static int dv_extract_audio(uint8_t *frame, uint8_t *ppcm[4],

                            const DVprofile *sys)

{

    int size, chan, i, j, d, of, smpls, freq, quant, half_ch;

    uint16_t lc, rc;

    const uint8_t *as_pack;

    uint8_t *pcm, ipcm;



    as_pack = dv_extract_pack(frame, dv_audio_source);

    if (!as_pack)    /* No audio ? */

        return 0;



    smpls = as_pack[1]      & 0x3f; /* samples in this frame - min. samples */

    freq  = as_pack[4] >> 3 & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */

    quant = as_pack[4]      & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */



    if (quant > 1)

        return -1;  /* unsupported quantization */



    if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency))

        return AVERROR_INVALIDDATA;



    size    = (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */

    half_ch = sys->difseg_size / 2;



    /* We work with 720p frames split in half, thus even frames have

     * channels 0,1 and odd 2,3. */

    ipcm = (sys->height == 720 && !(frame[1] & 0x0C)) ? 2 : 0;



    /* for each DIF channel */

    for (chan = 0; chan < sys->n_difchan; chan++) {

        /* next stereo channel (50Mbps and 100Mbps only) */

        pcm = ppcm[ipcm++];

        if (!pcm)

            break;



        /* for each DIF segment */

        for (i = 0; i < sys->difseg_size; i++) {

            frame += 6 * 80; /* skip DIF segment header */

            if (quant == 1 && i == half_ch) {

                /* next stereo channel (12bit mode only) */

                pcm = ppcm[ipcm++];

                if (!pcm)

                    break;

            }



            /* for each AV sequence */

            for (j = 0; j < 9; j++) {

                for (d = 8; d < 80; d += 2) {

                    if (quant == 0) {  /* 16bit quantization */

                        of = sys->audio_shuffle[i][j] +

                             (d - 8) / 2 * sys->audio_stride;

                        if (of * 2 >= size)

                            continue;



                        /* FIXME: maybe we have to admit that DV is a

                         * big-endian PCM */

                        pcm[of * 2]     = frame[d + 1];

                        pcm[of * 2 + 1] = frame[d];



                        if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)

                            pcm[of * 2 + 1] = 0;

                    } else {           /* 12bit quantization */

                        lc = ((uint16_t)frame[d]     << 4) |

                             ((uint16_t)frame[d + 2] >> 4);

                        rc = ((uint16_t)frame[d + 1] << 4) |

                             ((uint16_t)frame[d + 2] & 0x0f);

                        lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));

                        rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc));



                        of = sys->audio_shuffle[i % half_ch][j] +

                             (d - 8) / 3 * sys->audio_stride;

                        if (of * 2 >= size)

                            continue;



                        /* FIXME: maybe we have to admit that DV is a

                         * big-endian PCM */

                        pcm[of * 2]     = lc & 0xff;

                        pcm[of * 2 + 1] = lc >> 8;

                        of = sys->audio_shuffle[i % half_ch + half_ch][j] +

                             (d - 8) / 3 * sys->audio_stride;

                        /* FIXME: maybe we have to admit that DV is a

                         * big-endian PCM */

                        pcm[of * 2]     = rc & 0xff;

                        pcm[of * 2 + 1] = rc >> 8;

                        ++d;

                    }

                }



                frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */

            }

        }

    }



    return size;

}

static int dv_extract_audio(uint8_t *frame, uint8_t *ppcm[4],

                            const DVprofile *sys)

{

    int size, chan, i, j, d, of, smpls, freq, quant, half_ch;

    uint16_t lc, rc;

    const uint8_t *as_pack;

    uint8_t *pcm, ipcm;



    as_pack = dv_extract_pack(frame, dv_audio_source);

    if (!as_pack)    

        return 0;



    smpls = as_pack[1]      & 0x3f; 

    freq  = as_pack[4] >> 3 & 0x07; 

    quant = as_pack[4]      & 0x07; 



    if (quant > 1)

        return -1;  



    if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency))

        return AVERROR_INVALIDDATA;



    size    = (sys->audio_min_samples[freq] + smpls) * 4; 

    half_ch = sys->difseg_size / 2;



    



    ipcm = (sys->height == 720 && !(frame[1] & 0x0C)) ? 2 : 0;



    

    for (chan = 0; chan < sys->n_difchan; chan++) {

        

        pcm = ppcm[ipcm++];

        if (!pcm)

            break;



        

        for (i = 0; i < sys->difseg_size; i++) {

            frame += 6 * 80; 

            if (quant == 1 && i == half_ch) {

                

                pcm = ppcm[ipcm++];

                if (!pcm)

                    break;

            }



            

            for (j = 0; j < 9; j++) {

                for (d = 8; d < 80; d += 2) {

                    if (quant == 0) {  

                        of = sys->audio_shuffle[i][j] +

                             (d - 8) / 2 * sys->audio_stride;

                        if (of * 2 >= size)

                            continue;



                        



                        pcm[of * 2]     = frame[d + 1];

                        pcm[of * 2 + 1] = frame[d];



                        if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)

                            pcm[of * 2 + 1] = 0;

                    } else {           

                        lc = ((uint16_t)frame[d]     << 4) |

                             ((uint16_t)frame[d + 2] >> 4);

                        rc = ((uint16_t)frame[d + 1] << 4) |

                             ((uint16_t)frame[d + 2] & 0x0f);

                        lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));

                        rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc));



                        of = sys->audio_shuffle[i % half_ch][j] +

                             (d - 8) / 3 * sys->audio_stride;

                        if (of * 2 >= size)

                            continue;



                        



                        pcm[of * 2]     = lc & 0xff;

                        pcm[of * 2 + 1] = lc >> 8;

                        of = sys->audio_shuffle[i % half_ch + half_ch][j] +

                             (d - 8) / 3 * sys->audio_stride;

                        



                        pcm[of * 2]     = rc & 0xff;

                        pcm[of * 2 + 1] = rc >> 8;

                        ++d;

                    }

                }



                frame += 16 * 80; 

            }

        }

    }



    return size;

}

static int dv_extract_audio(uint8_t *frame, uint8_t *ppcm[4],
static int FUNC_0(uint8_t *VAR_0, uint8_t *VAR_1[4],

                            const DVprofile *VAR_2)

{

    int VAR_3, VAR_4, VAR_5, VAR_6, VAR_7, VAR_8, VAR_9, VAR_10, VAR_11, VAR_12;

    uint16_t lc, rc;

    const uint8_t *VAR_13;

    uint8_t *pcm, ipcm;



    VAR_13 = dv_extract_pack(VAR_0, dv_audio_source);

    if (!VAR_13)    

        return 0;



    VAR_9 = VAR_13[1]      & 0x3f; 

    VAR_10  = VAR_13[4] >> 3 & 0x07; 

    VAR_11 = VAR_13[4]      & 0x07; 



    if (VAR_11 > 1)

        return -1;  



    if (VAR_10 >= FF_ARRAY_ELEMS(dv_audio_frequency))

        return AVERROR_INVALIDDATA;



    VAR_3    = (VAR_2->audio_min_samples[VAR_10] + VAR_9) * 4; 

    VAR_12 = VAR_2->difseg_size / 2;



    



    ipcm = (VAR_2->height == 720 && !(VAR_0[1] & 0x0C)) ? 2 : 0;



    

    for (VAR_4 = 0; VAR_4 < VAR_2->n_difchan; VAR_4++) {

        

        pcm = VAR_1[ipcm++];

        if (!pcm)

            break;



        

        for (VAR_5 = 0; VAR_5 < VAR_2->difseg_size; VAR_5++) {

            VAR_0 += 6 * 80; 

            if (VAR_11 == 1 && VAR_5 == VAR_12) {

                

                pcm = VAR_1[ipcm++];

                if (!pcm)

                    break;

            }



            

            for (VAR_6 = 0; VAR_6 < 9; VAR_6++) {

                for (VAR_7 = 8; VAR_7 < 80; VAR_7 += 2) {

                    if (VAR_11 == 0) {  

                        VAR_8 = VAR_2->audio_shuffle[VAR_5][VAR_6] +

                             (VAR_7 - 8) / 2 * VAR_2->audio_stride;

                        if (VAR_8 * 2 >= VAR_3)

                            continue;



                        



                        pcm[VAR_8 * 2]     = VAR_0[VAR_7 + 1];

                        pcm[VAR_8 * 2 + 1] = VAR_0[VAR_7];



                        if (pcm[VAR_8 * 2 + 1] == 0x80 && pcm[VAR_8 * 2] == 0x00)

                            pcm[VAR_8 * 2 + 1] = 0;

                    } else {           

                        lc = ((uint16_t)VAR_0[VAR_7]     << 4) |

                             ((uint16_t)VAR_0[VAR_7 + 2] >> 4);

                        rc = ((uint16_t)VAR_0[VAR_7 + 1] << 4) |

                             ((uint16_t)VAR_0[VAR_7 + 2] & 0x0f);

                        lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));

                        rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc));



                        VAR_8 = VAR_2->audio_shuffle[VAR_5 % VAR_12][VAR_6] +

                             (VAR_7 - 8) / 3 * VAR_2->audio_stride;

                        if (VAR_8 * 2 >= VAR_3)

                            continue;



                        



                        pcm[VAR_8 * 2]     = lc & 0xff;

                        pcm[VAR_8 * 2 + 1] = lc >> 8;

                        VAR_8 = VAR_2->audio_shuffle[VAR_5 % VAR_12 + VAR_12][VAR_6] +

                             (VAR_7 - 8) / 3 * VAR_2->audio_stride;

                        



                        pcm[VAR_8 * 2]     = rc & 0xff;

                        pcm[VAR_8 * 2 + 1] = rc >> 8;

                        ++VAR_7;

                    }

                }



                VAR_0 += 16 * 80; 

            }

        }

    }



    return VAR_3;

}

static int FUNC_0(uint8_t *VAR_0, uint8_t *VAR_1[4],
const DVprofile *VAR_2)
{ int VAR_3, VAR_4, VAR_5, VAR_6, VAR_7, VAR_8, VAR_9, VAR_10, VAR_11, VAR_12; uint16_t lc, rc; const uint8_t *VAR_13; uint8_t *pcm, ipcm; VAR_13 = dv_extract_pack(VAR_0, dv_audio_source); if (!VAR_13)
return 0; VAR_9 = VAR_13[1]      & 0x3f; VAR_10  = VAR_13[4] >> 3 & 0x07; VAR_11 = VAR_13[4]      & 0x07; if (VAR_11 > 1)
return -1; if (VAR_10 >= FF_ARRAY_ELEMS(dv_audio_frequency))
return AVERROR_INVALIDDATA; VAR_3    = (VAR_2->audio_min_samples[VAR_10] + VAR_9) * 4; VAR_12 = VAR_2->difseg_size / 2; ipcm = (VAR_2->height == 720 && !(VAR_0[1] & 0x0C)) ? 2 : 0; for (VAR_4 = 0; VAR_4 < VAR_2->n_difchan; VAR_4++) { pcm = VAR_1[ipcm++]; if (!pcm)
break; for (VAR_5 = 0; VAR_5 < VAR_2->difseg_size; VAR_5++) { VAR_0 += 6 * 80; if (VAR_11 == 1 && VAR_5 == VAR_12) { pcm = VAR_1[ipcm++]; if (!pcm)
break; } for (VAR_6 = 0; VAR_6 < 9; VAR_6++) { for (VAR_7 = 8; VAR_7 < 80; VAR_7 += 2) { if (VAR_11 == 0) { VAR_8 = VAR_2->audio_shuffle[VAR_5][VAR_6] +
(VAR_7 - 8) / 2 * VAR_2->audio_stride; if (VAR_8 * 2 >= VAR_3)
continue; pcm[VAR_8 * 2]     = VAR_0[VAR_7 + 1]; pcm[VAR_8 * 2 + 1] = VAR_0[VAR_7]; if (pcm[VAR_8 * 2 + 1] == 0x80 && pcm[VAR_8 * 2] == 0x00)
pcm[VAR_8 * 2 + 1] = 0; } else { lc = ((uint16_t)VAR_0[VAR_7]     << 4) |
((uint16_t)VAR_0[VAR_7 + 2] >> 4); rc = ((uint16_t)VAR_0[VAR_7 + 1] << 4) |
((uint16_t)VAR_0[VAR_7 + 2] & 0x0f); lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc)); rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc)); VAR_8 = VAR_2->audio_shuffle[VAR_5 % VAR_12][VAR_6] +
(VAR_7 - 8) / 3 * VAR_2->audio_stride; if (VAR_8 * 2 >= VAR_3)
continue; pcm[VAR_8 * 2]     = lc & 0xff; pcm[VAR_8 * 2 + 1] = lc >> 8; VAR_8 = VAR_2->audio_shuffle[VAR_5 % VAR_12 + VAR_12][VAR_6] +
(VAR_7 - 8) / 3 * VAR_2->audio_stride; pcm[VAR_8 * 2]     = rc & 0xff; pcm[VAR_8 * 2 + 1] = rc >> 8; ++VAR_7; } } VAR_0 += 16 * 80; } } } return VAR_3; }