libavcodec/qdm2.c:1967: error: Buffer Overrun L1
  Offset: [417, +oo] (⇐ [33, +oo] + 384) Size: 2 by call to `qdm2_decode`.

libavcodec/qdm2.c:1967:13: Call
1965. 
1966.     for (i = 0; i < 16; i++) {
1967.         if (qdm2_decode(s, buf, out) < 0)
                  ^
1968.             return -1;
1969.         out += s->channels * s->frame_size;

libavcodec/qdm2.c:1911:13: Call
  1909.     if (!q->has_errors) {
  1910.         if (q->sub_packet == 2)
  1911.             qdm2_decode_fft_packets(q);
                    ^
  1912. 
  1913.         qdm2_fft_tone_synthesizer(q, q->sub_packet);

libavcodec/qdm2.c:1967:13: Call
1965. 
1966.     for (i = 0; i < 16; i++) {
1967.         if (qdm2_decode(s, buf, out) < 0)
                  ^
1968.             return -1;
1969.         out += s->channels * s->frame_size;

libavcodec/qdm2.c:1886:1: Parameter `q->synth_buf_offset[*]`
  1884. 
  1885. 
  1886. static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
        ^
  1887. {
  1888.     int ch, i;

libavcodec/qdm2.c:1928:9: Call
  1926.     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
  1927.     if (!q->has_errors && q->do_synth_filter)
  1928.         qdm2_synthesis_filter(q, q->sub_packet);
                ^
  1929. 
  1930.     q->sub_packet = (q->sub_packet + 1) % 16;

libavcodec/qdm2.c:1602:1: Parameter `q->synth_buf[*]`
    1600.  * @param index    subpacket number
    1601.  */
    1602. static void qdm2_synthesis_filter (QDM2Context *q, int index)
          ^
    1603. {
    1604.     OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];

libavcodec/qdm2.c:1619:13: Call
    1617. 
    1618.         for (i = 0; i < 8; i++) {
    1619.             ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
                      ^
    1620.                 ff_mpa_synth_window, &dither_state,
    1621.                 samples_ptr, q->nb_channels,

libavcodec/mpegaudiodec.c:627:1: Parameter `*synth_buf_ptr`
      625. /* XXX: optimize by avoiding ring buffer usage */
      626. #if !CONFIG_FLOAT
      627. void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
           ^
      628.                          MPA_INT *window, int *dither_state,
      629.                          OUT_INT *samples, int incr,

libavcodec/mpegaudiodec.c:636:5: Assignment
      634. 
      635.     offset = *synth_buf_offset;
      636.     synth_buf = synth_buf_ptr + offset;
               ^
      637. 
      638.     dct32(synth_buf, sb_samples);

libavcodec/mpegaudiodec.c:639:5: Call
      637. 
      638.     dct32(synth_buf, sb_samples);
      639.     apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
               ^
      640. 
      641.     offset = (offset - 32) & 511;

libavcodec/mpegaudiodec.c:570:1: <Length trace>
568. }
569. 
570. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
     ^
571.                                int *dither_state, OUT_INT *samples, int incr)
572. {

libavcodec/mpegaudiodec.c:570:1: Parameter `*synth_buf`
        568. }
        569. 
        570. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
             ^
        571.                                int *dither_state, OUT_INT *samples, int incr)
        572. {

libavcodec/mpegaudiodec.c:604:9: Assignment
        602.         p = synth_buf + 16 + j;
        603.         SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
        604.         p = synth_buf + 48 - j;
                     ^
        605.         SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
        606. 

libavcodec/mpegaudiodec.c:605:9: Array access: Offset: [417, +oo] (⇐ [33, +oo] + 384) Size: 2 by call to `qdm2_decode` 
        603.         SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
        604.         p = synth_buf + 48 - j;
        605.         SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
                     ^
        606. 
        607.         *samples = round_sample(&sum);
