
 * @param v_in input filter vector
 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
 * @param lag per-subframe lag array, each element is
 *        - between 16 and 143 if its corresponding pfrac is 0,
 *        - between 16 and 139 otherwise
 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
 *        otherwise
 *
 * @return filter output vector
 */
static const float *do_pitchfilter(float memory[303], const float v_in[160],
                                   const float gain[4], const uint8_t *lag,
                                   const uint8_t pfrac[4])
{
    int         i, j;
    float       *v_lag, *v_out;
    const float *v_len;

    v_out = memory + 143; // Output vector starts at memory[143].

    for(i=0; i<4; i++)
    {
        if(gain[i])
        {
            v_lag = memory + 143 + 40 * i - lag[i];
            for(v_len=v_in+40; v_in<v_len; v_in++)
            {
                if(pfrac[i]) // If it is a fractional lag...
                {
                    for(j=0, *v_out=0.; j<4; j++)
                        *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
                }else
                    *v_out = *v_lag;

                *v_out = *v_in + gain[i] * *v_out;

                v_lag++;
                v_out++;
            }
        }else
        {
            memcpy(v_out, v_in, 40 * sizeof(float));
            v_in  += 40;
            v_out += 40;
        }
    }

    memmove(memory, memory + 160, 143 * sizeof(float));
    return memory + 143;
}

/**
 * Interpolates LSP frequencies and computes LPC coefficients
 * for a given bitrate & pitch subframe.
 *
 * TIA/EIA/IS-733 2.4.3.3.4
 *
 * @param q the context
 * @param curr_lspf LSP frequencies vector of the current frame
 * @param lpc float vector for the resulting LPC
 * @param subframe_num frame number in decoded stream
 */
void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
                     const int subframe_num)
{
    float interpolated_lspf[10];
    float weight;

    if(q->bitrate >= RATE_QUARTER)
        weight = 0.25 * (subframe_num + 1);
    else if(q->bitrate == RATE_OCTAVE && !subframe_num)
        weight = 0.625;
    else
        weight = 1.0;

    if(weight != 1.0)
    {
        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
                             weight, 1.0 - weight, 10);
        qcelp_lspf2lpc(interpolated_lspf, lpc);
    }else if(q->bitrate >= RATE_QUARTER || (q->bitrate == I_F_Q && !subframe_num))
        qcelp_lspf2lpc(curr_lspf, lpc);
}

static int buf_size2bitrate(const int buf_size)
{
    switch(buf_size)
    {
        case 35:
            return RATE_FULL;
        case 17:
            return RATE_HALF;
        case  8:
            return RATE_QUARTER;
        case  4:
            return RATE_OCTAVE;
        case  1:
            return SILENCE;
    }

    return -1;
}

static void warn_insufficient_frame_quality(AVCodecContext *avctx,
                                            const char *message)
{
    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
           message);
}

AVCodec qcelp_decoder =
{
    .name   = "qcelp",
    .type   = CODEC_TYPE_AUDIO,
    .id     = CODEC_ID_QCELP,
    .init   = qcelp_decode_init,
    .decode = qcelp_decode_frame,
    .priv_data_size = sizeof(QCELPContext),
    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
};