libavcodec/mpegaudiodec.c:2501: error: Buffer Overrun L2
  Offset: [0, 8] Size: 2 by call to `mp_decode_frame`.

libavcodec/mpegaudiodec.c:2464:1: Parameter `*avpkt->data`
2462. 
2463. 
2464. static int decode_frame_mp3on4(AVCodecContext * avctx,
      ^
2465.                         void *data, int *data_size,
2466.                         AVPacket *avpkt)

libavcodec/mpegaudiodec.c:2495:9: Assignment
2493.         assert (m != NULL);
2494. 
2495.         header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
              ^
2496. 
2497.         if (ff_mpa_check_header(header) < 0) // Bad header, discard block

libavcodec/mpegaudiodec.c:2500:9: Call
2498.             break;
2499. 
2500.         ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
              ^
2501.         out_size += mp_decode_frame(m, outptr, buf, fsize);
2502.         buf += fsize;

libavcodec/mpegaudiodecheader.c:39:9: Assignment
  37.     if (header & (1<<20)) {
  38.         s->lsf = (header & (1<<19)) ? 0 : 1;
  39.         mpeg25 = 0;
              ^
  40.     } else {
  41.         s->lsf = 1;

libavcodec/mpegaudiodecheader.c:49:5: Assignment
  47.     sample_rate_index = (header >> 10) & 3;
  48.     sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
  49.     sample_rate_index += 3 * (s->lsf + mpeg25);
          ^
  50.     s->sample_rate_index = sample_rate_index;
  51.     s->error_protection = ((header >> 16) & 1) ^ 1;

libavcodec/mpegaudiodecheader.c:50:5: Assignment
  48.     sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
  49.     sample_rate_index += 3 * (s->lsf + mpeg25);
  50.     s->sample_rate_index = sample_rate_index;
          ^
  51.     s->error_protection = ((header >> 16) & 1) ^ 1;
  52.     s->sample_rate = sample_rate;

libavcodec/mpegaudiodec.c:2501:21: Call
2499. 
2500.         ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
2501.         out_size += mp_decode_frame(m, outptr, buf, fsize);
                          ^
2502.         buf += fsize;
2503.         len -= fsize;

libavcodec/mpegaudiodec.c:2188:1: Parameter `s->nb_channels`
  2186. }
  2187. 
  2188. static int mp_decode_frame(MPADecodeContext *s,
        ^
  2189.                            OUT_INT *samples, const uint8_t *buf, int buf_size)
  2190. {

libavcodec/mpegaudiodec.c:2204:21: Call
  2202.     case 1:
  2203.         s->avctx->frame_size = 384;
  2204.         nb_frames = mp_decode_layer1(s);
                            ^
  2205.         break;
  2206.     case 2:

libavcodec/mpegaudiodec.c:1119:1: <Offset trace>
1117. 
1118. /* return the number of decoded frames */
1119. static int mp_decode_layer1(MPADecodeContext *s)
      ^
1120. {
1121.     int bound, i, v, n, ch, j, mant;

libavcodec/mpegaudiodec.c:1119:1: Parameter `s->nb_channels`
    1117. 
    1118. /* return the number of decoded frames */
    1119. static int mp_decode_layer1(MPADecodeContext *s)
          ^
    1120. {
    1121.     int bound, i, v, n, ch, j, mant;

libavcodec/mpegaudiodec.c:1119:1: <Length trace>
1117. 
1118. /* return the number of decoded frames */
1119. static int mp_decode_layer1(MPADecodeContext *s)
      ^
1120. {
1121.     int bound, i, v, n, ch, j, mant;

libavcodec/mpegaudiodec.c:1119:1: Array declaration
    1117. 
    1118. /* return the number of decoded frames */
    1119. static int mp_decode_layer1(MPADecodeContext *s)
          ^
    1120. {
    1121.     int bound, i, v, n, ch, j, mant;

libavcodec/mpegaudiodec.c:1133:13: Array access: Offset: [0, 8] Size: 2 by call to `mp_decode_frame` 
    1131.     for(i=0;i<bound;i++) {
    1132.         for(ch=0;ch<s->nb_channels;ch++) {
    1133.             allocation[ch][i] = get_bits(&s->gb, 4);
                      ^
    1134.         }
    1135.     }

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;
    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);
    }
    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);
        }
    }
    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 decode_frame_mp3on4(AVCodecContext * avctx,\n                        void *data, int *data_size,\n                        AVPacket *avpkt)\n{\n    const uint8_t *buf = avpkt->data;\n    int buf_size = avpkt->size;\n    MP3On4DecodeContext *s = avctx->priv_data;\n    MPADecodeContext *m;\n    int fsize, len = buf_size, out_size = 0;\n    uint32_t header;\n    OUT_INT *out_samples = data;\n    OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];\n    OUT_INT *outptr, *bp;\n    int fr, j, n;\n    *data_size = 0;\n    if (buf_size < HEADER_SIZE)\n        return -1;\n    outptr = s->frames == 1 ? out_samples : decoded_buf;\n    avctx->bit_rate = 0;\n    for (fr = 0; fr < s->frames; fr++) {\n        fsize = AV_RB16(buf) >> 4;\n        fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);\n        m = s->mp3decctx[fr];\n        assert (m != NULL);\n        header = (AV_RB32(buf) & 0x000fffff) | s->syncword;\n        if (ff_mpa_check_header(header) < 0)\n            break;\n        ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);\n        out_size += mp_decode_frame(m, outptr, buf, fsize);\n        buf += fsize;\n        len -= fsize;\n        if(s->frames > 1) {\n            n = m->avctx->frame_size*m->nb_channels;\n            bp = out_samples + s->coff[fr];\n            if(m->nb_channels == 1) {\n                for(j = 0; j < n; j++) {\n                    *bp = decoded_buf[j];\n                    bp += avctx->channels;\n                }\n            } else {\n                for(j = 0; j < n; j++) {\n                    bp[0] = decoded_buf[j++];\n                    bp[1] = decoded_buf[j];\n                    bp += avctx->channels;\n                }\n            }\n        }\n        avctx->bit_rate += m->bit_rate;\n    }\n    avctx->sample_rate = s->mp3decctx[0]->sample_rate;\n    *data_size = out_size;\n    return buf_size;\n}', 'int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)\n{\n    int sample_rate, frame_size, mpeg25, padding;\n    int sample_rate_index, bitrate_index;\n    if (header & (1<<20)) {\n        s->lsf = (header & (1<<19)) ? 0 : 1;\n        mpeg25 = 0;\n    } else {\n        s->lsf = 1;\n        mpeg25 = 1;\n    }\n    s->layer = 4 - ((header >> 17) & 3);\n    sample_rate_index = (header >> 10) & 3;\n    sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);\n    sample_rate_index += 3 * (s->lsf + mpeg25);\n    s->sample_rate_index = sample_rate_index;\n    s->error_protection = ((header >> 16) & 1) ^ 1;\n    s->sample_rate = sample_rate;\n    bitrate_index = (header >> 12) & 0xf;\n    padding = (header >> 9) & 1;\n    s->mode = (header >> 6) & 3;\n    s->mode_ext = (header >> 4) & 3;\n    if (s->mode == MPA_MONO)\n        s->nb_channels = 1;\n    else\n        s->nb_channels = 2;\n    if (bitrate_index != 0) {\n        frame_size = ff_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];\n        s->bit_rate = frame_size * 1000;\n        switch(s->layer) {\n        case 1:\n            frame_size = (frame_size * 12000) / sample_rate;\n            frame_size = (frame_size + padding) * 4;\n            break;\n        case 2:\n            frame_size = (frame_size * 144000) / sample_rate;\n            frame_size += padding;\n            break;\n        default:\n        case 3:\n            frame_size = (frame_size * 144000) / (sample_rate << s->lsf);\n            frame_size += padding;\n            break;\n        }\n        s->frame_size = frame_size;\n    } else {\n        return 1;\n    }\n#if defined(DEBUG)\n    dprintf(s->avctx, "layer%d, %d Hz, %d kbits/s, ",\n           s->layer, s->sample_rate, s->bit_rate);\n    if (s->nb_channels == 2) {\n        if (s->layer == 3) {\n            if (s->mode_ext & MODE_EXT_MS_STEREO)\n                dprintf(s->avctx, "ms-");\n            if (s->mode_ext & MODE_EXT_I_STEREO)\n                dprintf(s->avctx, "i-");\n        }\n        dprintf(s->avctx, "stereo");\n    } else {\n        dprintf(s->avctx, "mono");\n    }\n    dprintf(s->avctx, "\\n");\n#endif\n    return 0;\n}', 'static int mp_decode_frame(MPADecodeContext *s,\n                           OUT_INT *samples, const uint8_t *buf, int buf_size)\n{\n    int i, nb_frames, ch;\n    OUT_INT *samples_ptr;\n    init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);\n    if (s->error_protection)\n        skip_bits(&s->gb, 16);\n    dprintf(s->avctx, "frame %d:\\n", s->frame_count);\n    switch(s->layer) {\n    case 1:\n        s->avctx->frame_size = 384;\n        nb_frames = mp_decode_layer1(s);\n        break;\n    case 2:\n        s->avctx->frame_size = 1152;\n        nb_frames = mp_decode_layer2(s);\n        break;\n    case 3:\n        s->avctx->frame_size = s->lsf ? 576 : 1152;\n    default:\n        nb_frames = mp_decode_layer3(s);\n        s->last_buf_size=0;\n        if(s->in_gb.buffer){\n            align_get_bits(&s->gb);\n            i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3;\n            if(i >= 0 && i <= BACKSTEP_SIZE){\n                memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);\n                s->last_buf_size=i;\n            }else\n                av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\\n", i);\n            s->gb= s->in_gb;\n            s->in_gb.buffer= NULL;\n        }\n        align_get_bits(&s->gb);\n        assert((get_bits_count(&s->gb) & 7) == 0);\n        i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3;\n        if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){\n            if(i<0)\n                av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\\n", i);\n            i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);\n        }\n        assert(i <= buf_size - HEADER_SIZE && i>= 0);\n        memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);\n        s->last_buf_size += i;\n        break;\n    }\n    for(ch=0;ch<s->nb_channels;ch++) {\n        samples_ptr = samples + ch;\n        for(i=0;i<nb_frames;i++) {\n            ff_mpa_synth_filter(s->synth_buf[ch], &(s->synth_buf_offset[ch]),\n                         window, &s->dither_state,\n                         samples_ptr, s->nb_channels,\n                         s->sb_samples[ch][i]);\n            samples_ptr += 32 * s->nb_channels;\n        }\n    }\n    return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;\n}', 'static int mp_decode_layer1(MPADecodeContext *s)\n{\n    int bound, i, v, n, ch, j, mant;\n    uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];\n    uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];\n    if (s->mode == MPA_JSTEREO)\n        bound = (s->mode_ext + 1) * 4;\n    else\n        bound = SBLIMIT;\n    for(i=0;i<bound;i++) {\n        for(ch=0;ch<s->nb_channels;ch++) {\n            allocation[ch][i] = get_bits(&s->gb, 4);\n        }\n    }\n    for(i=bound;i<SBLIMIT;i++) {\n        allocation[0][i] = get_bits(&s->gb, 4);\n    }\n    for(i=0;i<bound;i++) {\n        for(ch=0;ch<s->nb_channels;ch++) {\n            if (allocation[ch][i])\n                scale_factors[ch][i] = get_bits(&s->gb, 6);\n        }\n    }\n    for(i=bound;i<SBLIMIT;i++) {\n        if (allocation[0][i]) {\n            scale_factors[0][i] = get_bits(&s->gb, 6);\n            scale_factors[1][i] = get_bits(&s->gb, 6);\n        }\n    }\n    for(j=0;j<12;j++) {\n        for(i=0;i<bound;i++) {\n            for(ch=0;ch<s->nb_channels;ch++) {\n                n = allocation[ch][i];\n                if (n) {\n                    mant = get_bits(&s->gb, n + 1);\n                    v = l1_unscale(n, mant, scale_factors[ch][i]);\n                } else {\n                    v = 0;\n                }\n                s->sb_samples[ch][j][i] = v;\n            }\n        }\n        for(i=bound;i<SBLIMIT;i++) {\n            n = allocation[0][i];\n            if (n) {\n                mant = get_bits(&s->gb, n + 1);\n                v = l1_unscale(n, mant, scale_factors[0][i]);\n                s->sb_samples[0][j][i] = v;\n                v = l1_unscale(n, mant, scale_factors[1][i]);\n                s->sb_samples[1][j][i] = v;\n            } else {\n                s->sb_samples[0][j][i] = 0;\n                s->sb_samples[1][j][i] = 0;\n            }\n        }\n    }\n    return 12;\n}']