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FFmpeg/libavcodec/qdm2.c
Andreas Rheinhardt 20f9727018 avcodec/codec_internal: Add FFCodec, hide internal part of AVCodec 
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.

This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-03-21 01:33:09 +01:00

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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 <stdio.h>
#include "libavutil/channel_layout.h"
#include "libavutil/mem_internal.h"
#include "libavutil/thread.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "rdft.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 QDM2Complex {
float re;
float im;
} QDM2Complex;
typedef struct FFTTone {
float level;
QDM2Complex *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, QDM2Complex, complex)[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];
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 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;
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)
*/
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 tmp_val, tmp, size;
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;
}
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
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;
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) {
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, 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 ((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",
.p.long_name = NULL_IF_CONFIG_SMALL("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,
.decode = qdm2_decode_frame,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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