mpegaudiodec: Consistently handle fixed/float templating

1 files changed, 1947 insertions, 0 deletions

diff --git a/libavcodec/mpegaudiodec_template.c b/libavcodec/mpegaudiodec_template.c
new file mode 100644
index 0000000000..9427dbfc55
--- /dev/null
+++ b/libavcodec/mpegaudiodec_template.c
@@ -0,0 +1,1947 @@
+/*
+ * MPEG Audio decoder
+ * Copyright (c) 2001, 2002 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * MPEG Audio decoder
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "mathops.h"
+#include "mpegaudiodsp.h"
+
+/*
+ * TODO:
+ *  - test lsf / mpeg25 extensively.
+ */
+
+#include "mpegaudio.h"
+#include "mpegaudiodecheader.h"
+
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+#define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
+
+/* layer 3 "granule" */
+typedef struct GranuleDef {
+    uint8_t scfsi;
+    int part2_3_length;
+    int big_values;
+    int global_gain;
+    int scalefac_compress;
+    uint8_t block_type;
+    uint8_t switch_point;
+    int table_select[3];
+    int subblock_gain[3];
+    uint8_t scalefac_scale;
+    uint8_t count1table_select;
+    int region_size[3]; /* number of huffman codes in each region */
+    int preflag;
+    int short_start, long_end; /* long/short band indexes */
+    uint8_t scale_factors[40];
+    DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
+} GranuleDef;
+
+typedef struct MPADecodeContext {
+    MPA_DECODE_HEADER
+    uint8_t last_buf[LAST_BUF_SIZE];
+    int last_buf_size;
+    /* next header (used in free format parsing) */
+    uint32_t free_format_next_header;
+    GetBitContext gb;
+    GetBitContext in_gb;
+    DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
+    int synth_buf_offset[MPA_MAX_CHANNELS];
+    DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
+    INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+    GranuleDef granules[2][2]; /* Used in Layer 3 */
+    int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+    int dither_state;
+    int err_recognition;
+    AVCodecContext* avctx;
+    MPADSPContext mpadsp;
+    AVFloatDSPContext fdsp;
+    AVFrame *frame;
+} MPADecodeContext;
+
+#define HEADER_SIZE 4
+
+#include "mpegaudiodata.h"
+#include "mpegaudiodectab.h"
+
+/* vlc structure for decoding layer 3 huffman tables */
+static VLC huff_vlc[16];
+static VLC_TYPE huff_vlc_tables[
+    0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
+  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
+  ][2];
+static const int huff_vlc_tables_sizes[16] = {
+    0,  128,  128,  128,  130,  128,  154,  166,
+  142,  204,  190,  170,  542,  460,  662,  414
+};
+static VLC huff_quad_vlc[2];
+static VLC_TYPE  huff_quad_vlc_tables[128+16][2];
+static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
+/* computed from band_size_long */
+static uint16_t band_index_long[9][23];
+#include "mpegaudio_tablegen.h"
+/* intensity stereo coef table */
+static INTFLOAT is_table[2][16];
+static INTFLOAT is_table_lsf[2][2][16];
+static INTFLOAT csa_table[8][4];
+
+static int16_t division_tab3[1<<6 ];
+static int16_t division_tab5[1<<8 ];
+static int16_t division_tab9[1<<11];
+
+static int16_t * const division_tabs[4] = {
+    division_tab3, division_tab5, NULL, division_tab9
+};
+
+/* lower 2 bits: modulo 3, higher bits: shift */
+static uint16_t scale_factor_modshift[64];
+/* [i][j]:  2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
+static int32_t scale_factor_mult[15][3];
+/* mult table for layer 2 group quantization */
+
+#define SCALE_GEN(v) \
+{ FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
+
+static const int32_t scale_factor_mult2[3][3] = {
+    SCALE_GEN(4.0 / 3.0), /* 3 steps */
+    SCALE_GEN(4.0 / 5.0), /* 5 steps */
+    SCALE_GEN(4.0 / 9.0), /* 9 steps */
+};
+
+/**
+ * Convert region offsets to region sizes and truncate
+ * size to big_values.
+ */
+static void region_offset2size(GranuleDef *g)
+{
+    int i, k, j = 0;
+    g->region_size[2] = 576 / 2;
+    for (i = 0; i < 3; i++) {
+        k = FFMIN(g->region_size[i], g->big_values);
+        g->region_size[i] = k - j;
+        j = k;
+    }
+}
+
+static void init_short_region(MPADecodeContext *s, GranuleDef *g)
+{
+    if (g->block_type == 2) {
+        if (s->sample_rate_index != 8)
+            g->region_size[0] = (36 / 2);
+        else
+            g->region_size[0] = (72 / 2);
+    } else {
+        if (s->sample_rate_index <= 2)
+            g->region_size[0] = (36 / 2);
+        else if (s->sample_rate_index != 8)
+            g->region_size[0] = (54 / 2);
+        else
+            g->region_size[0] = (108 / 2);
+    }
+    g->region_size[1] = (576 / 2);
+}
+
+static void init_long_region(MPADecodeContext *s, GranuleDef *g,
+                             int ra1, int ra2)
+{
+    int l;
+    g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
+    /* should not overflow */
+    l = FFMIN(ra1 + ra2 + 2, 22);
+    g->region_size[1] = band_index_long[s->sample_rate_index][      l] >> 1;
+}
+
+static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
+{
+    if (g->block_type == 2) {
+        if (g->switch_point) {
+            /* if switched mode, we handle the 36 first samples as
+                long blocks.  For 8000Hz, we handle the 72 first
+                exponents as long blocks */
+            if (s->sample_rate_index <= 2)
+                g->long_end = 8;
+            else
+                g->long_end = 6;
+
+            g->short_start = 3;
+        } else {
+            g->long_end    = 0;
+            g->short_start = 0;
+        }
+    } else {
+        g->short_start = 13;
+        g->long_end    = 22;
+    }
+}
+
+/* layer 1 unscaling */
+/* n = number of bits of the mantissa minus 1 */
+static inline int l1_unscale(int n, int mant, int scale_factor)
+{
+    int shift, mod;
+    int64_t val;
+
+    shift   = scale_factor_modshift[scale_factor];
+    mod     = shift & 3;
+    shift >>= 2;
+    val     = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
+    shift  += n;
+    /* NOTE: at this point, 1 <= shift >= 21 + 15 */
+    return (int)((val + (1LL << (shift - 1))) >> shift);
+}
+
+static inline int l2_unscale_group(int steps, int mant, int scale_factor)
+{
+    int shift, mod, val;
+
+    shift   = scale_factor_modshift[scale_factor];
+    mod     = shift & 3;
+    shift >>= 2;
+
+    val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
+    /* NOTE: at this point, 0 <= shift <= 21 */
+    if (shift > 0)
+        val = (val + (1 << (shift - 1))) >> shift;
+    return val;
+}
+
+/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
+static inline int l3_unscale(int value, int exponent)
+{
+    unsigned int m;
+    int e;
+
+    e  = table_4_3_exp  [4 * value + (exponent & 3)];
+    m  = table_4_3_value[4 * value + (exponent & 3)];
+    e -= exponent >> 2;
+    assert(e >= 1);
+    if (e > 31)
+        return 0;
+    m = (m + (1 << (e - 1))) >> e;
+
+    return m;
+}
+
+static av_cold void decode_init_static(void)
+{
+    int i, j, k;
+    int offset;
+
+    /* scale factors table for layer 1/2 */
+    for (i = 0; i < 64; i++) {
+        int shift, mod;
+        /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
+        shift = i / 3;
+        mod   = i % 3;
+        scale_factor_modshift[i] = mod | (shift << 2);
+    }
+
+    /* scale factor multiply for layer 1 */
+    for (i = 0; i < 15; i++) {
+        int n, norm;
+        n = i + 2;
+        norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
+        scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0          * 2.0), FRAC_BITS);
+        scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
+        scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
+        av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
+                scale_factor_mult[i][0],
+                scale_factor_mult[i][1],
+                scale_factor_mult[i][2]);
+    }
+
+    RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
+
+    /* huffman decode tables */
+    offset = 0;
+    for (i = 1; i < 16; i++) {
+        const HuffTable *h = &mpa_huff_tables[i];
+        int xsize, x, y;
+        uint8_t  tmp_bits [512] = { 0 };
+        uint16_t tmp_codes[512] = { 0 };
+
+        xsize = h->xsize;
+
+        j = 0;
+        for (x = 0; x < xsize; x++) {
+            for (y = 0; y < xsize; y++) {
+                tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j  ];
+                tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
+            }
+        }
+
+        /* XXX: fail test */
+        huff_vlc[i].table = huff_vlc_tables+offset;
+        huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
+        init_vlc(&huff_vlc[i], 7, 512,
+                 tmp_bits, 1, 1, tmp_codes, 2, 2,
+                 INIT_VLC_USE_NEW_STATIC);
+        offset += huff_vlc_tables_sizes[i];
+    }
+    assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
+
+    offset = 0;
+    for (i = 0; i < 2; i++) {
+        huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
+        huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
+        init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
+                 mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
+                 INIT_VLC_USE_NEW_STATIC);
+        offset += huff_quad_vlc_tables_sizes[i];
+    }
+    assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
+
+    for (i = 0; i < 9; i++) {
+        k = 0;
+        for (j = 0; j < 22; j++) {
+            band_index_long[i][j] = k;
+            k += band_size_long[i][j];
+        }
+        band_index_long[i][22] = k;
+    }
+
+    /* compute n ^ (4/3) and store it in mantissa/exp format */
+
+    mpegaudio_tableinit();
+
+    for (i = 0; i < 4; i++) {
+        if (ff_mpa_quant_bits[i] < 0) {
+            for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
+                int val1, val2, val3, steps;
+                int val = j;
+                steps   = ff_mpa_quant_steps[i];
+                val1    = val % steps;
+                val    /= steps;
+                val2    = val % steps;
+                val3    = val / steps;
+                division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
+            }
+        }
+    }
+
+
+    for (i = 0; i < 7; i++) {
+        float f;
+        INTFLOAT v;
+        if (i != 6) {
+            f = tan((double)i * M_PI / 12.0);
+            v = FIXR(f / (1.0 + f));
+        } else {
+            v = FIXR(1.0);
+        }
+        is_table[0][    i] = v;
+        is_table[1][6 - i] = v;
+    }
+    /* invalid values */
+    for (i = 7; i < 16; i++)
+        is_table[0][i] = is_table[1][i] = 0.0;
+
+    for (i = 0; i < 16; i++) {
+        double f;
+        int e, k;
+
+        for (j = 0; j < 2; j++) {
+            e = -(j + 1) * ((i + 1) >> 1);
+            f = pow(2.0, e / 4.0);
+            k = i & 1;
+            is_table_lsf[j][k ^ 1][i] = FIXR(f);
+            is_table_lsf[j][k    ][i] = FIXR(1.0);
+            av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
+                    i, j, (float) is_table_lsf[j][0][i],
+                    (float) is_table_lsf[j][1][i]);
+        }
+    }
+
+    for (i = 0; i < 8; i++) {
+        float ci, cs, ca;
+        ci = ci_table[i];
+        cs = 1.0 / sqrt(1.0 + ci * ci);
+        ca = cs * ci;
+#if !CONFIG_FLOAT
+        csa_table[i][0] = FIXHR(cs/4);
+        csa_table[i][1] = FIXHR(ca/4);
+        csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
+        csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
+#else
+        csa_table[i][0] = cs;
+        csa_table[i][1] = ca;
+        csa_table[i][2] = ca + cs;
+        csa_table[i][3] = ca - cs;
+#endif
+    }
+}
+
+static av_cold int decode_init(AVCodecContext * avctx)
+{
+    static int initialized_tables = 0;
+    MPADecodeContext *s = avctx->priv_data;
+
+    if (!initialized_tables) {
+        decode_init_static();
+        initialized_tables = 1;
+    }
+
+    s->avctx = avctx;
+
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+    ff_mpadsp_init(&s->mpadsp);
+
+    if (avctx->request_sample_fmt == OUT_FMT &&
+        avctx->codec_id != AV_CODEC_ID_MP3ON4)
+        avctx->sample_fmt = OUT_FMT;
+    else
+        avctx->sample_fmt = OUT_FMT_P;
+    s->err_recognition = avctx->err_recognition;
+
+    if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
+        s->adu_mode = 1;
+
+    return 0;
+}
+
+#define C3 FIXHR(0.86602540378443864676/2)
+#define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
+#define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
+#define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
+
+/* 12 points IMDCT. We compute it "by hand" by factorizing obvious
+   cases. */
+static void imdct12(INTFLOAT *out, INTFLOAT *in)
+{
+    INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
+
+    in0  = in[0*3];
+    in1  = in[1*3] + in[0*3];
+    in2  = in[2*3] + in[1*3];
+    in3  = in[3*3] + in[2*3];
+    in4  = in[4*3] + in[3*3];
+    in5  = in[5*3] + in[4*3];
+    in5 += in3;
+    in3 += in1;
+
+    in2  = MULH3(in2, C3, 2);
+    in3  = MULH3(in3, C3, 4);
+
+    t1   = in0 - in4;
+    t2   = MULH3(in1 - in5, C4, 2);
+
+    out[ 7] =
+    out[10] = t1 + t2;
+    out[ 1] =
+    out[ 4] = t1 - t2;
+
+    in0    += SHR(in4, 1);
+    in4     = in0 + in2;
+    in5    += 2*in1;
+    in1     = MULH3(in5 + in3, C5, 1);
+    out[ 8] =
+    out[ 9] = in4 + in1;
+    out[ 2] =
+    out[ 3] = in4 - in1;
+
+    in0    -= in2;
+    in5     = MULH3(in5 - in3, C6, 2);
+    out[ 0] =
+    out[ 5] = in0 - in5;
+    out[ 6] =
+    out[11] = in0 + in5;
+}
+
+/* return the number of decoded frames */
+static int mp_decode_layer1(MPADecodeContext *s)
+{
+    int bound, i, v, n, ch, j, mant;
+    uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
+    uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
+
+    if (s->mode == MPA_JSTEREO)
+        bound = (s->mode_ext + 1) * 4;
+    else
+        bound = SBLIMIT;
+
+    /* allocation bits */
+    for (i = 0; i < bound; i++) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            allocation[ch][i] = get_bits(&s->gb, 4);
+        }
+    }
+    for (i = bound; i < SBLIMIT; i++)
+        allocation[0][i] = get_bits(&s->gb, 4);
+
+    /* scale factors */
+    for (i = 0; i < bound; i++) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            if (allocation[ch][i])
+                scale_factors[ch][i] = get_bits(&s->gb, 6);
+        }
+    }
+    for (i = bound; i < SBLIMIT; i++) {
+        if (allocation[0][i]) {
+            scale_factors[0][i] = get_bits(&s->gb, 6);
+            scale_factors[1][i] = get_bits(&s->gb, 6);
+        }
+    }
+
+    /* compute samples */
+    for (j = 0; j < 12; j++) {
+        for (i = 0; i < bound; i++) {
+            for (ch = 0; ch < s->nb_channels; ch++) {
+                n = allocation[ch][i];
+                if (n) {
+                    mant = get_bits(&s->gb, n + 1);
+                    v = l1_unscale(n, mant, scale_factors[ch][i]);
+                } else {
+                    v = 0;
+                }
+                s->sb_samples[ch][j][i] = v;
+            }
+        }
+        for (i = bound; i < SBLIMIT; i++) {
+            n = allocation[0][i];
+            if (n) {
+                mant = get_bits(&s->gb, n + 1);
+                v = l1_unscale(n, mant, scale_factors[0][i]);
+                s->sb_samples[0][j][i] = v;
+                v = l1_unscale(n, mant, scale_factors[1][i]);
+                s->sb_samples[1][j][i] = v;
+            } else {
+                s->sb_samples[0][j][i] = 0;
+                s->sb_samples[1][j][i] = 0;
+            }
+        }
+    }
+    return 12;
+}
+
+static int mp_decode_layer2(MPADecodeContext *s)
+{
+    int sblimit; /* number of used subbands */
+    const unsigned char *alloc_table;
+    int table, bit_alloc_bits, i, j, ch, bound, v;
+    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
+    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
+    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
+    int scale, qindex, bits, steps, k, l, m, b;
+
+    /* select decoding table */
+    table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
+                                   s->sample_rate, s->lsf);
+    sblimit     = ff_mpa_sblimit_table[table];
+    alloc_table = ff_mpa_alloc_tables[table];
+
+    if (s->mode == MPA_JSTEREO)
+        bound = (s->mode_ext + 1) * 4;
+    else
+        bound = sblimit;
+
+    av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
+
+    /* sanity check */
+    if (bound > sblimit)
+        bound = sblimit;
+
+    /* parse bit allocation */
+    j = 0;
+    for (i = 0; i < bound; i++) {
+        bit_alloc_bits = alloc_table[j];
+        for (ch = 0; ch < s->nb_channels; ch++)
+            bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
+        j += 1 << bit_alloc_bits;
+    }
+    for (i = bound; i < sblimit; i++) {
+        bit_alloc_bits = alloc_table[j];
+        v = get_bits(&s->gb, bit_alloc_bits);
+        bit_alloc[0][i] = v;
+        bit_alloc[1][i] = v;
+        j += 1 << bit_alloc_bits;
+    }
+
+    /* scale codes */
+    for (i = 0; i < sblimit; i++) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            if (bit_alloc[ch][i])
+                scale_code[ch][i] = get_bits(&s->gb, 2);
+        }
+    }
+
+    /* scale factors */
+    for (i = 0; i < sblimit; i++) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            if (bit_alloc[ch][i]) {
+                sf = scale_factors[ch][i];
+                switch (scale_code[ch][i]) {
+                default:
+                case 0:
+                    sf[0] = get_bits(&s->gb, 6);
+                    sf[1] = get_bits(&s->gb, 6);
+                    sf[2] = get_bits(&s->gb, 6);
+                    break;
+                case 2:
+                    sf[0] = get_bits(&s->gb, 6);
+                    sf[1] = sf[0];
+                    sf[2] = sf[0];
+                    break;
+                case 1:
+                    sf[0] = get_bits(&s->gb, 6);
+                    sf[2] = get_bits(&s->gb, 6);
+                    sf[1] = sf[0];
+                    break;
+                case 3:
+                    sf[0] = get_bits(&s->gb, 6);
+                    sf[2] = get_bits(&s->gb, 6);
+                    sf[1] = sf[2];
+                    break;
+                }
+            }
+        }
+    }
+
+    /* samples */
+    for (k = 0; k < 3; k++) {
+        for (l = 0; l < 12; l += 3) {
+            j = 0;
+            for (i = 0; i < bound; i++) {
+                bit_alloc_bits = alloc_table[j];
+                for (ch = 0; ch < s->nb_channels; ch++) {
+                    b = bit_alloc[ch][i];
+                    if (b) {
+                        scale = scale_factors[ch][i][k];
+                        qindex = alloc_table[j+b];
+                        bits = ff_mpa_quant_bits[qindex];
+                        if (bits < 0) {
+                            int v2;
+                            /* 3 values at the same time */
+                            v = get_bits(&s->gb, -bits);
+                            v2 = division_tabs[qindex][v];
+                            steps  = ff_mpa_quant_steps[qindex];
+
+                            s->sb_samples[ch][k * 12 + l + 0][i] =
+                                l2_unscale_group(steps,  v2       & 15, scale);
+                            s->sb_samples[ch][k * 12 + l + 1][i] =
+                                l2_unscale_group(steps, (v2 >> 4) & 15, scale);
+                            s->sb_samples[ch][k * 12 + l + 2][i] =
+                                l2_unscale_group(steps,  v2 >> 8      , scale);
+                        } else {
+                            for (m = 0; m < 3; m++) {
+                                v = get_bits(&s->gb, bits);
+                                v = l1_unscale(bits - 1, v, scale);
+                                s->sb_samples[ch][k * 12 + l + m][i] = v;
+                            }
+                        }
+                    } else {
+                        s->sb_samples[ch][k * 12 + l + 0][i] = 0;
+                        s->sb_samples[ch][k * 12 + l + 1][i] = 0;
+                        s->sb_samples[ch][k * 12 + l + 2][i] = 0;
+                    }
+                }
+                /* next subband in alloc table */
+                j += 1 << bit_alloc_bits;
+            }
+            /* XXX: find a way to avoid this duplication of code */
+            for (i = bound; i < sblimit; i++) {
+                bit_alloc_bits = alloc_table[j];
+                b = bit_alloc[0][i];
+                if (b) {
+                    int mant, scale0, scale1;
+                    scale0 = scale_factors[0][i][k];
+                    scale1 = scale_factors[1][i][k];
+                    qindex = alloc_table[j+b];
+                    bits = ff_mpa_quant_bits[qindex];
+                    if (bits < 0) {
+                        /* 3 values at the same time */
+                        v = get_bits(&s->gb, -bits);
+                        steps = ff_mpa_quant_steps[qindex];
+                        mant = v % steps;
+                        v = v / steps;
+                        s->sb_samples[0][k * 12 + l + 0][i] =
+                            l2_unscale_group(steps, mant, scale0);
+                        s->sb_samples[1][k * 12 + l + 0][i] =
+                            l2_unscale_group(steps, mant, scale1);
+                        mant = v % steps;
+                        v = v / steps;
+                        s->sb_samples[0][k * 12 + l + 1][i] =
+                            l2_unscale_group(steps, mant, scale0);
+                        s->sb_samples[1][k * 12 + l + 1][i] =
+                            l2_unscale_group(steps, mant, scale1);
+                        s->sb_samples[0][k * 12 + l + 2][i] =
+                            l2_unscale_group(steps, v, scale0);
+                        s->sb_samples[1][k * 12 + l + 2][i] =
+                            l2_unscale_group(steps, v, scale1);
+                    } else {
+                        for (m = 0; m < 3; m++) {
+                            mant = get_bits(&s->gb, bits);
+                            s->sb_samples[0][k * 12 + l + m][i] =
+                                l1_unscale(bits - 1, mant, scale0);
+                            s->sb_samples[1][k * 12 + l + m][i] =
+                                l1_unscale(bits - 1, mant, scale1);
+                        }
+                    }
+                } else {
+                    s->sb_samples[0][k * 12 + l + 0][i] = 0;
+                    s->sb_samples[0][k * 12 + l + 1][i] = 0;
+                    s->sb_samples[0][k * 12 + l + 2][i] = 0;
+                    s->sb_samples[1][k * 12 + l + 0][i] = 0;
+                    s->sb_samples[1][k * 12 + l + 1][i] = 0;
+                    s->sb_samples[1][k * 12 + l + 2][i] = 0;
+                }
+                /* next subband in alloc table */
+                j += 1 << bit_alloc_bits;
+            }
+            /* fill remaining samples to zero */
+            for (i = sblimit; i < SBLIMIT; i++) {
+                for (ch = 0; ch < s->nb_channels; ch++) {
+                    s->sb_samples[ch][k * 12 + l + 0][i] = 0;
+                    s->sb_samples[ch][k * 12 + l + 1][i] = 0;
+                    s->sb_samples[ch][k * 12 + l + 2][i] = 0;
+                }
+            }
+        }
+    }
+    return 3 * 12;
+}
+
+#define SPLIT(dst,sf,n)             \
+    if (n == 3) {                   \
+        int m = (sf * 171) >> 9;    \
+        dst   = sf - 3 * m;         \
+        sf    = m;                  \
+    } else if (n == 4) {            \
+        dst  = sf & 3;              \
+        sf >>= 2;                   \
+    } else if (n == 5) {            \
+        int m = (sf * 205) >> 10;   \
+        dst   = sf - 5 * m;         \
+        sf    = m;                  \
+    } else if (n == 6) {            \
+        int m = (sf * 171) >> 10;   \
+        dst   = sf - 6 * m;         \
+        sf    = m;                  \
+    } else {                        \
+        dst = 0;                    \
+    }
+
+static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
+                                           int n3)
+{
+    SPLIT(slen[3], sf, n3)
+    SPLIT(slen[2], sf, n2)
+    SPLIT(slen[1], sf, n1)
+    slen[0] = sf;
+}
+
+static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
+                                         int16_t *exponents)
+{
+    const uint8_t *bstab, *pretab;
+    int len, i, j, k, l, v0, shift, gain, gains[3];
+    int16_t *exp_ptr;
+
+    exp_ptr = exponents;
+    gain    = g->global_gain - 210;
+    shift   = g->scalefac_scale + 1;
+
+    bstab  = band_size_long[s->sample_rate_index];
+    pretab = mpa_pretab[g->preflag];
+    for (i = 0; i < g->long_end; i++) {
+        v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
+        len = bstab[i];
+        for (j = len; j > 0; j--)
+            *exp_ptr++ = v0;
+    }
+
+    if (g->short_start < 13) {
+        bstab    = band_size_short[s->sample_rate_index];
+        gains[0] = gain - (g->subblock_gain[0] << 3);
+        gains[1] = gain - (g->subblock_gain[1] << 3);
+        gains[2] = gain - (g->subblock_gain[2] << 3);
+        k        = g->long_end;
+        for (i = g->short_start; i < 13; i++) {
+            len = bstab[i];
+            for (l = 0; l < 3; l++) {
+                v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
+                for (j = len; j > 0; j--)
+                    *exp_ptr++ = v0;
+            }
+        }
+    }
+}
+
+/* handle n = 0 too */
+static inline int get_bitsz(GetBitContext *s, int n)
+{
+    return n ? get_bits(s, n) : 0;
+}
+
+
+static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
+                          int *end_pos2)
+{
+    if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
+        s->gb           = s->in_gb;
+        s->in_gb.buffer = NULL;
+        assert((get_bits_count(&s->gb) & 7) == 0);
+        skip_bits_long(&s->gb, *pos - *end_pos);
+        *end_pos2 =
+        *end_pos  = *end_pos2 + get_bits_count(&s->gb) - *pos;
+        *pos      = get_bits_count(&s->gb);
+    }
+}
+
+/* Following is a optimized code for
+            INTFLOAT v = *src
+            if(get_bits1(&s->gb))
+                v = -v;
+            *dst = v;
+*/
+#if CONFIG_FLOAT
+#define READ_FLIP_SIGN(dst,src)                     \
+    v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31);  \
+    AV_WN32A(dst, v);
+#else
+#define READ_FLIP_SIGN(dst,src)     \
+    v      = -get_bits1(&s->gb);    \
+    *(dst) = (*(src) ^ v) - v;
+#endif
+
+static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
+                          int16_t *exponents, int end_pos2)
+{
+    int s_index;
+    int i;
+    int last_pos, bits_left;
+    VLC *vlc;
+    int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
+
+    /* low frequencies (called big values) */
+    s_index = 0;
+    for (i = 0; i < 3; i++) {
+        int j, k, l, linbits;
+        j = g->region_size[i];
+        if (j == 0)
+            continue;
+        /* select vlc table */
+        k       = g->table_select[i];
+        l       = mpa_huff_data[k][0];
+        linbits = mpa_huff_data[k][1];
+        vlc     = &huff_vlc[l];
+
+        if (!l) {
+            memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
+            s_index += 2 * j;
+            continue;
+        }
+
+        /* read huffcode and compute each couple */
+        for (; j > 0; j--) {
+            int exponent, x, y;
+            int v;
+            int pos = get_bits_count(&s->gb);
+
+            if (pos >= end_pos){
+                switch_buffer(s, &pos, &end_pos, &end_pos2);
+                if (pos >= end_pos)
+                    break;
+            }
+            y = get_vlc2(&s->gb, vlc->table, 7, 3);
+
+            if (!y) {
+                g->sb_hybrid[s_index  ] =
+                g->sb_hybrid[s_index+1] = 0;
+                s_index += 2;
+                continue;
+            }
+
+            exponent= exponents[s_index];
+
+            av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
+                    i, g->region_size[i] - j, x, y, exponent);
+            if (y & 16) {
+                x = y >> 5;
+                y = y & 0x0f;
+                if (x < 15) {
+                    READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
+                } else {
+                    x += get_bitsz(&s->gb, linbits);
+                    v  = l3_unscale(x, exponent);
+                    if (get_bits1(&s->gb))
+                        v = -v;
+                    g->sb_hybrid[s_index] = v;
+                }
+                if (y < 15) {
+                    READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
+                } else {
+                    y += get_bitsz(&s->gb, linbits);
+                    v  = l3_unscale(y, exponent);
+                    if (get_bits1(&s->gb))
+                        v = -v;
+                    g->sb_hybrid[s_index+1] = v;
+                }
+            } else {
+                x = y >> 5;
+                y = y & 0x0f;
+                x += y;
+                if (x < 15) {
+                    READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
+                } else {
+                    x += get_bitsz(&s->gb, linbits);
+                    v  = l3_unscale(x, exponent);
+                    if (get_bits1(&s->gb))
+                        v = -v;
+                    g->sb_hybrid[s_index+!!y] = v;
+                }
+                g->sb_hybrid[s_index + !y] = 0;
+            }
+            s_index += 2;
+        }
+    }
+
+    /* high frequencies */
+    vlc = &huff_quad_vlc[g->count1table_select];
+    last_pos = 0;
+    while (s_index <= 572) {
+        int pos, code;
+        pos = get_bits_count(&s->gb);
+        if (pos >= end_pos) {
+            if (pos > end_pos2 && last_pos) {
+                /* some encoders generate an incorrect size for this
+                   part. We must go back into the data */
+                s_index -= 4;
+                skip_bits_long(&s->gb, last_pos - pos);
+                av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
+                if(s->err_recognition & AV_EF_BITSTREAM)
+                    s_index=0;
+                break;
+            }
+            switch_buffer(s, &pos, &end_pos, &end_pos2);
+            if (pos >= end_pos)
+                break;
+        }
+        last_pos = pos;
+
+        code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
+        av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
+        g->sb_hybrid[s_index+0] =
+        g->sb_hybrid[s_index+1] =
+        g->sb_hybrid[s_index+2] =
+        g->sb_hybrid[s_index+3] = 0;
+        while (code) {
+            static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
+            int v;
+            int pos = s_index + idxtab[code];
+            code   ^= 8 >> idxtab[code];
+            READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
+        }
+        s_index += 4;
+    }
+    /* skip extension bits */
+    bits_left = end_pos2 - get_bits_count(&s->gb);
+    if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
+        av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+        s_index=0;
+    } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
+        av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+        s_index = 0;
+    }
+    memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
+    skip_bits_long(&s->gb, bits_left);
+
+    i = get_bits_count(&s->gb);
+    switch_buffer(s, &i, &end_pos, &end_pos2);
+
+    return 0;
+}
+
+/* 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->fdsp.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);
+                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);
+                init_long_region(s, g, region_address1, region_address2);
+            }
+            region_offset2size(g);
+            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) {
+        av_assert0(s->frame != NULL);
+        s->frame->nb_samples = s->avctx->frame_size;
+        if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+            return ret;
+        }
+        samples = (OUT_INT **)s->frame->extended_data;
+    }
+
+    /* apply the synthesis filter */
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        int sample_stride;
+        if (s->avctx->sample_fmt == OUT_FMT_P) {
+            samples_ptr   = samples[ch];
+            sample_stride = 1;
+        } else {
+            samples_ptr   = samples[0] + ch;
+            sample_stride = s->nb_channels;
+        }
+        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,
+                                        sample_stride, s->sb_samples[ch][i]);
+            samples_ptr += 32 * sample_stride;
+        }
+    }
+
+    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;
+    }
+
+    s->frame = data;
+
+    ret = mp_decode_frame(s, NULL, buf, buf_size);
+    if (ret >= 0) {
+        s->frame->nb_samples = avctx->frame_size;
+        *got_frame_ptr       = 1;
+        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;
+
+    s->frame = data;
+
+    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;
+
+    return buf_size;
+}
+#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
+
+#if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
+
+/**
+ * Context for MP3On4 decoder
+ */
+typedef struct MP3On4DecodeContext {
+    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
+} 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]);
+
+    return 0;
+}
+
+
+static av_cold 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);
+    // 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;
+    }
+
+    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)
+{
+    AVFrame *frame         = data;
+    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[2];
+    int fr, ch, ret;
+
+    /* get output buffer */
+    frame->nb_samples = MPA_FRAME_SIZE;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+    out_samples = (OUT_INT **)frame->extended_data;
+
+    // Discard too short frames
+    if (buf_size < HEADER_SIZE)
+        return AVERROR_INVALIDDATA;
+
+    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 ||
+            s->coff[fr] + 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;
+
+        outptr[0] = out_samples[s->coff[fr]];
+        if (m->nb_channels > 1)
+            outptr[1] = out_samples[s->coff[fr] + 1];
+
+        if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
+            return ret;
+
+        out_size += ret;
+        buf      += fsize;
+        len      -= fsize;
+
+        avctx->bit_rate += m->bit_rate;
+    }
+
+    /* update codec info */
+    avctx->sample_rate = s->mp3decctx[0]->sample_rate;
+
+    frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
+    *got_frame_ptr    = 1;
+
+    return buf_size;
+}
+#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */