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ffmpeg/libavcodec/qdm2.c

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/*
* 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 "libavutil/channel_layout.h"
#include "libavutil/mem_internal.h"
#include "libavutil/thread.h"
#include "libavutil/tx.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "decode.h"
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "qdm2_tablegen.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 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)
} 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 FFTTone {
float level;
AVComplexFloat *complex;
const float *table;
int phase;
int phase_shift;
int duration;
short time_index;
short cutoff;
} FFTTone;
typedef struct FFTCoefficient {
int16_t sub_packet;
uint8_t channel;
int16_t offset;
int16_t exp;
uint8_t phase;
} FFTCoefficient;
typedef struct QDM2FFT {
DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1];
DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
* QDM2 decoder context
*/
typedef struct QDM2Context {
/// 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];
AVTXContext *rdft_ctx;
av_tx_fn rdft_fn;
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 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 int qdm2_get_vlc(GetBitContext *gb, const 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(const 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 checksummed
* @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);
// FIXME: this depends on bitreader-internal data
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
}
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 && list->packet) {
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 int fix_coding_method_array(int sb, int channels,
sb_int8_array coding_method)
{
int j, k;
int ch;
int run, case_val;
for (ch = 0; ch < channels; ch++) {
for (j = 0; j < 64; ) {
if (coding_method[ch][sb][j] < 8)
return -1;
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) {
int sbjk = sb + (j + k) / 64;
if (sbjk > 29) {
SAMPLES_NEEDED
continue;
}
if (coding_method[ch][sbjk][(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;
}
}
return 0;
}
/**
* 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];
}
}
}
}
}
}
/**
* 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];
}
}
/**
* 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;
int type34_first;
float type34_div = 0;
float type34_predictor;
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 0;
}
for (sb = sb_min; sb < sb_max; sb++) {
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);
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];
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;
}
channels = 1;
}
for (ch = 0; ch < channels; ch++) {
FIX_NOISE_IDX(q->noise_idx);
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) {
for (k = 0; k < run && j + k < 128; k++) {
q->sb_samples[0][j + k][sb] =
q->tone_level[0][sb][(j + k) / 2] * samples[k];
if (q->nb_channels == 2) {
if (sign_bits[(j + k) / 8])
q->sb_samples[1][j + k][sb] =
q->tone_level[1][sb][(j + k) / 2] * -samples[k];
else
q->sb_samples[1][j + k][sb] =
q->tone_level[1][sb][(j + k) / 2] * samples[k];
}
}
} 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;
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])
process_subpacket_9(q, nodes[0]);
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
if (nodes[1])
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] && nodes[1] && nodes[2])
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] && nodes[1] && nodes[3])
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) {
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 {
if (local_int_10 <= 2) {
av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
return;
}
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);
if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
return;
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)
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;
AVComplexFloat 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(AVComplexFloat));
}
/* 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;
AVComplexFloat 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;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
q->fft.complex[channel][q->fft_size].re = 0.0f;
q->fft.complex[channel][q->fft_size].im = 0.0f;
q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel],
sizeof(AVComplexFloat));
/* add samples to output buffer */
for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) {
out[0] += q->fft.temp[channel][i].re * gain;
out[q->channels] += q->fft.temp[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)
*/
static av_cold void qdm2_init_static_data(void) {
qdm2_init_vlc();
softclip_table_init();
rnd_table_init();
init_noise_samples();
ff_mpa_synth_init_float();
}
/**
* Init parameters from codec extradata
*/
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
{
static AVOnce init_static_once = AV_ONCE_INIT;
QDM2Context *s = avctx->priv_data;
int ret, tmp_val, tmp, size;
float scale = 1.0f / 2.0f;
GetByteContext gb;
/* 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 AVERROR_INVALIDDATA;
}
bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
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;
bytestream2_skip(&gb, 1);
}
if (bytestream2_get_bytes_left(&gb) < 12) {
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
bytestream2_get_bytes_left(&gb));
return AVERROR_INVALIDDATA;
}
bytestream2_skip(&gb, 8);
size = bytestream2_get_be32(&gb);
if (size > bytestream2_get_bytes_left(&gb)) {
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
bytestream2_get_bytes_left(&gb), size);
return AVERROR_INVALIDDATA;
}
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
return AVERROR_INVALIDDATA;
}
bytestream2_skip(&gb, 4);
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;
}
av_channel_layout_uninit(&avctx->ch_layout);
av_channel_layout_default(&avctx->ch_layout, s->channels);
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 || s->checksum_size <= 1) {
av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
return AVERROR_INVALIDDATA;
}
s->fft_order = av_log2(s->fft_size) + 1;
// Fail on unknown fft order
if ((s->fft_order < 7) || (s->fft_order > 9)) {
avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
return AVERROR_PATCHWELCOME;
}
// 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));
if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
avpriv_request_sample(avctx, "large frames");
return AVERROR_PATCHWELCOME;
}
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 (avctx->bit_rate <= 8000)
s->coeff_per_sb_select = 0;
else if (avctx->bit_rate < 16000)
s->coeff_per_sb_select = 1;
else
s->coeff_per_sb_select = 2;
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);
return AVERROR_INVALIDDATA;
}
ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0);
if (ret < 0)
return ret;
ff_mpadsp_init(&s->mpadsp);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
ff_thread_once(&init_static_once, qdm2_init_static_data);
return 0;
}
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
av_tx_uninit(&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) {
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 16-bit 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, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
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 ((ret = qdm2_decode(s, buf, out)) < 0)
return ret;
out += s->channels * s->frame_size;
}
*got_frame_ptr = 1;
return s->checksum_size;
}
const FFCodec ff_qdm2_decoder = {
.p.name = "qdm2",
CODEC_LONG_NAME("QDesign Music Codec 2"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_QDM2,
.priv_data_size = sizeof(QDM2Context),
.init = qdm2_decode_init,
.close = qdm2_decode_close,
FF_CODEC_DECODE_CB(qdm2_decode_frame),
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
};
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