mpegaudiodec.c

/* Reorder short blocks from bitstream order to interleaved order. It
   would be faster to do it in parsing, but the code would be far more
   complicated */
static void reorder_block(MPADecodeContext *s, GranuleDef *g)
{
    int i, j, len;
    INTFLOAT *ptr, *dst, *ptr1;
    INTFLOAT tmp[576];

    if (g->block_type != 2)
        return;

    if (g->switch_point) {
        if (s->sample_rate_index != 8)
            ptr = g->sb_hybrid + 36;
        else
            ptr = g->sb_hybrid + 72;
    } else {
        ptr = g->sb_hybrid;
    }

    for (i = g->short_start; i < 13; i++) {
        len  = band_size_short[s->sample_rate_index][i];
        ptr1 = ptr;
        dst  = tmp;
        for (j = len; j > 0; j--) {
            *dst++ = ptr[0*len];
            *dst++ = ptr[1*len];
            *dst++ = ptr[2*len];
            ptr++;
        }
        ptr += 2 * len;
        memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
    }
}

#define ISQRT2 FIXR(0.70710678118654752440)

static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
{
    int i, j, k, l;
    int sf_max, sf, len, non_zero_found;
    INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
    int non_zero_found_short[3];

    /* intensity stereo */
    if (s->mode_ext & MODE_EXT_I_STEREO) {
        if (!s->lsf) {
            is_tab = is_table;
            sf_max = 7;
        } else {
            is_tab = is_table_lsf[g1->scalefac_compress & 1];
            sf_max = 16;
        }

        tab0 = g0->sb_hybrid + 576;
        tab1 = g1->sb_hybrid + 576;

        non_zero_found_short[0] = 0;
        non_zero_found_short[1] = 0;
        non_zero_found_short[2] = 0;
        k = (13 - g1->short_start) * 3 + g1->long_end - 3;
        for (i = 12; i >= g1->short_start; i--) {
            /* for last band, use previous scale factor */
            if (i != 11)
                k -= 3;
            len = band_size_short[s->sample_rate_index][i];
            for (l = 2; l >= 0; l--) {
                tab0 -= len;
                tab1 -= len;
                if (!non_zero_found_short[l]) {
                    /* test if non zero band. if so, stop doing i-stereo */
                    for (j = 0; j < len; j++) {
                        if (tab1[j] != 0) {
                            non_zero_found_short[l] = 1;
                            goto found1;
                        }
                    }
                    sf = g1->scale_factors[k + l];
                    if (sf >= sf_max)
                        goto found1;

                    v1 = is_tab[0][sf];
                    v2 = is_tab[1][sf];
                    for (j = 0; j < len; j++) {
                        tmp0    = tab0[j];
                        tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
                        tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
                    }
                } else {
found1:
                    if (s->mode_ext & MODE_EXT_MS_STEREO) {
                        /* lower part of the spectrum : do ms stereo
                           if enabled */
                        for (j = 0; j < len; j++) {
                            tmp0    = tab0[j];
                            tmp1    = tab1[j];
                            tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
                            tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
                        }
                    }
                }
            }
        }

        non_zero_found = non_zero_found_short[0] |
                         non_zero_found_short[1] |
                         non_zero_found_short[2];

        for (i = g1->long_end - 1;i >= 0;i--) {
            len   = band_size_long[s->sample_rate_index][i];
            tab0 -= len;
            tab1 -= len;
            /* test if non zero band. if so, stop doing i-stereo */
            if (!non_zero_found) {
                for (j = 0; j < len; j++) {
                    if (tab1[j] != 0) {
                        non_zero_found = 1;
                        goto found2;
                    }
                }
                /* for last band, use previous scale factor */
                k  = (i == 21) ? 20 : i;
                sf = g1->scale_factors[k];
                if (sf >= sf_max)
                    goto found2;
                v1 = is_tab[0][sf];
                v2 = is_tab[1][sf];
                for (j = 0; j < len; j++) {
                    tmp0    = tab0[j];
                    tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
                    tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
                }
            } else {
found2:
                if (s->mode_ext & MODE_EXT_MS_STEREO) {
                    /* lower part of the spectrum : do ms stereo
                       if enabled */
                    for (j = 0; j < len; j++) {
                        tmp0    = tab0[j];
                        tmp1    = tab1[j];
                        tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
                        tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
                    }
                }
            }
        }
    } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
        /* ms stereo ONLY */
        /* NOTE: the 1/sqrt(2) normalization factor is included in the
           global gain */
#if CONFIG_FLOAT
       s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
#else
        tab0 = g0->sb_hybrid;
        tab1 = g1->sb_hybrid;
        for (i = 0; i < 576; i++) {
            tmp0    = tab0[i];
            tmp1    = tab1[i];
            tab0[i] = tmp0 + tmp1;
            tab1[i] = tmp0 - tmp1;
        }
#endif
    }
}

#if CONFIG_FLOAT
#define AA(j) do {                                                      \
        float tmp0 = ptr[-1-j];                                         \
        float tmp1 = ptr[   j];                                         \
        ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1];    \
        ptr[   j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0];    \
    } while (0)
#else
#define AA(j) do {                                              \
        int tmp0 = ptr[-1-j];                                   \
        int tmp1 = ptr[   j];                                   \
        int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]);          \
        ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2]));   \
        ptr[   j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3]));   \
    } while (0)
#endif

static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
{
    INTFLOAT *ptr;
    int n, i;

    /* we antialias only "long" bands */
    if (g->block_type == 2) {
        if (!g->switch_point)
            return;
        /* XXX: check this for 8000Hz case */
        n = 1;
    } else {
        n = SBLIMIT - 1;
    }

    ptr = g->sb_hybrid + 18;
    for (i = n; i > 0; i--) {
        AA(0);
        AA(1);
        AA(2);
        AA(3);
        AA(4);
        AA(5);
        AA(6);
        AA(7);

        ptr += 18;
    }
}

static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
                          INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
{
    INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
    INTFLOAT out2[12];
    int i, j, mdct_long_end, sblimit;

    /* find last non zero block */
    ptr  = g->sb_hybrid + 576;
    ptr1 = g->sb_hybrid + 2 * 18;
    while (ptr >= ptr1) {
        int32_t *p;
        ptr -= 6;
        p    = (int32_t*)ptr;
        if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
            break;
    }
    sblimit = ((ptr - g->sb_hybrid) / 18) + 1;

    if (g->block_type == 2) {
        /* XXX: check for 8000 Hz */
        if (g->switch_point)
            mdct_long_end = 2;
        else
            mdct_long_end = 0;
    } else {
        mdct_long_end = sblimit;
    }

    s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
                                     mdct_long_end, g->switch_point,
                                     g->block_type);

    buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
    ptr = g->sb_hybrid + 18 * mdct_long_end;

    for (j = mdct_long_end; j < sblimit; j++) {
        /* select frequency inversion */
        win     = RENAME(ff_mdct_win)[2 + (4  & -(j & 1))];
        out_ptr = sb_samples + j;

        for (i = 0; i < 6; i++) {
            *out_ptr = buf[4*i];
            out_ptr += SBLIMIT;
        }
        imdct12(out2, ptr + 0);
        for (i = 0; i < 6; i++) {
            *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*1)];
            buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
            out_ptr += SBLIMIT;
        }
        imdct12(out2, ptr + 1);
        for (i = 0; i < 6; i++) {
            *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*2)];
            buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
            out_ptr += SBLIMIT;
        }
        imdct12(out2, ptr + 2);
        for (i = 0; i < 6; i++) {
            buf[4*(i + 6*0)] = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*0)];
            buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
            buf[4*(i + 6*2)] = 0;
        }
        ptr += 18;
        buf += (j&3) != 3 ? 1 : (4*18-3);
    }
    /* zero bands */
    for (j = sblimit; j < SBLIMIT; j++) {
        /* overlap */
        out_ptr = sb_samples + j;
        for (i = 0; i < 18; i++) {
            *out_ptr = buf[4*i];
            buf[4*i]   = 0;
            out_ptr += SBLIMIT;
        }
        buf += (j&3) != 3 ? 1 : (4*18-3);
    }
}

/* main layer3 decoding function */
static int mp_decode_layer3(MPADecodeContext *s)
{
    int nb_granules, main_data_begin;
    int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
    GranuleDef *g;
    int16_t exponents[576]; //FIXME try INTFLOAT

    /* read side info */
    if (s->lsf) {
        main_data_begin = get_bits(&s->gb, 8);
        skip_bits(&s->gb, s->nb_channels);
        nb_granules = 1;
    } else {
        main_data_begin = get_bits(&s->gb, 9);
        if (s->nb_channels == 2)
            skip_bits(&s->gb, 3);
        else
            skip_bits(&s->gb, 5);
        nb_granules = 2;
        for (ch = 0; ch < s->nb_channels; ch++) {
            s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
            s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
        }
    }

    for (gr = 0; gr < nb_granules; gr++) {
        for (ch = 0; ch < s->nb_channels; ch++) {
            av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
            g = &s->granules[ch][gr];
            g->part2_3_length = get_bits(&s->gb, 12);
            g->big_values     = get_bits(&s->gb,  9);
            if (g->big_values > 288) {
                av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
                return AVERROR_INVALIDDATA;
            }

            g->global_gain = get_bits(&s->gb, 8);
            /* if MS stereo only is selected, we precompute the
               1/sqrt(2) renormalization factor */
            if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
                MODE_EXT_MS_STEREO)
                g->global_gain -= 2;
            if (s->lsf)
                g->scalefac_compress = get_bits(&s->gb, 9);
            else
                g->scalefac_compress = get_bits(&s->gb, 4);
            blocksplit_flag = get_bits1(&s->gb);
            if (blocksplit_flag) {
                g->block_type = get_bits(&s->gb, 2);
                if (g->block_type == 0) {
                    av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
                    return AVERROR_INVALIDDATA;
                }
                g->switch_point = get_bits1(&s->gb);
                for (i = 0; i < 2; i++)
                    g->table_select[i] = get_bits(&s->gb, 5);
                for (i = 0; i < 3; i++)
                    g->subblock_gain[i] = get_bits(&s->gb, 3);
                ff_init_short_region(s, g);
            } else {
                int region_address1, region_address2;
                g->block_type = 0;
                g->switch_point = 0;
                for (i = 0; i < 3; i++)
                    g->table_select[i] = get_bits(&s->gb, 5);
                /* compute huffman coded region sizes */
                region_address1 = get_bits(&s->gb, 4);
                region_address2 = get_bits(&s->gb, 3);
                av_dlog(s->avctx, "region1=%d region2=%d\n",
                        region_address1, region_address2);
                ff_init_long_region(s, g, region_address1, region_address2);
            }
            ff_region_offset2size(g);
            ff_compute_band_indexes(s, g);

            g->preflag = 0;
            if (!s->lsf)
                g->preflag = get_bits1(&s->gb);
            g->scalefac_scale     = get_bits1(&s->gb);
            g->count1table_select = get_bits1(&s->gb);
            av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
                    g->block_type, g->switch_point);
        }
    }

    if (!s->adu_mode) {
        int skip;
        const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
        int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
                                FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
        assert((get_bits_count(&s->gb) & 7) == 0);
        /* now we get bits from the main_data_begin offset */
        av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
                main_data_begin, s->last_buf_size);

        memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
        s->in_gb = s->gb;
        init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
#if !UNCHECKED_BITSTREAM_READER
        s->gb.size_in_bits_plus8 += extrasize * 8;
#endif
        s->last_buf_size <<= 3;
        for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
            for (ch = 0; ch < s->nb_channels; ch++) {
                g = &s->granules[ch][gr];
                s->last_buf_size += g->part2_3_length;
                memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
                compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
            }
        }
        skip = s->last_buf_size - 8 * main_data_begin;
        if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
            skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
            s->gb           = s->in_gb;
            s->in_gb.buffer = NULL;
        } else {
            skip_bits_long(&s->gb, skip);
        }
    } else {
        gr = 0;
    }

    for (; gr < nb_granules; gr++) {
        for (ch = 0; ch < s->nb_channels; ch++) {
            g = &s->granules[ch][gr];
            bits_pos = get_bits_count(&s->gb);

            if (!s->lsf) {
                uint8_t *sc;
                int slen, slen1, slen2;

                /* MPEG1 scale factors */
                slen1 = slen_table[0][g->scalefac_compress];
                slen2 = slen_table[1][g->scalefac_compress];
                av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
                if (g->block_type == 2) {
                    n = g->switch_point ? 17 : 18;
                    j = 0;
                    if (slen1) {
                        for (i = 0; i < n; i++)
                            g->scale_factors[j++] = get_bits(&s->gb, slen1);
                    } else {
                        for (i = 0; i < n; i++)
                            g->scale_factors[j++] = 0;
                    }
                    if (slen2) {
                        for (i = 0; i < 18; i++)
                            g->scale_factors[j++] = get_bits(&s->gb, slen2);
                        for (i = 0; i < 3; i++)
                            g->scale_factors[j++] = 0;
                    } else {
                        for (i = 0; i < 21; i++)
                            g->scale_factors[j++] = 0;
                    }
                } else {
                    sc = s->granules[ch][0].scale_factors;
                    j = 0;
                    for (k = 0; k < 4; k++) {
                        n = k == 0 ? 6 : 5;
                        if ((g->scfsi & (0x8 >> k)) == 0) {
                            slen = (k < 2) ? slen1 : slen2;
                            if (slen) {
                                for (i = 0; i < n; i++)
                                    g->scale_factors[j++] = get_bits(&s->gb, slen);
                            } else {
                                for (i = 0; i < n; i++)
                                    g->scale_factors[j++] = 0;
                            }
                        } else {
                            /* simply copy from last granule */
                            for (i = 0; i < n; i++) {
                                g->scale_factors[j] = sc[j];
                                j++;
                            }
                        }
                    }
                    g->scale_factors[j++] = 0;
                }
            } else {
                int tindex, tindex2, slen[4], sl, sf;

                /* LSF scale factors */
                if (g->block_type == 2)
                    tindex = g->switch_point ? 2 : 1;
                else
                    tindex = 0;

                sf = g->scalefac_compress;
                if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
                    /* intensity stereo case */
                    sf >>= 1;
                    if (sf < 180) {
                        lsf_sf_expand(slen, sf, 6, 6, 0);
                        tindex2 = 3;
                    } else if (sf < 244) {
                        lsf_sf_expand(slen, sf - 180, 4, 4, 0);
                        tindex2 = 4;
                    } else {
                        lsf_sf_expand(slen, sf - 244, 3, 0, 0);
                        tindex2 = 5;
                    }
                } else {
                    /* normal case */
                    if (sf < 400) {
                        lsf_sf_expand(slen, sf, 5, 4, 4);
                        tindex2 = 0;
                    } else if (sf < 500) {
                        lsf_sf_expand(slen, sf - 400, 5, 4, 0);
                        tindex2 = 1;
                    } else {
                        lsf_sf_expand(slen, sf - 500, 3, 0, 0);
                        tindex2 = 2;
                        g->preflag = 1;
                    }
                }

                j = 0;
                for (k = 0; k < 4; k++) {
                    n  = lsf_nsf_table[tindex2][tindex][k];
                    sl = slen[k];
                    if (sl) {
                        for (i = 0; i < n; i++)
                            g->scale_factors[j++] = get_bits(&s->gb, sl);
                    } else {
                        for (i = 0; i < n; i++)
                            g->scale_factors[j++] = 0;
                    }
                }
                /* XXX: should compute exact size */
                for (; j < 40; j++)
                    g->scale_factors[j] = 0;
            }

            exponents_from_scale_factors(s, g, exponents);

            /* read Huffman coded residue */
            huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
        } /* ch */

        if (s->mode == MPA_JSTEREO)
            compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);

        for (ch = 0; ch < s->nb_channels; ch++) {
            g = &s->granules[ch][gr];

            reorder_block(s, g);
            compute_antialias(s, g);
            compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
        }
    } /* gr */
    if (get_bits_count(&s->gb) < 0)
        skip_bits_long(&s->gb, -get_bits_count(&s->gb));
    return nb_granules * 18;
}

static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
                           const uint8_t *buf, int buf_size)
{
    int i, nb_frames, ch, ret;
    OUT_INT *samples_ptr;

    init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);

    /* skip error protection field */
    if (s->error_protection)
        skip_bits(&s->gb, 16);

    switch(s->layer) {
    case 1:
        s->avctx->frame_size = 384;
        nb_frames = mp_decode_layer1(s);
        break;
    case 2:
        s->avctx->frame_size = 1152;
        nb_frames = mp_decode_layer2(s);
        break;
    case 3:
        s->avctx->frame_size = s->lsf ? 576 : 1152;
    default:
        nb_frames = mp_decode_layer3(s);

        if (nb_frames < 0)
            return nb_frames;

        s->last_buf_size=0;
        if (s->in_gb.buffer) {
            align_get_bits(&s->gb);
            i = get_bits_left(&s->gb)>>3;
            if (i >= 0 && i <= BACKSTEP_SIZE) {
                memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
                s->last_buf_size=i;
            } else
                av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
            s->gb           = s->in_gb;
            s->in_gb.buffer = NULL;
        }

        align_get_bits(&s->gb);
        assert((get_bits_count(&s->gb) & 7) == 0);
        i = get_bits_left(&s->gb) >> 3;

        if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
            if (i < 0)
                av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
            i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
        }
        assert(i <= buf_size - HEADER_SIZE && i >= 0);
        memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
        s->last_buf_size += i;
    }

    /* get output buffer */
    if (!samples) {
        s->frame.nb_samples = s->avctx->frame_size;
        if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {
            av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
            return ret;
        }
        samples = (OUT_INT *)s->frame.data[0];
    }

    /* apply the synthesis filter */
    for (ch = 0; ch < s->nb_channels; ch++) {
        samples_ptr = samples + ch;
        for (i = 0; i < nb_frames; i++) {
            RENAME(ff_mpa_synth_filter)(
                         &s->mpadsp,
                         s->synth_buf[ch], &(s->synth_buf_offset[ch]),
                         RENAME(ff_mpa_synth_window), &s->dither_state,
                         samples_ptr, s->nb_channels,
                         s->sb_samples[ch][i]);
            samples_ptr += 32 * s->nb_channels;
        }
    }

    return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}

static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
                        AVPacket *avpkt)
{
    const uint8_t *buf  = avpkt->data;
    int buf_size        = avpkt->size;
    MPADecodeContext *s = avctx->priv_data;
    uint32_t header;
    int ret;

    if (buf_size < HEADER_SIZE)
        return AVERROR_INVALIDDATA;

    header = AV_RB32(buf);
    if (ff_mpa_check_header(header) < 0) {
        av_log(avctx, AV_LOG_ERROR, "Header missing\n");
        return AVERROR_INVALIDDATA;
    }

    if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
        /* free format: prepare to compute frame size */
        s->frame_size = -1;
        return AVERROR_INVALIDDATA;
    }
    /* update codec info */
    avctx->channels       = s->nb_channels;
    avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
    if (!avctx->bit_rate)
        avctx->bit_rate = s->bit_rate;

    if (s->frame_size <= 0 || s->frame_size > buf_size) {
        av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
        return AVERROR_INVALIDDATA;
    } else if (s->frame_size < buf_size) {
        buf_size= s->frame_size;
    }

    ret = mp_decode_frame(s, NULL, buf, buf_size);
    if (ret >= 0) {
        *got_frame_ptr   = 1;
        *(AVFrame *)data = s->frame;
        avctx->sample_rate = s->sample_rate;
        //FIXME maybe move the other codec info stuff from above here too
    } else {
        av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
        /* Only return an error if the bad frame makes up the whole packet or
         * the error is related to buffer management.
         * If there is more data in the packet, just consume the bad frame
         * instead of returning an error, which would discard the whole
         * packet. */
        *got_frame_ptr = 0;
        if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
            return ret;
    }
    s->frame_size = 0;
    return buf_size;
}

static void mp_flush(MPADecodeContext *ctx)
{
    memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
    ctx->last_buf_size = 0;
}

static void flush(AVCodecContext *avctx)
{
    mp_flush(avctx->priv_data);
}

#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
static int decode_frame_adu(AVCodecContext *avctx, void *data,
                            int *got_frame_ptr, AVPacket *avpkt)
{
    const uint8_t *buf  = avpkt->data;
    int buf_size        = avpkt->size;
    MPADecodeContext *s = avctx->priv_data;
    uint32_t header;
    int len, ret;

    len = buf_size;

    // Discard too short frames
    if (buf_size < HEADER_SIZE) {
        av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
        return AVERROR_INVALIDDATA;
    }


    if (len > MPA_MAX_CODED_FRAME_SIZE)
        len = MPA_MAX_CODED_FRAME_SIZE;

    // Get header and restore sync word
    header = AV_RB32(buf) | 0xffe00000;

    if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
        av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
        return AVERROR_INVALIDDATA;
    }

    avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
    /* update codec info */
    avctx->sample_rate = s->sample_rate;
    avctx->channels    = s->nb_channels;
    if (!avctx->bit_rate)
        avctx->bit_rate = s->bit_rate;

    s->frame_size = len;

    ret = mp_decode_frame(s, NULL, buf, buf_size);
    if (ret < 0) {
        av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
        return ret;
    }

    *got_frame_ptr   = 1;
    *(AVFrame *)data = s->frame;

    return buf_size;
}
#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */

#if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER

/**
 * Context for MP3On4 decoder
 */
typedef struct MP3On4DecodeContext {
    AVFrame *frame;
    int frames;                     ///< number of mp3 frames per block (number of mp3 decoder instances)
    int syncword;                   ///< syncword patch
    const uint8_t *coff;            ///< channel offsets in output buffer
    MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
    OUT_INT *decoded_buf;           ///< output buffer for decoded samples
} MP3On4DecodeContext;

#include "mpeg4audio.h"

/* Next 3 arrays are indexed by channel config number (passed via codecdata) */

/* number of mp3 decoder instances */
static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };

/* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
static const uint8_t chan_offset[8][5] = {
    { 0             },
    { 0             },  // C
    { 0             },  // FLR
    { 2, 0          },  // C FLR
    { 2, 0, 3       },  // C FLR BS
    { 2, 0, 3       },  // C FLR BLRS
    { 2, 0, 4, 3    },  // C FLR BLRS LFE
    { 2, 0, 6, 4, 3 },  // C FLR BLRS BLR LFE
};

/* mp3on4 channel layouts */
static const int16_t chan_layout[8] = {
    0,
    AV_CH_LAYOUT_MONO,
    AV_CH_LAYOUT_STEREO,
    AV_CH_LAYOUT_SURROUND,
    AV_CH_LAYOUT_4POINT0,
    AV_CH_LAYOUT_5POINT0,
    AV_CH_LAYOUT_5POINT1,
    AV_CH_LAYOUT_7POINT1
};

static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
{
    MP3On4DecodeContext *s = avctx->priv_data;
    int i;

    for (i = 0; i < s->frames; i++)
        av_free(s->mp3decctx[i]);

    av_freep(&s->decoded_buf);

    return 0;
}


static int decode_init_mp3on4(AVCodecContext * avctx)
{
    MP3On4DecodeContext *s = avctx->priv_data;
    MPEG4AudioConfig cfg;
    int i;

    if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
        av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
        return AVERROR_INVALIDDATA;
    }

    avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
                                 avctx->extradata_size * 8, 1);
    if (!cfg.chan_config || cfg.chan_config > 7) {
        av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
        return AVERROR_INVALIDDATA;
    }
    s->frames             = mp3Frames[cfg.chan_config];
    s->coff               = chan_offset[cfg.chan_config];
    avctx->channels       = ff_mpeg4audio_channels[cfg.chan_config];
    avctx->channel_layout = chan_layout[cfg.chan_config];

    if (cfg.sample_rate < 16000)
        s->syncword = 0xffe00000;
    else
        s->syncword = 0xfff00000;

    /* Init the first mp3 decoder in standard way, so that all tables get builded
     * We replace avctx->priv_data with the context of the first decoder so that
     * decode_init() does not have to be changed.
     * Other decoders will be initialized here copying data from the first context
     */
    // Allocate zeroed memory for the first decoder context
    s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
    if (!s->mp3decctx[0])
        goto alloc_fail;
    // Put decoder context in place to make init_decode() happy
    avctx->priv_data = s->mp3decctx[0];
    decode_init(avctx);
    s->frame = avctx->coded_frame;
    // Restore mp3on4 context pointer
    avctx->priv_data = s;
    s->mp3decctx[0]->adu_mode = 1; // Set adu mode

    /* Create a separate codec/context for each frame (first is already ok).
     * Each frame is 1 or 2 channels - up to 5 frames allowed
     */
    for (i = 1; i < s->frames; i++) {
        s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
        if (!s->mp3decctx[i])
            goto alloc_fail;
        s->mp3decctx[i]->adu_mode = 1;
        s->mp3decctx[i]->avctx = avctx;
        s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
    }

    /* Allocate buffer for multi-channel output if needed */
    if (s->frames > 1) {
        s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
                                   sizeof(*s->decoded_buf));
        if (!s->decoded_buf)
            goto alloc_fail;
    }

    return 0;
alloc_fail:
    decode_close_mp3on4(avctx);
    return AVERROR(ENOMEM);
}


static void flush_mp3on4(AVCodecContext *avctx)
{
    int i;
    MP3On4DecodeContext *s = avctx->priv_data;

    for (i = 0; i < s->frames; i++)
        mp_flush(s->mp3decctx[i]);
}


static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
{
    const uint8_t *buf     = avpkt->data;
    int buf_size           = avpkt->size;
    MP3On4DecodeContext *s = avctx->priv_data;
    MPADecodeContext *m;
    int fsize, len = buf_size, out_size = 0;
    uint32_t header;
    OUT_INT *out_samples;
    OUT_INT *outptr, *bp;
    int fr, j, n, ch, ret;

    /* get output buffer */
    s->frame->nb_samples = MPA_FRAME_SIZE;
    if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }
    out_samples = (OUT_INT *)s->frame->data[0];

    // Discard too short frames
    if (buf_size < HEADER_SIZE)
        return AVERROR_INVALIDDATA;

    // If only one decoder interleave is not needed
    outptr = s->frames == 1 ? out_samples : s->decoded_buf;

    avctx->bit_rate = 0;

    ch = 0;
    for (fr = 0; fr < s->frames; fr++) {
        fsize = AV_RB16(buf) >> 4;
        fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
        m     = s->mp3decctx[fr];
        assert(m != NULL);

        if (fsize < HEADER_SIZE) {
            av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
            return AVERROR_INVALIDDATA;
        }
        header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header

        if (ff_mpa_check_header(header) < 0) // Bad header, discard block
            break;

        avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);

        if (ch + m->nb_channels > avctx->channels) {
            av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
                                        "channel count\n");
            return AVERROR_INVALIDDATA;
        }
        ch += m->nb_channels;

        if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
            return ret;

        out_size += ret;
        buf      += fsize;
        len      -= fsize;

        if (s->frames > 1) {
            n = m->avctx->frame_size*m->nb_channels;
            /* interleave output data */
            bp = out_samples + s->coff[fr];
            if (m->nb_channels == 1) {
                for (j = 0; j < n; j++) {
                    *bp = s->decoded_buf[j];
                    bp += avctx->channels;
                }
            } else {
                for (j = 0; j < n; j++) {
                    bp[0] = s->decoded_buf[j++];
                    bp[1] = s->decoded_buf[j];
                    bp   += avctx->channels;
                }
            }
        }
        avctx->bit_rate += m->bit_rate;
    }

    /* update codec info */
    avctx->sample_rate = s->mp3decctx[0]->sample_rate;

    s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
    *got_frame_ptr   = 1;
    *(AVFrame *)data = *s->frame;

    return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */

#if !CONFIG_FLOAT
#if CONFIG_MP1_DECODER
AVCodec ff_mp1_decoder = {
    .name           = "mp1",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_MP1,
    .priv_data_size = sizeof(MPADecodeContext),
    .init           = decode_init,
    .decode         = decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .flush          = flush,
    .long_name      = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
};
#endif
#if CONFIG_MP2_DECODER
AVCodec ff_mp2_decoder = {