libavcodec/mpegaudiodec.c:2495: error: Buffer Overrun L1
  Offset: [368, +oo] (⇐ [48, +oo] + 320) Size: 2 by call to `mp_decode_frame`.

libavcodec/mpegaudiodec.c:2460:1: Parameter `*buf`
2458. 
2459. 
2460. static int decode_frame_mp3on4(AVCodecContext * avctx,
      ^
2461.                         void *data, int *data_size,
2462.                         const uint8_t * buf, int buf_size)

libavcodec/mpegaudiodec.c:2489:9: Assignment
2487.         assert (m != NULL);
2488. 
2489.         header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
              ^
2490. 
2491.         if (ff_mpa_check_header(header) < 0) // Bad header, discard block

libavcodec/mpegaudiodec.c:2494:9: Call
2492.             break;
2493. 
2494.         ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
              ^
2495.         out_size += mp_decode_frame(m, outptr, buf, fsize);
2496.         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:2495:21: Call
2493. 
2494.         ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
2495.         out_size += mp_decode_frame(m, outptr, buf, fsize);
                          ^
2496.         buf += fsize;
2497.         len -= fsize;

libavcodec/mpegaudiodec.c:2188:1: Parameter `s->synth_buf[*]`
  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:2248:13: Call
  2246.         samples_ptr = samples + ch;
  2247.         for(i=0;i<nb_frames;i++) {
  2248.             ff_mpa_synth_filter(s->synth_buf[ch], &(s->synth_buf_offset[ch]),
                    ^
  2249.                          window, &s->dither_state,
  2250.                          samples_ptr, s->nb_channels,

libavcodec/mpegaudiodec.c:854:1: <Length trace>
852.    32 samples. */
853. /* XXX: optimize by avoiding ring buffer usage */
854. void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
     ^
855.                          MPA_INT *window, int *dither_state,
856.                          OUT_INT *samples, int incr,

libavcodec/mpegaudiodec.c:854:1: Parameter `*synth_buf_ptr`
    852.    32 samples. */
    853. /* XXX: optimize by avoiding ring buffer usage */
    854. void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
         ^
    855.                          MPA_INT *window, int *dither_state,
    856.                          OUT_INT *samples, int incr,

libavcodec/mpegaudiodec.c:873:5: Assignment
    871. 
    872.     offset = *synth_buf_offset;
    873.     synth_buf = synth_buf_ptr + offset;
             ^
    874. 
    875.     for(j=0;j<32;j++) {

libavcodec/mpegaudiodec.c:894:5: Assignment
    892.     p = synth_buf + 16;
    893.     SUM8(MACS, sum, w, p);
    894.     p = synth_buf + 48;
             ^
    895.     SUM8(MLSS, sum, w + 32, p);
    896.     *samples = round_sample(&sum);

libavcodec/mpegaudiodec.c:895:5: Array access: Offset: [368, +oo] (⇐ [48, +oo] + 320) Size: 2 by call to `mp_decode_frame` 
    893.     SUM8(MACS, sum, w, p);
    894.     p = synth_buf + 48;
    895.     SUM8(MLSS, sum, w + 32, p);
             ^
    896.     *samples = round_sample(&sum);
    897.     samples += incr;
