adding ffmpeg specific version

1 files changed, 2014 insertions, 0 deletions

diff --git a/ffmpeg1/libavcodec/qdm2.c b/ffmpeg1/libavcodec/qdm2.c
new file mode 100644
index 0000000..108c327
--- /dev/null
+++ b/ffmpeg1/libavcodec/qdm2.c
@@ -0,0 +1,2014 @@
+/*
+ * QDM2 compatible decoder
+ * Copyright (c) 2003 Ewald Snel
+ * Copyright (c) 2005 Benjamin Larsson
+ * Copyright (c) 2005 Alex Beregszaszi
+ * Copyright (c) 2005 Roberto Togni
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * QDM2 decoder
+ * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
+ *
+ * The decoder is not perfect yet, there are still some distortions
+ * especially on files encoded with 16 or 8 subbands.
+ */
+
+#include <math.h>
+#include <stddef.h>
+#include <stdio.h>
+
+#define BITSTREAM_READER_LE
+#include "libavutil/channel_layout.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "rdft.h"
+#include "mpegaudiodsp.h"
+#include "mpegaudio.h"
+
+#include "qdm2data.h"
+#include "qdm2_tablegen.h"
+
+#undef NDEBUG
+#include <assert.h>
+
+
+#define QDM2_LIST_ADD(list, size, packet) \
+do { \
+      if (size > 0) { \
+    list[size - 1].next = &list[size]; \
+      } \
+      list[size].packet = packet; \
+      list[size].next = NULL; \
+      size++; \
+} while(0)
+
+// Result is 8, 16 or 30
+#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
+
+#define FIX_NOISE_IDX(noise_idx) \
+  if ((noise_idx) >= 3840) \
+    (noise_idx) -= 3840; \
+
+#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
+
+#define SAMPLES_NEEDED \
+     av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
+
+#define SAMPLES_NEEDED_2(why) \
+     av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
+
+#define QDM2_MAX_FRAME_SIZE 512
+
+typedef int8_t sb_int8_array[2][30][64];
+
+/**
+ * Subpacket
+ */
+typedef struct {
+    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)
+} QDM2SubPacket;
+
+/**
+ * A node in the subpacket list
+ */
+typedef struct QDM2SubPNode {
+    QDM2SubPacket *packet;      ///< packet
+    struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
+} QDM2SubPNode;
+
+typedef struct {
+    float re;
+    float im;
+} QDM2Complex;
+
+typedef struct {
+    float level;
+    QDM2Complex *complex;
+    const float *table;
+    int   phase;
+    int   phase_shift;
+    int   duration;
+    short time_index;
+    short cutoff;
+} FFTTone;
+
+typedef struct {
+    int16_t sub_packet;
+    uint8_t channel;
+    int16_t offset;
+    int16_t exp;
+    uint8_t phase;
+} FFTCoefficient;
+
+typedef struct {
+    DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
+} QDM2FFT;
+
+/**
+ * QDM2 decoder context
+ */
+typedef struct {
+    /// Parameters from codec header, do not change during playback
+    int nb_channels;         ///< number of channels
+    int channels;            ///< number of channels
+    int group_size;          ///< size of frame group (16 frames per group)
+    int fft_size;            ///< size of FFT, in complex numbers
+    int checksum_size;       ///< size of data block, used also for checksum
+
+    /// Parameters built from header parameters, do not change during playback
+    int group_order;         ///< order of frame group
+    int fft_order;           ///< order of FFT (actually fftorder+1)
+    int frame_size;          ///< size of data frame
+    int frequency_range;
+    int sub_sampling;        ///< subsampling: 0=25%, 1=50%, 2=100% */
+    int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
+    int cm_table_select;     ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
+
+    /// Packets and packet lists
+    QDM2SubPacket sub_packets[16];      ///< the packets themselves
+    QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
+    QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
+    int sub_packets_B;                  ///< number of packets on 'B' list
+    QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
+    QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
+
+    /// FFT and tones
+    FFTTone fft_tones[1000];
+    int fft_tone_start;
+    int fft_tone_end;
+    FFTCoefficient fft_coefs[1000];
+    int fft_coefs_index;
+    int fft_coefs_min_index[5];
+    int fft_coefs_max_index[5];
+    int fft_level_exp[6];
+    RDFTContext rdft_ctx;
+    QDM2FFT fft;
+
+    /// I/O data
+    const uint8_t *compressed_data;
+    int compressed_size;
+    float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
+
+    /// Synthesis filter
+    MPADSPContext mpadsp;
+    DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
+    int synth_buf_offset[MPA_MAX_CHANNELS];
+    DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+    DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
+
+    /// Mixed temporary data used in decoding
+    float tone_level[MPA_MAX_CHANNELS][30][64];
+    int8_t coding_method[MPA_MAX_CHANNELS][30][64];
+    int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
+    int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
+    int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
+    int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
+    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
+    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
+    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
+
+    // Flags
+    int has_errors;         ///< packet has errors
+    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
+    int do_synth_filter;    ///< used to perform or skip synthesis filter
+
+    int sub_packet;
+    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 av_cold void qdm2_init_vlc(void)
+{
+    static int vlcs_initialized = 0;
+    static VLC_TYPE qdm2_table[3838][2];
+
+    if (!vlcs_initialized) {
+
+        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);
+
+        vlcs_initialized=1;
+    }
+}
+
+static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
+{
+    int value;
+
+    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
+
+    /* stage-2, 3 bits exponent escape sequence */
+    if (value-- == 0)
+        value = get_bits (gb, get_bits (gb, 3) + 1);
+
+    /* stage-3, optional */
+    if (flag) {
+        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));
+        value = tmp;
+    }
+
+    return value;
+}
+
+
+static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
+{
+    int value = qdm2_get_vlc (gb, vlc, 0, depth);
+
+    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
+}
+
+
+/**
+ * QDM2 checksum
+ *
+ * @param data      pointer to data to be checksum'ed
+ * @param length    data length
+ * @param value     checksum value
+ *
+ * @return          0 if checksum is OK
+ */
+static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
+    int i;
+
+    for (i=0; i < length; i++)
+        value -= data[i];
+
+    return (uint16_t)(value & 0xffff);
+}
+
+
+/**
+ * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
+ *
+ * @param gb            bitreader context
+ * @param sub_packet    packet under analysis
+ */
+static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
+{
+    sub_packet->type = get_bits (gb, 8);
+
+    if (sub_packet->type == 0) {
+        sub_packet->size = 0;
+        sub_packet->data = NULL;
+    } else {
+        sub_packet->size = get_bits (gb, 8);
+
+      if (sub_packet->type & 0x80) {
+          sub_packet->size <<= 8;
+          sub_packet->size  |= get_bits (gb, 8);
+          sub_packet->type  &= 0x7f;
+      }
+
+      if (sub_packet->type == 0x7f)
+          sub_packet->type |= (get_bits (gb, 8) << 8);
+
+      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
+    }
+
+    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
+        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
+}
+
+
+/**
+ * Return node pointer to first packet of requested type in list.
+ *
+ * @param list    list of subpackets to be scanned
+ * @param type    type of searched subpacket
+ * @return        node pointer for subpacket if found, else NULL
+ */
+static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
+{
+    while (list != NULL && list->packet != NULL) {
+        if (list->packet->type == type)
+            return list;
+        list = list->next;
+    }
+    return NULL;
+}
+
+
+/**
+ * Replace 8 elements with their average value.
+ * Called by qdm2_decode_superblock before starting subblock decoding.
+ *
+ * @param q       context
+ */
+static void average_quantized_coeffs (QDM2Context *q)
+{
+    int i, j, n, ch, sum;
+
+    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
+
+    for (ch = 0; ch < q->nb_channels; ch++)
+        for (i = 0; i < n; i++) {
+            sum = 0;
+
+            for (j = 0; j < 8; j++)
+                sum += q->quantized_coeffs[ch][i][j];
+
+            sum /= 8;
+            if (sum > 0)
+                sum--;
+
+            for (j=0; j < 8; j++)
+                q->quantized_coeffs[ch][i][j] = sum;
+        }
+}
+
+
+/**
+ * Build subband samples with noise weighted by q->tone_level.
+ * Called by synthfilt_build_sb_samples.
+ *
+ * @param q     context
+ * @param sb    subband index
+ */
+static void build_sb_samples_from_noise (QDM2Context *q, int sb)
+{
+    int ch, j;
+
+    FIX_NOISE_IDX(q->noise_idx);
+
+    if (!q->nb_channels)
+        return;
+
+    for (ch = 0; ch < q->nb_channels; ch++)
+        for (j = 0; j < 64; j++) {
+            q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+            q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+        }
+}
+
+
+/**
+ * Called while processing data from subpackets 11 and 12.
+ * Used after making changes to coding_method array.
+ *
+ * @param sb               subband index
+ * @param channels         number of channels
+ * @param coding_method    q->coding_method[0][0][0]
+ */
+static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
+{
+    int j,k;
+    int ch;
+    int run, case_val;
+    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};
+
+    for (ch = 0; ch < channels; ch++) {
+        for (j = 0; j < 64; ) {
+            if((coding_method[ch][sb][j] - 8) > 22) {
+                run = 1;
+                case_val = 8;
+            } else {
+                switch (switchtable[coding_method[ch][sb][j]-8]) {
+                    case 0: run = 10; case_val = 10; break;
+                    case 1: run = 1; case_val = 16; break;
+                    case 2: run = 5; case_val = 24; break;
+                    case 3: run = 3; case_val = 30; break;
+                    case 4: run = 1; case_val = 30; break;
+                    case 5: run = 1; case_val = 8; break;
+                    default: run = 1; case_val = 8; break;
+                }
+            }
+            for (k = 0; k < run; k++)
+                if (j + k < 128)
+                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
+                        if (k > 0) {
+                           SAMPLES_NEEDED
+                            //not debugged, almost never used
+                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
+                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
+                        }
+            j += run;
+        }
+    }
+}
+
+
+/**
+ * Related to synthesis filter
+ * Called by process_subpacket_10
+ *
+ * @param q       context
+ * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
+ */
+static void fill_tone_level_array (QDM2Context *q, int flag)
+{
+    int i, sb, ch, sb_used;
+    int tmp, tab;
+
+    for (ch = 0; ch < q->nb_channels; ch++)
+        for (sb = 0; sb < 30; sb++)
+            for (i = 0; i < 8; i++) {
+                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
+                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
+                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
+                else
+                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
+                if(tmp < 0)
+                    tmp += 0xff;
+                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
+            }
+
+    sb_used = QDM2_SB_USED(q->sub_sampling);
+
+    if ((q->superblocktype_2_3 != 0) && !flag) {
+        for (sb = 0; sb < sb_used; sb++)
+            for (ch = 0; ch < q->nb_channels; ch++)
+                for (i = 0; i < 64; i++) {
+                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
+                    if (q->tone_level_idx[ch][sb][i] < 0)
+                        q->tone_level[ch][sb][i] = 0;
+                    else
+                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
+                }
+    } else {
+        tab = q->superblocktype_2_3 ? 0 : 1;
+        for (sb = 0; sb < sb_used; sb++) {
+            if ((sb >= 4) && (sb <= 23)) {
+                for (ch = 0; ch < q->nb_channels; ch++)
+                    for (i = 0; i < 64; i++) {
+                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
+                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
+                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
+                              q->tone_level_idx_hi2[ch][sb - 4];
+                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
+                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
+                            q->tone_level[ch][sb][i] = 0;
+                        else
+                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+                }
+            } else {
+                if (sb > 4) {
+                    for (ch = 0; ch < q->nb_channels; ch++)
+                        for (i = 0; i < 64; i++) {
+                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
+                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
+                                  q->tone_level_idx_hi2[ch][sb - 4];
+                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
+                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
+                                q->tone_level[ch][sb][i] = 0;
+                            else
+                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+                    }
+                } else {
+                    for (ch = 0; ch < q->nb_channels; ch++)
+                        for (i = 0; i < 64; i++) {
+                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
+                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
+                                q->tone_level[ch][sb][i] = 0;
+                            else
+                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+                        }
+                }
+            }
+        }
+    }
+
+    return;
+}
+
+
+/**
+ * Related to synthesis filter
+ * Called by process_subpacket_11
+ * c is built with data from subpacket 11
+ * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
+ *
+ * @param tone_level_idx
+ * @param tone_level_idx_temp
+ * @param coding_method        q->coding_method[0][0][0]
+ * @param nb_channels          number of channels
+ * @param c                    coming from subpacket 11, passed as 8*c
+ * @param superblocktype_2_3   flag based on superblock packet type
+ * @param cm_table_select      q->cm_table_select
+ */
+static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
+                sb_int8_array coding_method, int nb_channels,
+                int c, int superblocktype_2_3, int cm_table_select)
+{
+    int ch, sb, j;
+    int tmp, acc, esp_40, comp;
+    int add1, add2, add3, add4;
+    int64_t multres;
+
+    if (!superblocktype_2_3) {
+        /* This case is untested, no samples available */
+        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;
+                    if (add1 < 0)
+                        add1 = 0;
+                    add2 = add3 = add4 = 0;
+                    if (sb > 1) {
+                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
+                        if (add2 < 0)
+                            add2 = 0;
+                    }
+                    if (sb > 0) {
+                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
+                        if (add3 < 0)
+                            add3 = 0;
+                    }
+                    if (sb < 29) {
+                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
+                        if (add4 < 0)
+                            add4 = 0;
+                    }
+                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
+                    if (tmp < 0)
+                        tmp = 0;
+                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
+                }
+                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 = 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++)
+                    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 (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++)
+                for (j = 0; j < 64; j++)
+                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
+    }
+
+    return;
+}
+
+
+/**
+ *
+ * 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 sb 8-sb_used
+ *
+ * @param q         context
+ * @param gb        bitreader context
+ * @param length    packet length in bits
+ * @param sb_min    lower subband processed (sb_min included)
+ * @param sb_max    higher subband processed (sb_max excluded)
+ */
+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;
+    int type34_first;
+    float type34_div = 0;
+    float type34_predictor;
+    float samples[10], sign_bits[16];
+
+    if (length == 0) {
+        // If no data use noise
+        for (sb=sb_min; sb < sb_max; sb++)
+            build_sb_samples_from_noise (q, sb);
+
+        return 0;
+    }
+
+    for (sb = sb_min; sb < sb_max; sb++) {
+        FIX_NOISE_IDX(q->noise_idx);
+
+        channels = q->nb_channels;
+
+        if (q->nb_channels <= 1 || sb < 12)
+            joined_stereo = 0;
+        else if (sb >= 24)
+            joined_stereo = 1;
+        else
+            joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
+
+        if (joined_stereo) {
+            if (get_bits_left(gb) >= 16)
+                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];
+
+            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
+            channels = 1;
+        }
+
+        for (ch = 0; ch < channels; ch++) {
+            zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
+            type34_predictor = 0.0;
+            type34_first = 1;
+
+            for (j = 0; j < 128; ) {
+                switch (q->coding_method[ch][sb][j / 2]) {
+                    case 8:
+                        if (get_bits_left(gb) >= 10) {
+                            if (zero_encoding) {
+                                for (k = 0; k < 5; k++) {
+                                    if ((j + 2 * k) >= 128)
+                                        break;
+                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
+                                }
+                            } 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]];
+                            }
+                            for (k = 0; k < 5; k++)
+                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        } else {
+                            for (k = 0; k < 10; k++)
+                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        }
+                        run = 10;
+                        break;
+
+                    case 10:
+                        if (get_bits_left(gb) >= 1) {
+                            float f = 0.81;
+
+                            if (get_bits1(gb))
+                                f = -f;
+                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
+                            samples[0] = f;
+                        } else {
+                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        }
+                        run = 1;
+                        break;
+
+                    case 16:
+                        if (get_bits_left(gb) >= 10) {
+                            if (zero_encoding) {
+                                for (k = 0; k < 5; k++) {
+                                    if ((j + k) >= 128)
+                                        break;
+                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(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]];
+                            }
+                        } else {
+                            for (k = 0; k < 5; k++)
+                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        }
+                        run = 5;
+                        break;
+
+                    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 {
+                            for (k = 0; k < 3; k++)
+                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        }
+                        run = 3;
+                        break;
+
+                    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)) {
+                                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);
+
+                        run = 1;
+                        break;
+
+                    case 34:
+                        if (get_bits_left(gb) >= 7) {
+                            if (type34_first) {
+                                type34_div = (float)(1 << get_bits(gb, 2));
+                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
+                                type34_predictor = samples[0];
+                                type34_first = 0;
+                            } else {
+                                unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
+                                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);
+                        }
+                        run = 1;
+                        break;
+
+                    default:
+                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
+                        run = 1;
+                        break;
+                }
+
+                if (joined_stereo) {
+                    float tmp[10][MPA_MAX_CHANNELS];
+                    for (k = 0; k < run; k++) {
+                        tmp[k][0] = samples[k];
+                        if ((j + k) < 128)
+                            tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
+                    }
+                    for (chs = 0; chs < q->nb_channels; chs++)
+                        for (k = 0; k < run; k++)
+                            if ((j + k) < 128)
+                                q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
+                } else {
+                    for (k = 0; k < run; k++)
+                        if ((j + k) < 128)
+                            q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
+                }
+
+                j += run;
+            } // j loop
+        } // channel loop
+    } // subband loop
+    return 0;
+}
+
+
+/**
+ * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
+ * This is similar to process_subpacket_9, but for a single channel and for element [0]
+ * same VLC tables as process_subpacket_9 are used.
+ *
+ * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
+ * @param gb        bitreader context
+ */
+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 -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)
+            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)
+            return -1;
+        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
+
+        for (k = 1; k <= run; k++)
+            quantized_coeffs[i + k] = (level + ((k * diff) / run));
+
+        level += diff;
+        i += run;
+    }
+    return 0;
+}
+
+
+/**
+ * Related to synthesis filter, process data from packet 10
+ * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
+ * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
+ *
+ * @param q         context
+ * @param gb        bitreader context
+ */
+static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
+{
+    int sb, j, k, n, ch;
+
+    for (ch = 0; ch < q->nb_channels; ch++) {
+        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
+
+        if (get_bits_left(gb) < 16) {
+            memset(q->quantized_coeffs[ch][0], 0, 8);
+            break;
+        }
+    }
+
+    n = q->sub_sampling + 1;
+
+    for (sb = 0; sb < n; sb++)
+        for (ch = 0; ch < q->nb_channels; ch++)
+            for (j = 0; j < 8; j++) {
+                if (get_bits_left(gb) < 1)
+                    break;
+                if (get_bits1(gb)) {
+                    for (k=0; k < 8; k++) {
+                        if (get_bits_left(gb) < 16)
+                            break;
+                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
+                    }
+                } else {
+                    for (k=0; k < 8; k++)
+                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
+                }
+            }
+
+    n = QDM2_SB_USED(q->sub_sampling) - 4;
+
+    for (sb = 0; sb < n; sb++)
+        for (ch = 0; ch < q->nb_channels; ch++) {
+            if (get_bits_left(gb) < 16)
+                break;
+            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
+            if (sb > 19)
+                q->tone_level_idx_hi2[ch][sb] -= 16;
+            else
+                for (j = 0; j < 8; j++)
+                    q->tone_level_idx_mid[ch][sb][j] = -16;
+        }
+
+    n = QDM2_SB_USED(q->sub_sampling) - 5;
+
+    for (sb = 0; sb < n; sb++)
+        for (ch = 0; ch < q->nb_channels; ch++)
+            for (j = 0; j < 8; j++) {
+                if (get_bits_left(gb) < 16)
+                    break;
+                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
+            }
+}
+
+/**
+ * Process subpacket 9, init quantized_coeffs with data from it
+ *
+ * @param q       context
+ * @param node    pointer to node with packet
+ */
+static int process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
+{
+    GetBitContext gb;
+    int i, j, k, n, ch, run, level, diff;
+
+    init_get_bits(&gb, node->packet->data, node->packet->size*8);
+
+    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
+
+    for (i = 1; i < n; i++)
+        for (ch=0; ch < q->nb_channels; ch++) {
+            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
+            q->quantized_coeffs[ch][i][0] = level;
+
+            for (j = 0; j < (8 - 1); ) {
+                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));
+
+                level += diff;
+                j += run;
+            }
+        }
+
+    for (ch = 0; ch < q->nb_channels; ch++)
+        for (i = 0; i < 8; i++)
+            q->quantized_coeffs[ch][0][i] = 0;
+
+    return 0;
+}
+
+
+/**
+ * Process subpacket 10 if not null, else
+ *
+ * @param q         context
+ * @param node      pointer to node with packet
+ */
+static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
+{
+    GetBitContext gb;
+
+    if (node) {
+        init_get_bits(&gb, node->packet->data, node->packet->size * 8);
+        init_tone_level_dequantization(q, &gb);
+        fill_tone_level_array(q, 1);
+    } else {
+        fill_tone_level_array(q, 0);
+    }
+}
+
+
+/**
+ * Process subpacket 11
+ *
+ * @param q         context
+ * @param node      pointer to node with packet
+ */
+static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
+{
+    GetBitContext gb;
+    int length = 0;
+
+    if (node) {
+        length = node->packet->size * 8;
+        init_get_bits(&gb, node->packet->data, length);
+    }
+
+    if (length >= 32) {
+        int c = get_bits (&gb, 13);
+
+        if (c > 3)
+            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
+                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
+    }
+
+    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
+}
+
+
+/**
+ * Process subpacket 12
+ *
+ * @param q         context
+ * @param node      pointer to node with packet
+ */
+static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
+{
+    GetBitContext gb;
+    int length = 0;
+
+    if (node) {
+        length = node->packet->size * 8;
+        init_get_bits(&gb, node->packet->data, length);
+    }
+
+    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
+}
+
+/**
+ * Process new subpackets for synthesis filter
+ *
+ * @param q       context
+ * @param list    list with synthesis filter packets (list D)
+ */
+static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
+{
+    QDM2SubPNode *nodes[4];
+
+    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
+    if (nodes[0] != NULL)
+        process_subpacket_9(q, nodes[0]);
+
+    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
+    if (nodes[1] != NULL)
+        process_subpacket_10(q, nodes[1]);
+    else
+        process_subpacket_10(q, NULL);
+
+    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
+    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
+        process_subpacket_11(q, nodes[2]);
+    else
+        process_subpacket_11(q, NULL);
+
+    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
+    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
+        process_subpacket_12(q, nodes[3]);
+    else
+        process_subpacket_12(q, NULL);
+}
+
+
+/**
+ * Decode superblock, fill packet lists.
+ *
+ * @param q    context
+ */
+static void qdm2_decode_super_block (QDM2Context *q)
+{
+    GetBitContext gb;
+    QDM2SubPacket header, *packet;
+    int i, packet_bytes, sub_packet_size, sub_packets_D;
+    unsigned int next_index = 0;
+
+    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
+    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
+    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
+
+    q->sub_packets_B = 0;
+    sub_packets_D = 0;
+
+    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
+
+    init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
+    qdm2_decode_sub_packet_header(&gb, &header);
+
+    if (header.type < 2 || header.type >= 8) {
+        q->has_errors = 1;
+        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
+        return;
+    }
+
+    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
+    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
+
+    init_get_bits(&gb, header.data, header.size*8);
+
+    if (header.type == 2 || header.type == 4 || header.type == 5) {
+        int csum  = 257 * get_bits(&gb, 8);
+            csum +=   2 * get_bits(&gb, 8);
+
+        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
+
+        if (csum != 0) {
+            q->has_errors = 1;
+            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
+            return;
+        }
+    }
+
+    q->sub_packet_list_B[0].packet = NULL;
+    q->sub_packet_list_D[0].packet = NULL;
+
+    for (i = 0; i < 6; i++)
+        if (--q->fft_level_exp[i] < 0)
+            q->fft_level_exp[i] = 0;
+
+    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) {
+            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
+
+            /* seek to next block */
+            init_get_bits(&gb, header.data, header.size*8);
+            skip_bits(&gb, next_index*8);
+
+            if (next_index >= header.size)
+                break;
+        }
+
+        /* decode subpacket */
+        packet = &q->sub_packets[i];
+        qdm2_decode_sub_packet_header(&gb, packet);
+        next_index = packet->size + get_bits_count(&gb) / 8;
+        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
+
+        if (packet->type == 0)
+            break;
+
+        if (sub_packet_size > packet_bytes) {
+            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
+                break;
+            packet->size += packet_bytes - sub_packet_size;
+        }
+
+        packet_bytes -= sub_packet_size;
+
+        /* add subpacket to 'all subpackets' list */
+        q->sub_packet_list_A[i].packet = packet;
+
+        /* add subpacket to related list */
+        if (packet->type == 8) {
+            SAMPLES_NEEDED_2("packet type 8");
+            return;
+        } else if (packet->type >= 9 && packet->type <= 12) {
+            /* packets for MPEG Audio like Synthesis Filter */
+            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
+        } else if (packet->type == 13) {
+            for (j = 0; j < 6; j++)
+                q->fft_level_exp[j] = get_bits(&gb, 6);
+        } else if (packet->type == 14) {
+            for (j = 0; j < 6; j++)
+                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
+        } else if (packet->type == 15) {
+            SAMPLES_NEEDED_2("packet type 15")
+            return;
+        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
+            /* packets for FFT */
+            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
+        }
+    } // Packet bytes loop
+
+/* **************************************************************** */
+    if (q->sub_packet_list_D[0].packet != NULL) {
+        process_synthesis_subpackets(q, q->sub_packet_list_D);
+        q->do_synth_filter = 1;
+    } else if (q->do_synth_filter) {
+        process_subpacket_10(q, NULL);
+        process_subpacket_11(q, NULL);
+        process_subpacket_12(q, NULL);
+    }
+/* **************************************************************** */
+}
+
+
+static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
+                       int offset, int duration, int channel,
+                       int exp, int phase)
+{
+    if (q->fft_coefs_min_index[duration] < 0)
+        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
+
+    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
+    q->fft_coefs[q->fft_coefs_index].channel = channel;
+    q->fft_coefs[q->fft_coefs_index].offset = offset;
+    q->fft_coefs[q->fft_coefs_index].exp = exp;
+    q->fft_coefs[q->fft_coefs_index].phase = phase;
+    q->fft_coefs_index++;
+}
+
+
+static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
+{
+    int channel, stereo, phase, exp;
+    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
+    int local_int_14, stereo_exp, local_int_20, local_int_28;
+    int n, offset;
+
+    local_int_4 = 0;
+    local_int_28 = 0;
+    local_int_20 = 2;
+    local_int_8 = (4 - duration);
+    local_int_10 = 1 << (q->group_order - duration - 1);
+    offset = 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;
+                    local_int_28 += (1 << local_int_8);
+                } else {
+                    local_int_4 += 8*local_int_10;
+                    local_int_28 += (8 << local_int_8);
+                }
+            }
+            offset += (n - 2);
+        } else {
+            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
+            while (offset >= (local_int_10 - 1)) {
+                offset += (1 - (local_int_10 - 1));
+                local_int_4  += local_int_10;
+                local_int_28 += (1 << local_int_8);
+            }
+        }
+
+        if (local_int_4 >= q->group_size)
+            return;
+
+        local_int_14 = (offset >> local_int_8);
+        if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
+            return;
+
+        if (q->nb_channels > 1) {
+            channel = get_bits1(gb);
+            stereo = get_bits1(gb);
+        } else {
+            channel = 0;
+            stereo = 0;
+        }
+
+        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
+        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
+        exp = (exp < 0) ? 0 : exp;
+
+        phase = get_bits(gb, 3);
+        stereo_exp = 0;
+        stereo_phase = 0;
+
+        if (stereo) {
+            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
+            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
+            if (stereo_phase < 0)
+                stereo_phase += 8;
+        }
+
+        if (q->frequency_range > (local_int_14 + 1)) {
+            int sub_packet = (local_int_20 + local_int_28);
+
+            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
+            if (stereo)
+                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
+        }
+
+        offset++;
+    }
+}
+
+
+static void qdm2_decode_fft_packets (QDM2Context *q)
+{
+    int i, j, min, max, value, type, unknown_flag;
+    GetBitContext gb;
+
+    if (q->sub_packet_list_B[0].packet == NULL)
+        return;
+
+    /* reset minimum indexes for FFT coefficients */
+    q->fft_coefs_index = 0;
+    for (i=0; i < 5; i++)
+        q->fft_coefs_min_index[i] = -1;
+
+    /* process subpackets ordered by type, largest type first */
+    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
+        QDM2SubPacket *packet= NULL;
+
+        /* find subpacket with largest type less than max */
+        for (j = 0, min = 0; j < q->sub_packets_B; j++) {
+            value = q->sub_packet_list_B[j].packet->type;
+            if (value > min && value < max) {
+                min = value;
+                packet = q->sub_packet_list_B[j].packet;
+            }
+        }
+
+        max = min;
+
+        /* check for errors (?) */
+        if (!packet)
+            return;
+
+        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
+            return;
+
+        /* decode FFT tones */
+        init_get_bits (&gb, packet->data, packet->size*8);
+
+        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
+            unknown_flag = 1;
+        else
+            unknown_flag = 0;
+
+        type = packet->type;
+
+        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
+            int duration = q->sub_sampling + 5 - (type & 15);
+
+            if (duration >= 0 && duration < 4)
+                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
+        } else if (type == 31) {
+            for (j=0; j < 4; j++)
+                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
+        } else if (type == 46) {
+            for (j=0; j < 6; j++)
+                q->fft_level_exp[j] = get_bits(&gb, 6);
+            for (j=0; j < 4; j++)
+            qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
+        }
+    } // Loop on B packets
+
+    /* calculate maximum indexes for FFT coefficients */
+    for (i = 0, j = -1; i < 5; i++)
+        if (q->fft_coefs_min_index[i] >= 0) {
+            if (j >= 0)
+                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
+            j = i;
+        }
+    if (j >= 0)
+        q->fft_coefs_max_index[j] = q->fft_coefs_index;
+}
+
+
+static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
+{
+   float level, f[6];
+   int i;
+   QDM2Complex c;
+   const double iscale = 2.0*M_PI / 512.0;
+
+    tone->phase += tone->phase_shift;
+
+    /* calculate current level (maximum amplitude) of tone */
+    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
+    c.im = level * sin(tone->phase*iscale);
+    c.re = level * cos(tone->phase*iscale);
+
+    /* generate FFT coefficients for tone */
+    if (tone->duration >= 3 || tone->cutoff >= 3) {
+        tone->complex[0].im += c.im;
+        tone->complex[0].re += c.re;
+        tone->complex[1].im -= c.im;
+        tone->complex[1].re -= c.re;
+    } else {
+        f[1] = -tone->table[4];
+        f[0] =  tone->table[3] - tone->table[0];
+        f[2] =  1.0 - tone->table[2] - tone->table[3];
+        f[3] =  tone->table[1] + tone->table[4] - 1.0;
+        f[4] =  tone->table[0] - tone->table[1];
+        f[5] =  tone->table[2];
+        for (i = 0; i < 2; i++) {
+            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
+            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
+        }
+        for (i = 0; i < 4; i++) {
+            tone->complex[i].re += c.re * f[i+2];
+            tone->complex[i].im += c.im * f[i+2];
+        }
+    }
+
+    /* copy the tone if it has not yet died out */
+    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
+      memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
+      q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
+    }
+}
+
+
+static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
+{
+    int i, j, ch;
+    const double iscale = 0.25 * M_PI;
+
+    for (ch = 0; ch < q->channels; ch++) {
+        memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
+    }
+
+
+    /* apply FFT tones with duration 4 (1 FFT period) */
+    if (q->fft_coefs_min_index[4] >= 0)
+        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
+            float level;
+            QDM2Complex c;
+
+            if (q->fft_coefs[i].sub_packet != sub_packet)
+                break;
+
+            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
+            level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63];
+
+            c.re = level * cos(q->fft_coefs[i].phase * iscale);
+            c.im = level * sin(q->fft_coefs[i].phase * iscale);
+            q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
+            q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
+            q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
+            q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
+        }
+
+    /* generate existing FFT tones */
+    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
+        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
+        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
+    }
+
+    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
+    for (i = 0; i < 4; i++)
+        if (q->fft_coefs_min_index[i] >= 0) {
+            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
+                int offset, four_i;
+                FFTTone tone;
+
+                if (q->fft_coefs[j].sub_packet != sub_packet)
+                    break;
+
+                four_i = (4 - i);
+                offset = q->fft_coefs[j].offset >> four_i;
+                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
+
+                if (offset < q->frequency_range) {
+                    if (offset < 2)
+                        tone.cutoff = offset;
+                    else
+                        tone.cutoff = (offset >= 60) ? 3 : 2;
+
+                    tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
+                    tone.complex = &q->fft.complex[ch][offset];
+                    tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
+                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
+                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
+                    tone.duration = i;
+                    tone.time_index = 0;
+
+                    qdm2_fft_generate_tone(q, &tone);
+                }
+            }
+            q->fft_coefs_min_index[i] = j;
+        }
+}
+
+
+static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
+{
+    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
+    float *out = q->output_buffer + channel;
+    int i;
+    q->fft.complex[channel][0].re *= 2.0f;
+    q->fft.complex[channel][0].im = 0.0f;
+    q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
+    /* add samples to output buffer */
+    for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
+        out[0]           += q->fft.complex[channel][i].re * gain;
+        out[q->channels] += q->fft.complex[channel][i].im * gain;
+        out += 2 * q->channels;
+    }
+}
+
+
+/**
+ * @param q        context
+ * @param index    subpacket number
+ */
+static void qdm2_synthesis_filter (QDM2Context *q, int index)
+{
+    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
+
+    /* copy sb_samples */
+    sb_used = QDM2_SB_USED(q->sub_sampling);
+
+    for (ch = 0; ch < q->channels; ch++)
+        for (i = 0; i < 8; i++)
+            for (k=sb_used; k < SBLIMIT; k++)
+                q->sb_samples[ch][(8 * index) + i][k] = 0;
+
+    for (ch = 0; ch < q->nb_channels; ch++) {
+        float *samples_ptr = q->samples + ch;
+
+        for (i = 0; i < 8; i++) {
+            ff_mpa_synth_filter_float(&q->mpadsp,
+                q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+                ff_mpa_synth_window_float, &dither_state,
+                samples_ptr, q->nb_channels,
+                q->sb_samples[ch][(8 * index) + i]);
+            samples_ptr += 32 * q->nb_channels;
+        }
+    }
+
+    /* add samples to output buffer */
+    sub_sampling = (4 >> q->sub_sampling);
+
+    for (ch = 0; ch < q->channels; ch++)
+        for (i = 0; i < q->frame_size; i++)
+            q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
+}
+
+
+/**
+ * Init static data (does not depend on specific file)
+ *
+ * @param q    context
+ */
+static av_cold void qdm2_init(QDM2Context *q) {
+    static int initialized = 0;
+
+    if (initialized != 0)
+        return;
+    initialized = 1;
+
+    qdm2_init_vlc();
+    ff_mpa_synth_init_float(ff_mpa_synth_window_float);
+    softclip_table_init();
+    rnd_table_init();
+    init_noise_samples();
+
+    av_log(NULL, AV_LOG_DEBUG, "init done\n");
+}
+
+
+/**
+ * Init parameters from codec extradata
+ */
+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;
+
+    /* extradata parsing
+
+    Structure:
+    wave {
+        frma (QDM2)
+        QDCA
+        QDCP
+    }
+
+    32  size (including this field)
+    32  tag (=frma)
+    32  type (=QDM2 or QDMC)
+
+    32  size (including this field, in bytes)
+    32  tag (=QDCA) // maybe mandatory parameters
+    32  unknown (=1)
+    32  channels (=2)
+    32  samplerate (=44100)
+    32  bitrate (=96000)
+    32  block size (=4096)
+    32  frame size (=256) (for one channel)
+    32  packet size (=1300)
+
+    32  size (including this field, in bytes)
+    32  tag (=QDCP) // maybe some tuneable parameters
+    32  float1 (=1.0)
+    32  zero ?
+    32  float2 (=1.0)
+    32  float3 (=1.0)
+    32  unknown (27)
+    32  unknown (8)
+    32  zero ?
+    */
+
+    if (!avctx->extradata || (avctx->extradata_size < 48)) {
+        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
+        return -1;
+    }
+
+    extradata = avctx->extradata;
+    extradata_size = avctx->extradata_size;
+
+    while (extradata_size > 7) {
+        if (!memcmp(extradata, "frmaQDM", 7))
+            break;
+        extradata++;
+        extradata_size--;
+    }
+
+    if (extradata_size < 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;
+    }
+
+    extradata += 8;
+    extradata_size -= 8;
+
+    size = AV_RB32(extradata);
+
+    if(size > extradata_size){
+        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
+               extradata_size, size);
+        return -1;
+    }
+
+    extradata += 4;
+    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
+    if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
+        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
+        return -1;
+    }
+
+    extradata += 8;
+
+    avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
+    extradata += 4;
+    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);
+    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;
+    }
+
+    s->fft_order = av_log2(s->fft_size) + 1;
+
+    // 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;
+
+    s->sub_sampling = s->fft_order - 7;
+    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
+
+    switch ((s->sub_sampling * 2 + s->channels - 1)) {
+        case 0: tmp = 40; break;
+        case 1: tmp = 48; break;
+        case 2: tmp = 56; break;
+        case 3: tmp = 72; break;
+        case 4: tmp = 80; break;
+        case 5: tmp = 100;break;
+        default: tmp=s->sub_sampling; break;
+    }
+    tmp_val = 0;
+    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
+    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
+    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
+    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)
+        s->coeff_per_sb_select = 0;
+    else if (tmp <= 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;
+    }
+
+    ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+    ff_mpadsp_init(&s->mpadsp);
+
+    qdm2_init(s);
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    return 0;
+}
+
+
+static av_cold int qdm2_decode_close(AVCodecContext *avctx)
+{
+    QDM2Context *s = avctx->priv_data;
+
+    ff_rdft_end(&s->rdft_ctx);
+
+    return 0;
+}
+
+
+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;
+
+    /* copy old block, clear new block of output samples */
+    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
+    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
+
+    /* decode block of QDM2 compressed data */
+    if (q->sub_packet == 0) {
+        q->has_errors = 0; // zero it for a new super block
+        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
+        qdm2_decode_super_block(q);
+    }
+
+    /* parse subpackets */
+    if (!q->has_errors) {
+        if (q->sub_packet == 2)
+            qdm2_decode_fft_packets(q);
+
+        qdm2_fft_tone_synthesizer(q, q->sub_packet);
+    }
+
+    /* sound synthesis stage 1 (FFT) */
+    for (ch = 0; ch < q->channels; ch++) {
+        qdm2_calculate_fft(q, ch, q->sub_packet);
+
+        if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
+            SAMPLES_NEEDED_2("has errors, and C list is not empty")
+            return -1;
+        }
+    }
+
+    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
+    if (!q->has_errors && q->do_synth_filter)
+        qdm2_synthesis_filter(q, q->sub_packet);
+
+    q->sub_packet = (q->sub_packet + 1) % 16;
+
+    /* clip and convert output float[] to 16bit signed samples */
+    for (i = 0; i < frame_size; i++) {
+        int value = (int)q->output_buffer[i];
+
+        if (value > SOFTCLIP_THRESHOLD)
+            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
+        else if (value < -SOFTCLIP_THRESHOLD)
+            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
+
+        out[i] = value;
+    }
+
+    return 0;
+}
+
+
+static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    QDM2Context *s = avctx->priv_data;
+    int16_t *out;
+    int i, ret;
+
+    if(!buf)
+        return 0;
+    if(buf_size < s->checksum_size)
+        return -1;
+
+    /* get output buffer */
+    frame->nb_samples = 16 * s->frame_size;
+    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;
+        out += s->channels * s->frame_size;
+    }
+
+    *got_frame_ptr = 1;
+
+    return s->checksum_size;
+}
+
+AVCodec ff_qdm2_decoder =
+{
+    .name           = "qdm2",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_QDM2,
+    .priv_data_size = sizeof(QDM2Context),
+    .init           = qdm2_decode_init,
+    .close          = qdm2_decode_close,
+    .decode         = qdm2_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
+};