1 files changed, 66 insertions, 26 deletions

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 269a051f69..c8727c6b1c 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -5,20 +5,20 @@
  * Copyright (c) 2005 Alex Beregszaszi
  * Copyright (c) 2005 Roberto Togni
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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,
+ * FFmpeg 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
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -171,7 +171,7 @@ typedef struct {
     /// I/O data
     const uint8_t *compressed_data;
     int compressed_size;
-    float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
+    float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
 
     /// Synthesis filter
     MPADSPContext mpadsp;
@@ -345,7 +345,14 @@ static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
 
     /* stage-3, optional */
     if (flag) {
-        int tmp = vlc_stage3_values[value];
+        int tmp;
+
+        if (value >= 60) {
+            av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
+            return 0;
+        }
+
+        tmp= vlc_stage3_values[value];
 
         if ((value & ~3) > 0)
             tmp += get_bits (gb, (value >> 2));
@@ -755,7 +762,7 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra
  * @param sb_min    lower subband processed (sb_min included)
  * @param sb_max    higher subband processed (sb_max excluded)
  */
-static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
+static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
 {
     int sb, j, k, n, ch, run, channels;
     int joined_stereo, zero_encoding, chs;
@@ -769,7 +776,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
         for (sb=sb_min; sb < sb_max; sb++)
             build_sb_samples_from_noise (q, sb);
 
-        return;
+        return 0;
     }
 
     for (sb = sb_min; sb < sb_max; sb++) {
@@ -789,6 +796,11 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
                 for (j = 0; j < 16; j++)
                     sign_bits[j] = get_bits1 (gb);
 
+            if (q->coding_method[0][sb][0] <= 0) {
+                av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
+                return AVERROR_INVALIDDATA;
+            }
+
             for (j = 0; j < 64; j++)
                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
@@ -875,10 +887,11 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
                     case 30:
                         if (get_bits_left(gb) >= 4) {
                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
-                            if (index < FF_ARRAY_ELEMS(type30_dequant)) {
-                                samples[0] = type30_dequant[index];
-                            } else
-                                samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                            if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
+                                av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
+                                return AVERROR_INVALIDDATA;
+                            }
+                            samples[0] = type30_dequant[index];
                         } else
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
 
@@ -894,11 +907,12 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
                                 type34_first = 0;
                             } else {
                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
-                                if (index < FF_ARRAY_ELEMS(type34_delta)) {
-                                    samples[0] = type34_delta[index] / type34_div + type34_predictor;
-                                    type34_predictor = samples[0];
-                                } else
-                                    samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                                if (index >= FF_ARRAY_ELEMS(type34_delta)) {
+                                    av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
+                                    return AVERROR_INVALIDDATA;
+                                }
+                                samples[0] = type34_delta[index] / type34_div + type34_predictor;
+                                type34_predictor = samples[0];
                             }
                         } else {
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
@@ -933,6 +947,7 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
             } // j loop
         } // channel loop
     } // subband loop
+    return 0;
 }
 
 
@@ -944,23 +959,26 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
  * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
  * @param gb        bitreader context
  */
-static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
+static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
 {
     int i, k, run, level, diff;
 
     if (get_bits_left(gb) < 16)
-        return;
+        return -1;
     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
 
     quantized_coeffs[0] = level;
 
     for (i = 0; i < 7; ) {
         if (get_bits_left(gb) < 16)
-            break;
+            return -1;
         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
 
+        if (i + run >= 8)
+            return -1;
+
         if (get_bits_left(gb) < 16)
-            break;
+            return -1;
         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
 
         for (k = 1; k <= run; k++)
@@ -969,6 +987,7 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext
         level += diff;
         i += run;
     }
+    return 0;
 }
 
 
@@ -1043,7 +1062,7 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
  * @param q       context
  * @param node    pointer to node with packet
  */
-static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
+static int process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
 {
     GetBitContext gb;
     int i, j, k, n, ch, run, level, diff;
@@ -1061,6 +1080,9 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
 
+                if (j + run >= 8)
+                    return -1;
+
                 for (k = 1; k <= run; k++)
                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
 
@@ -1072,6 +1094,8 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
     for (ch = 0; ch < q->nb_channels; ch++)
         for (i = 0; i < 8; i++)
             q->quantized_coeffs[ch][0][i] = 0;
+
+    return 0;
 }
 
 
@@ -1142,7 +1166,7 @@ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
 }
 
-/*
+/**
  * Process new subpackets for synthesis filter
  *
  * @param q       context
@@ -1176,7 +1200,7 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
 }
 
 
-/*
+/**
  * Decode superblock, fill packet lists.
  *
  * @param q    context
@@ -1234,6 +1258,11 @@ static void qdm2_decode_super_block (QDM2Context *q)
     for (i = 0; packet_bytes > 0; i++) {
         int j;
 
+        if (i>=FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
+            SAMPLES_NEEDED_2("too many packet bytes");
+            return;
+        }
+
         q->sub_packet_list_A[i].next = NULL;
 
         if (i > 0) {
@@ -1332,9 +1361,14 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *
     local_int_10 = 1 << (q->group_order - duration - 1);
     offset = 1;
 
-    while (1) {
+    while (get_bits_left(gb)>0) {
         if (q->superblocktype_2_3) {
             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
+                if (get_bits_left(gb)<0) {
+                    if(local_int_4 < q->group_size)
+                        av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
+                    return;
+                }
                 offset = 1;
                 if (n == 0) {
                     local_int_4 += local_int_10;
@@ -1762,8 +1796,10 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
 
     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
     extradata += 4;
-    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
+    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
         return AVERROR_INVALIDDATA;
+    }
     avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
                                                    AV_CH_LAYOUT_MONO;
 
@@ -1790,6 +1826,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     // something like max decodable tones
     s->group_order = av_log2(s->group_size) + 1;
     s->frame_size = s->group_size / 16; // 16 iterations per super block
+
     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
         return AVERROR_INVALIDDATA;
 
@@ -1863,6 +1900,9 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
     int ch, i;
     const int frame_size = (q->frame_size * q->channels);
 
+    if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
+        return -1;
+
     /* select input buffer */
     q->compressed_data = in;
     q->compressed_size = q->checksum_size;