avcodec/mpegvideo: Use intra_scantable in dct_unquantize_h263_intra_c()
[ffmpeg.git] / libavcodec / qdm2.c
index 94bda91..88b6b19 100644 (file)
@@ -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
  */
 
 #include <stddef.h>
 #include <stdio.h>
 
-#define BITSTREAM_READER_LE
 #include "libavutil/channel_layout.h"
+
+#define BITSTREAM_READER_LE
 #include "avcodec.h"
 #include "get_bits.h"
+#include "bytestream.h"
 #include "internal.h"
-#include "rdft.h"
-#include "mpegaudiodsp.h"
 #include "mpegaudio.h"
+#include "mpegaudiodsp.h"
+#include "rdft.h"
 
-#include "qdm2data.h"
 #include "qdm2_tablegen.h"
 
-#undef NDEBUG
-#include <assert.h>
-
-
 #define QDM2_LIST_ADD(list, size, packet) \
 do { \
       if (size > 0) { \
@@ -83,7 +80,7 @@ typedef int8_t sb_int8_array[2][30][64];
 /**
  * Subpacket
  */
-typedef struct {
+typedef struct QDM2SubPacket {
     int type;            ///< subpacket type
     unsigned int size;   ///< subpacket size
     const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
@@ -97,12 +94,12 @@ typedef struct QDM2SubPNode {
     struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
 } QDM2SubPNode;
 
-typedef struct {
+typedef struct QDM2Complex {
     float re;
     float im;
 } QDM2Complex;
 
-typedef struct {
+typedef struct FFTTone {
     float level;
     QDM2Complex *complex;
     const float *table;
@@ -113,7 +110,7 @@ typedef struct {
     short cutoff;
 } FFTTone;
 
-typedef struct {
+typedef struct FFTCoefficient {
     int16_t sub_packet;
     uint8_t channel;
     int16_t offset;
@@ -121,14 +118,14 @@ typedef struct {
     uint8_t phase;
 } FFTCoefficient;
 
-typedef struct {
+typedef struct QDM2FFT {
     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
 } QDM2FFT;
 
 /**
  * QDM2 decoder context
  */
-typedef struct {
+typedef struct QDM2Context {
     /// Parameters from codec header, do not change during playback
     int nb_channels;         ///< number of channels
     int channels;            ///< number of channels
@@ -168,7 +165,7 @@ 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;
@@ -197,173 +194,11 @@ typedef struct {
     int noise_idx; ///< index for dithering noise table
 } QDM2Context;
 
-
-static VLC vlc_tab_level;
-static VLC vlc_tab_diff;
-static VLC vlc_tab_run;
-static VLC fft_level_exp_alt_vlc;
-static VLC fft_level_exp_vlc;
-static VLC fft_stereo_exp_vlc;
-static VLC fft_stereo_phase_vlc;
-static VLC vlc_tab_tone_level_idx_hi1;
-static VLC vlc_tab_tone_level_idx_mid;
-static VLC vlc_tab_tone_level_idx_hi2;
-static VLC vlc_tab_type30;
-static VLC vlc_tab_type34;
-static VLC vlc_tab_fft_tone_offset[5];
-
-static const uint16_t qdm2_vlc_offs[] = {
-    0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
-};
-
 static const int switchtable[23] = {
     0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
 };
 
-static av_cold void qdm2_init_vlc(void)
-{
-    static VLC_TYPE qdm2_table[3838][2];
-
-    vlc_tab_level.table           = &qdm2_table[qdm2_vlc_offs[0]];
-    vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
-    init_vlc(&vlc_tab_level, 8, 24,
-             vlc_tab_level_huffbits, 1, 1,
-             vlc_tab_level_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_diff.table           = &qdm2_table[qdm2_vlc_offs[1]];
-    vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
-    init_vlc(&vlc_tab_diff, 8, 37,
-             vlc_tab_diff_huffbits, 1, 1,
-             vlc_tab_diff_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_run.table           = &qdm2_table[qdm2_vlc_offs[2]];
-    vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
-    init_vlc(&vlc_tab_run, 5, 6,
-             vlc_tab_run_huffbits, 1, 1,
-             vlc_tab_run_huffcodes, 1, 1,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    fft_level_exp_alt_vlc.table           = &qdm2_table[qdm2_vlc_offs[3]];
-    fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] -
-                                            qdm2_vlc_offs[3];
-    init_vlc(&fft_level_exp_alt_vlc, 8, 28,
-             fft_level_exp_alt_huffbits, 1, 1,
-             fft_level_exp_alt_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    fft_level_exp_vlc.table           = &qdm2_table[qdm2_vlc_offs[4]];
-    fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
-    init_vlc(&fft_level_exp_vlc, 8, 20,
-             fft_level_exp_huffbits, 1, 1,
-             fft_level_exp_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    fft_stereo_exp_vlc.table           = &qdm2_table[qdm2_vlc_offs[5]];
-    fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] -
-                                         qdm2_vlc_offs[5];
-    init_vlc(&fft_stereo_exp_vlc, 6, 7,
-             fft_stereo_exp_huffbits, 1, 1,
-             fft_stereo_exp_huffcodes, 1, 1,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    fft_stereo_phase_vlc.table           = &qdm2_table[qdm2_vlc_offs[6]];
-    fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] -
-                                           qdm2_vlc_offs[6];
-    init_vlc(&fft_stereo_phase_vlc, 6, 9,
-             fft_stereo_phase_huffbits, 1, 1,
-             fft_stereo_phase_huffcodes, 1, 1,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_tone_level_idx_hi1.table =
-        &qdm2_table[qdm2_vlc_offs[7]];
-    vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] -
-                                                 qdm2_vlc_offs[7];
-    init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20,
-             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
-             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_tone_level_idx_mid.table =
-        &qdm2_table[qdm2_vlc_offs[8]];
-    vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] -
-                                                 qdm2_vlc_offs[8];
-    init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24,
-             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
-             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_tone_level_idx_hi2.table =
-        &qdm2_table[qdm2_vlc_offs[9]];
-    vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] -
-                                                 qdm2_vlc_offs[9];
-    init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24,
-             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
-             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_type30.table           = &qdm2_table[qdm2_vlc_offs[10]];
-    vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
-    init_vlc(&vlc_tab_type30, 6, 9,
-             vlc_tab_type30_huffbits, 1, 1,
-             vlc_tab_type30_huffcodes, 1, 1,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_type34.table           = &qdm2_table[qdm2_vlc_offs[11]];
-    vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
-    init_vlc(&vlc_tab_type34, 5, 10,
-             vlc_tab_type34_huffbits, 1, 1,
-             vlc_tab_type34_huffcodes, 1, 1,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_fft_tone_offset[0].table =
-        &qdm2_table[qdm2_vlc_offs[12]];
-    vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] -
-                                                 qdm2_vlc_offs[12];
-    init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23,
-             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
-             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_fft_tone_offset[1].table =
-        &qdm2_table[qdm2_vlc_offs[13]];
-    vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] -
-                                                 qdm2_vlc_offs[13];
-    init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28,
-             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
-             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_fft_tone_offset[2].table =
-        &qdm2_table[qdm2_vlc_offs[14]];
-    vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] -
-                                                 qdm2_vlc_offs[14];
-    init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32,
-             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
-             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_fft_tone_offset[3].table =
-        &qdm2_table[qdm2_vlc_offs[15]];
-    vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] -
-                                                 qdm2_vlc_offs[15];
-    init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35,
-             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
-             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-    vlc_tab_fft_tone_offset[4].table =
-        &qdm2_table[qdm2_vlc_offs[16]];
-    vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] -
-                                                 qdm2_vlc_offs[16];
-    init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38,
-             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
-             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2,
-             INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-}
-
-static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
+static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
 {
     int value;
 
@@ -375,7 +210,14 @@ 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));
@@ -385,7 +227,7 @@ static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
     return value;
 }
 
-static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
+static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
 {
     int value = qdm2_get_vlc(gb, vlc, 0, depth);
 
@@ -395,7 +237,7 @@ static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
 /**
  * QDM2 checksum
  *
- * @param data      pointer to data to be checksum'ed
+ * @param data      pointer to data to be checksummed
  * @param length    data length
  * @param value     checksum value
  *
@@ -694,8 +536,9 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
 
     if (!superblocktype_2_3) {
         /* This case is untested, no samples available */
-        SAMPLES_NEEDED
-        for (ch = 0; ch < nb_channels; ch++)
+        avpriv_request_sample(NULL, "!superblocktype_2_3");
+        return;
+        for (ch = 0; ch < nb_channels; ch++) {
             for (sb = 0; sb < 30; sb++) {
                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
                     add1 = tone_level_idx[ch][sb][j] - 10;
@@ -724,67 +567,68 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
                 }
                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
             }
-            acc = 0;
-            for (ch = 0; ch < nb_channels; ch++)
-                for (sb = 0; sb < 30; sb++)
-                    for (j = 0; j < 64; j++)
-                        acc += tone_level_idx_temp[ch][sb][j];
-
-            multres = 0x66666667 * (acc * 10);
-            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
-            for (ch = 0;  ch < nb_channels; ch++)
-                for (sb = 0; sb < 30; sb++)
-                    for (j = 0; j < 64; j++) {
-                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
-                        if (comp < 0)
-                            comp += 0xff;
-                        comp /= 256; // signed shift
-                        switch(sb) {
-                            case 0:
-                                if (comp < 30)
-                                    comp = 30;
-                                comp += 15;
-                                break;
-                            case 1:
-                                if (comp < 24)
-                                    comp = 24;
-                                comp += 10;
-                                break;
-                            case 2:
-                            case 3:
-                            case 4:
-                                if (comp < 16)
-                                    comp = 16;
-                        }
-                        if (comp <= 5)
-                            tmp = 0;
-                        else if (comp <= 10)
-                            tmp = 10;
-                        else if (comp <= 16)
-                            tmp = 16;
-                        else if (comp <= 24)
-                            tmp = -1;
-                        else
-                            tmp = 0;
-                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+        }
+        acc = 0;
+        for (ch = 0; ch < nb_channels; ch++)
+            for (sb = 0; sb < 30; sb++)
+                for (j = 0; j < 64; j++)
+                    acc += tone_level_idx_temp[ch][sb][j];
+
+        multres = 0x66666667LL * (acc * 10);
+        esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
+        for (ch = 0;  ch < nb_channels; ch++)
+            for (sb = 0; sb < 30; sb++)
+                for (j = 0; j < 64; j++) {
+                    comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
+                    if (comp < 0)
+                        comp += 0xff;
+                    comp /= 256; // signed shift
+                    switch(sb) {
+                        case 0:
+                            if (comp < 30)
+                                comp = 30;
+                            comp += 15;
+                            break;
+                        case 1:
+                            if (comp < 24)
+                                comp = 24;
+                            comp += 10;
+                            break;
+                        case 2:
+                        case 3:
+                        case 4:
+                            if (comp < 16)
+                                comp = 16;
                     }
+                    if (comp <= 5)
+                        tmp = 0;
+                    else if (comp <= 10)
+                        tmp = 10;
+                    else if (comp <= 16)
+                        tmp = 16;
+                    else if (comp <= 24)
+                        tmp = -1;
+                    else
+                        tmp = 0;
+                    coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+                }
+        for (sb = 0; sb < 30; sb++)
+            fix_coding_method_array(sb, nb_channels, coding_method);
+        for (ch = 0; ch < nb_channels; ch++)
             for (sb = 0; sb < 30; sb++)
-                fix_coding_method_array(sb, nb_channels, coding_method);
-            for (ch = 0; ch < nb_channels; ch++)
-                for (sb = 0; sb < 30; sb++)
-                    for (j = 0; j < 64; j++)
-                        if (sb >= 10) {
-                            if (coding_method[ch][sb][j] < 10)
-                                coding_method[ch][sb][j] = 10;
+                for (j = 0; j < 64; j++)
+                    if (sb >= 10) {
+                        if (coding_method[ch][sb][j] < 10)
+                            coding_method[ch][sb][j] = 10;
+                    } else {
+                        if (sb >= 2) {
+                            if (coding_method[ch][sb][j] < 16)
+                                coding_method[ch][sb][j] = 16;
                         } else {
-                            if (sb >= 2) {
-                                if (coding_method[ch][sb][j] < 16)
-                                    coding_method[ch][sb][j] = 16;
-                            } else {
-                                if (coding_method[ch][sb][j] < 30)
-                                    coding_method[ch][sb][j] = 30;
-                            }
+                            if (coding_method[ch][sb][j] < 30)
+                                coding_method[ch][sb][j] = 30;
                         }
+                    }
     } else { // superblocktype_2_3 != 0
         for (ch = 0; ch < nb_channels; ch++)
             for (sb = 0; sb < 30; sb++)
@@ -794,7 +638,6 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
 }
 
 /**
- *
  * Called by process_subpacket_11 to process more data from subpacket 11
  * with sb 0-8.
  * Called by process_subpacket_12 to process data from subpacket 12 with
@@ -806,7 +649,7 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
  * @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,
+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;
@@ -814,14 +657,15 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
     int type34_first;
     float type34_div = 0;
     float type34_predictor;
-    float samples[10], sign_bits[16];
+    float samples[10];
+    int sign_bits[16] = {0};
 
     if (length == 0) {
         // If no data use noise
         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++) {
@@ -845,6 +689,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
 
             if (fix_coding_method_array(sb, q->nb_channels,
                                             q->coding_method)) {
+                av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
                 build_sb_samples_from_noise(q, sb);
                 continue;
             }
@@ -869,6 +714,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                                 }
                             } else {
                                 n = get_bits(gb, 8);
+                                if (n >= 243) {
+                                    av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
+                                    return AVERROR_INVALIDDATA;
+                                }
+
                                 for (k = 0; k < 5; k++)
                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                             }
@@ -905,6 +755,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                                 }
                             } else {
                                 n = get_bits (gb, 8);
+                                if (n >= 243) {
+                                    av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
+                                    return AVERROR_INVALIDDATA;
+                                }
+
                                 for (k = 0; k < 5; k++)
                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                             }
@@ -918,6 +773,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                     case 24:
                         if (get_bits_left(gb) >= 7) {
                             n = get_bits(gb, 7);
+                            if (n >= 125) {
+                                av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
+                                return AVERROR_INVALIDDATA;
+                            }
+
                             for (k = 0; k < 3; k++)
                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
                         } else {
@@ -930,10 +790,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                     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);
 
@@ -949,11 +810,12 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                                 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);
@@ -990,6 +852,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
             } // j loop
         } // channel loop
     } // subband loop
+    return 0;
 }
 
 /**
@@ -1002,24 +865,27 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
  * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
  * @param gb        bitreader context
  */
-static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
+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++)
@@ -1028,6 +894,7 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
         level += diff;
         i += run;
     }
+    return 0;
 }
 
 /**
@@ -1102,7 +969,7 @@ 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;
@@ -1120,6 +987,9 @@ 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));
 
@@ -1131,6 +1001,8 @@ 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;
 }
 
 /**
@@ -1200,7 +1072,7 @@ 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
@@ -1233,7 +1105,7 @@ static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
         process_subpacket_12(q, NULL);
 }
 
-/*
+/**
  * Decode superblock, fill packet lists.
  *
  * @param q    context
@@ -1393,9 +1265,14 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
     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;
@@ -1708,12 +1585,19 @@ static void qdm2_synthesis_filter(QDM2Context *q, int index)
  *
  * @param q    context
  */
-static av_cold void qdm2_init_static_data(AVCodec *codec) {
+static av_cold void qdm2_init_static_data(void) {
+    static int done;
+
+    if(done)
+        return;
+
     qdm2_init_vlc();
     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
     softclip_table_init();
     rnd_table_init();
     init_noise_samples();
+
+    done = 1;
 }
 
 /**
@@ -1722,9 +1606,10 @@ static av_cold void qdm2_init_static_data(AVCodec *codec) {
 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
 {
     QDM2Context *s = avctx->priv_data;
-    uint8_t *extradata;
-    int extradata_size;
     int tmp_val, tmp, size;
+    GetByteContext gb;
+
+    qdm2_init_static_data();
 
     /* extradata parsing
 
@@ -1762,76 +1647,54 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
 
     if (!avctx->extradata || (avctx->extradata_size < 48)) {
         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
-    extradata      = avctx->extradata;
-    extradata_size = avctx->extradata_size;
+    bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
 
-    while (extradata_size > 7) {
-        if (!memcmp(extradata, "frmaQDM", 7))
+    while (bytestream2_get_bytes_left(&gb) > 8) {
+        if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
+                                            (uint64_t)MKBETAG('Q','D','M','2')))
             break;
-        extradata++;
-        extradata_size--;
+        bytestream2_skip(&gb, 1);
     }
 
-    if (extradata_size < 12) {
+    if (bytestream2_get_bytes_left(&gb) < 12) {
         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
-               extradata_size);
-        return -1;
-    }
-
-    if (memcmp(extradata, "frmaQDM", 7)) {
-        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
-        return -1;
-    }
-
-    if (extradata[7] == 'C') {
-//        s->is_qdmc = 1;
-        av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
-        return -1;
+               bytestream2_get_bytes_left(&gb));
+        return AVERROR_INVALIDDATA;
     }
 
-    extradata += 8;
-    extradata_size -= 8;
+    bytestream2_skip(&gb, 8);
+    size = bytestream2_get_be32(&gb);
 
-    size = AV_RB32(extradata);
-
-    if(size > extradata_size){
+    if (size > bytestream2_get_bytes_left(&gb)) {
         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
-               extradata_size, size);
-        return -1;
+               bytestream2_get_bytes_left(&gb), size);
+        return AVERROR_INVALIDDATA;
     }
 
-    extradata += 4;
     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
-    if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
+    if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
-    extradata += 8;
+    bytestream2_skip(&gb, 4);
 
-    avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
-    extradata += 4;
-    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
+    avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
+    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;
 
-    avctx->sample_rate = AV_RB32(extradata);
-    extradata += 4;
-
-    avctx->bit_rate = AV_RB32(extradata);
-    extradata += 4;
-
-    s->group_size = AV_RB32(extradata);
-    extradata += 4;
-
-    s->fft_size = AV_RB32(extradata);
-    extradata += 4;
-
-    s->checksum_size = AV_RB32(extradata);
+    avctx->sample_rate = bytestream2_get_be32(&gb);
+    avctx->bit_rate = bytestream2_get_be32(&gb);
+    s->group_size = bytestream2_get_be32(&gb);
+    s->fft_size = bytestream2_get_be32(&gb);
+    s->checksum_size = bytestream2_get_be32(&gb);
     if (s->checksum_size >= 1U << 28) {
         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
         return AVERROR_INVALIDDATA;
@@ -1842,6 +1705,7 @@ 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;
 
@@ -1864,26 +1728,17 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
     s->cm_table_select = tmp_val;
 
-    if (s->sub_sampling == 0)
-        tmp = 7999;
-    else
-        tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
-    /*
-    0: 7999 -> 0
-    1: 20000 -> 2
-    2: 28000 -> 2
-    */
-    if (tmp < 8000)
+    if (avctx->bit_rate <= 8000)
         s->coeff_per_sb_select = 0;
-    else if (tmp <= 16000)
+    else if (avctx->bit_rate < 16000)
         s->coeff_per_sb_select = 1;
     else
         s->coeff_per_sb_select = 2;
 
     // Fail on unknown fft order
     if ((s->fft_order < 7) || (s->fft_order > 9)) {
-        av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
-        return -1;
+        avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
+        return AVERROR_PATCHWELCOME;
     }
     if (s->fft_size != (1 << (s->fft_order - 1))) {
         av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
@@ -1912,6 +1767,9 @@ 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;
@@ -1951,7 +1809,7 @@ static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
 
     q->sub_packet = (q->sub_packet + 1) % 16;
 
-    /* clip and convert output float[] to 16bit signed samples */
+    /* clip and convert output float[] to 16-bit signed samples */
     for (i = 0; i < frame_size; i++) {
         int value = (int)q->output_buffer[i];
 
@@ -1983,15 +1841,13 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
 
     /* get output buffer */
     frame->nb_samples = 16 * s->frame_size;
-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
-    }
     out = (int16_t *)frame->data[0];
 
     for (i = 0; i < 16; i++) {
-        if (qdm2_decode(s, buf, out) < 0)
-            return -1;
+        if ((ret = qdm2_decode(s, buf, out)) < 0)
+            return ret;
         out += s->channels * s->frame_size;
     }
 
@@ -2007,8 +1863,7 @@ AVCodec ff_qdm2_decoder = {
     .id               = AV_CODEC_ID_QDM2,
     .priv_data_size   = sizeof(QDM2Context),
     .init             = qdm2_decode_init,
-    .init_static_data = qdm2_init_static_data,
     .close            = qdm2_decode_close,
     .decode           = qdm2_decode_frame,
-    .capabilities     = CODEC_CAP_DR1,
+    .capabilities     = AV_CODEC_CAP_DR1,
 };