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FFmpeg/libavcodec/qdmc.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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/*
* QDMC compatible decoder
* Copyright (c) 2017 Paul B Mahol
*
* 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
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#define BITSTREAM_READER_LE
#include "libavutil/channel_layout.h"
#include "libavutil/thread.h"
#include "avcodec.h"
#include "bytestream.h"
#include "get_bits.h"
#include "internal.h"
#include "fft.h"
typedef struct QDMCTone {
uint8_t mode;
uint8_t phase;
uint8_t offset;
int16_t freq;
int16_t amplitude;
} QDMCTone;
typedef struct QDMCContext {
AVCodecContext *avctx;
uint8_t frame_bits;
int band_index;
int frame_size;
int subframe_size;
int fft_offset;
int buffer_offset;
int nb_channels;
int checksum_size;
uint8_t noise[2][19][17];
QDMCTone tones[5][8192];
int nb_tones[5];
int cur_tone[5];
float alt_sin[5][31];
float fft_buffer[4][8192 * 2];
float noise2_buffer[4096 * 2];
float noise_buffer[4096 * 2];
float buffer[2 * 32768];
float *buffer_ptr;
int rndval;
DECLARE_ALIGNED(32, FFTComplex, cmplx)[2][512];
FFTContext fft_ctx;
} QDMCContext;
static float sin_table[512];
static VLC vtable[6];
static const unsigned code_prefix[] = {
0x0, 0x1, 0x2, 0x3, 0x4, 0x6, 0x8, 0xA,
0xC, 0x10, 0x14, 0x18, 0x1C, 0x24, 0x2C, 0x34,
0x3C, 0x4C, 0x5C, 0x6C, 0x7C, 0x9C, 0xBC, 0xDC,
0xFC, 0x13C, 0x17C, 0x1BC, 0x1FC, 0x27C, 0x2FC, 0x37C,
0x3FC, 0x4FC, 0x5FC, 0x6FC, 0x7FC, 0x9FC, 0xBFC, 0xDFC,
0xFFC, 0x13FC, 0x17FC, 0x1BFC, 0x1FFC, 0x27FC, 0x2FFC, 0x37FC,
0x3FFC, 0x4FFC, 0x5FFC, 0x6FFC, 0x7FFC, 0x9FFC, 0xBFFC, 0xDFFC,
0xFFFC, 0x13FFC, 0x17FFC, 0x1BFFC, 0x1FFFC, 0x27FFC, 0x2FFFC, 0x37FFC,
0x3FFFC
};
static const float amplitude_tab[64] = {
1.18750000f, 1.68359380f, 2.37500000f, 3.36718750f, 4.75000000f,
6.73437500f, 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f,
38.0000000f, 53.8750000f, 76.0000000f, 107.750000f, 152.000000f,
215.500000f, 304.000000f, 431.000000f, 608.000000f, 862.000000f,
1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f, 4864.00000f,
6896.00000f, 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f,
38912.0000f, 55168.0000f, 77824.0000f, 110336.000f, 155648.000f,
220672.000f, 311296.000f, 441344.000f, 622592.000f, 882688.000f,
1245184.00f, 1765376.00f, 2490368.00f, 3530752.00f, 4980736.00f,
7061504.00f, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};
static const uint16_t qdmc_nodes[112] = {
0, 1, 2, 4, 6, 8, 12, 16, 24, 32, 48, 56, 64,
80, 96, 120, 144, 176, 208, 240, 256,
0, 2, 4, 8, 16, 24, 32, 48, 56, 64, 80, 104,
128, 160, 208, 256, 0, 0, 0, 0, 0,
0, 2, 4, 8, 16, 32, 48, 64, 80, 112, 160, 208,
256, 0, 0, 0, 0, 0, 0, 0, 0,
0, 4, 8, 16, 32, 48, 64, 96, 144, 208, 256,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 4, 16, 32, 64, 256, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
};
static const uint8_t noise_bands_size[] = {
19, 14, 11, 9, 4, 2, 0
};
static const uint8_t noise_bands_selector[] = {
4, 3, 2, 1, 0, 0, 0,
};
static const uint8_t qdmc_hufftab[][2] = {
/* Noise value - 27 entries */
{ 1, 2 }, { 10, 7 }, { 26, 9 }, { 22, 9 }, { 24, 9 }, { 14, 9 },
{ 8, 6 }, { 6, 5 }, { 7, 5 }, { 9, 7 }, { 30, 9 }, { 32, 10 },
{ 13, 10 }, { 20, 9 }, { 28, 9 }, { 12, 7 }, { 15, 11 }, { 36, 12 },
{ 0, 12 }, { 34, 10 }, { 18, 9 }, { 11, 9 }, { 16, 9 }, { 5, 3 },
{ 2, 3 }, { 4, 3 }, { 3, 2 },
/* Noise segment length - 12 entries */
{ 1, 1 }, { 2, 2 }, { 3, 4 }, { 8, 9 }, { 9, 10 }, { 0, 10 },
{ 13, 8 }, { 7, 7 }, { 6, 6 }, { 17, 5 }, { 4, 4 }, { 5, 4 },
/* Amplitude - 28 entries */
{ 18, 3 }, { 16, 3 }, { 22, 7 }, { 8, 10 }, { 4, 10 }, { 3, 9 },
{ 2, 8 }, { 23, 8 }, { 10, 8 }, { 11, 7 }, { 21, 5 }, { 20, 4 },
{ 1, 7 }, { 7, 10 }, { 5, 10 }, { 9, 9 }, { 6, 10 }, { 25, 11 },
{ 26, 12 }, { 27, 13 }, { 0, 13 }, { 24, 9 }, { 12, 6 }, { 13, 5 },
{ 14, 4 }, { 19, 3 }, { 15, 3 }, { 17, 2 },
/* Frequency differences - 47 entries */
{ 2, 4 }, { 14, 6 }, { 26, 7 }, { 31, 8 }, { 32, 9 }, { 35, 9 },
{ 7, 5 }, { 10, 5 }, { 22, 7 }, { 27, 7 }, { 19, 7 }, { 20, 7 },
{ 4, 5 }, { 13, 5 }, { 17, 6 }, { 15, 6 }, { 8, 5 }, { 5, 4 },
{ 28, 7 }, { 33, 9 }, { 36, 11 }, { 38, 12 }, { 42, 14 }, { 45, 16 },
{ 44, 18 }, { 0, 18 }, { 46, 17 }, { 43, 15 }, { 40, 13 }, { 37, 11 },
{ 39, 12 }, { 41, 12 }, { 34, 8 }, { 16, 6 }, { 11, 5 }, { 9, 4 },
{ 1, 2 }, { 3, 4 }, { 30, 7 }, { 29, 7 }, { 23, 6 }, { 24, 6 },
{ 18, 6 }, { 6, 4 }, { 12, 5 }, { 21, 6 }, { 25, 6 },
/* Amplitude differences - 9 entries */
{ 1, 2 }, { 3, 3 }, { 4, 4 }, { 5, 5 }, { 6, 6 }, { 7, 7 },
{ 8, 8 }, { 0, 8 }, { 2, 1 },
/* Phase differences - 9 entries */
{ 2, 2 }, { 1, 2 }, { 3, 4 }, { 7, 4 }, { 6, 5 }, { 5, 6 },
{ 0, 6 }, { 4, 4 }, { 8, 2 },
};
static const uint8_t huff_sizes[] = {
27, 12, 28, 47, 9, 9
};
static const uint8_t huff_bits[] = {
12, 10, 12, 12, 8, 6
};
static av_cold void qdmc_init_static_data(void)
{
const uint8_t (*hufftab)[2] = qdmc_hufftab;
int i;
for (unsigned i = 0, offset = 0; i < FF_ARRAY_ELEMS(vtable); i++) {
static VLC_TYPE vlc_buffer[13698][2];
vtable[i].table = &vlc_buffer[offset];
vtable[i].table_allocated = FF_ARRAY_ELEMS(vlc_buffer) - offset;
ff_init_vlc_from_lengths(&vtable[i], huff_bits[i], huff_sizes[i],
&hufftab[0][1], 2, &hufftab[0][0], 2, 1, -1,
INIT_VLC_LE | INIT_VLC_STATIC_OVERLONG, NULL);
hufftab += huff_sizes[i];
offset += vtable[i].table_size;
}
for (i = 0; i < 512; i++)
sin_table[i] = sin(2.0f * i * M_PI * 0.001953125f);
}
static void make_noises(QDMCContext *s)
{
int i, j, n0, n1, n2, diff;
float *nptr;
for (j = 0; j < noise_bands_size[s->band_index]; j++) {
n0 = qdmc_nodes[j + 21 * s->band_index ];
n1 = qdmc_nodes[j + 21 * s->band_index + 1];
n2 = qdmc_nodes[j + 21 * s->band_index + 2];
nptr = s->noise_buffer + 256 * j;
for (i = 0; i + n0 < n1; i++, nptr++)
nptr[0] = i / (float)(n1 - n0);
diff = n2 - n1;
nptr = s->noise_buffer + (j << 8) + n1 - n0;
for (i = n1; i < n2; i++, nptr++, diff--)
nptr[0] = diff / (float)(n2 - n1);
}
}
static av_cold int qdmc_decode_init(AVCodecContext *avctx)
{
static AVOnce init_static_once = AV_ONCE_INIT;
QDMCContext *s = avctx->priv_data;
int ret, fft_size, fft_order, size, g, j, x;
GetByteContext b;
ff_thread_once(&init_static_once, qdmc_init_static_data);
if (!avctx->extradata || (avctx->extradata_size < 48)) {
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
return AVERROR_INVALIDDATA;
}
bytestream2_init(&b, avctx->extradata, avctx->extradata_size);
while (bytestream2_get_bytes_left(&b) > 8) {
if (bytestream2_peek_be64(&b) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
(uint64_t)MKBETAG('Q','D','M','C')))
break;
bytestream2_skipu(&b, 1);
}
bytestream2_skipu(&b, 8);
if (bytestream2_get_bytes_left(&b) < 36) {
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
bytestream2_get_bytes_left(&b));
return AVERROR_INVALIDDATA;
}
size = bytestream2_get_be32u(&b);
if (size > bytestream2_get_bytes_left(&b)) {
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
bytestream2_get_bytes_left(&b), size);
return AVERROR_INVALIDDATA;
}
if (bytestream2_get_be32u(&b) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
return AVERROR_INVALIDDATA;
}
bytestream2_skipu(&b, 4);
avctx->channels = s->nb_channels = bytestream2_get_be32u(&b);
if (s->nb_channels <= 0 || s->nb_channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR_INVALIDDATA;
}
avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
AV_CH_LAYOUT_MONO;
avctx->sample_rate = bytestream2_get_be32u(&b);
avctx->bit_rate = bytestream2_get_be32u(&b);
bytestream2_skipu(&b, 4);
fft_size = bytestream2_get_be32u(&b);
fft_order = av_log2(fft_size) + 1;
s->checksum_size = bytestream2_get_be32u(&b);
if (s->checksum_size >= 1U << 28) {
av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
return AVERROR_INVALIDDATA;
}
if (avctx->sample_rate >= 32000) {
x = 28000;
s->frame_bits = 13;
} else if (avctx->sample_rate >= 16000) {
x = 20000;
s->frame_bits = 12;
} else {
x = 16000;
s->frame_bits = 11;
}
s->frame_size = 1 << s->frame_bits;
s->subframe_size = s->frame_size >> 5;
if (avctx->channels == 2)
x = 3 * x / 2;
s->band_index = noise_bands_selector[FFMIN(6, llrint(floor(avctx->bit_rate * 3.0 / (double)x + 0.5)))];
if ((fft_order < 7) || (fft_order > 9)) {
avpriv_request_sample(avctx, "Unknown FFT order %d", fft_order);
return AVERROR_PATCHWELCOME;
}
if (fft_size != (1 << (fft_order - 1))) {
av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", fft_size);
return AVERROR_INVALIDDATA;
}
ret = ff_fft_init(&s->fft_ctx, fft_order, 1);
if (ret < 0)
return ret;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
for (g = 5; g > 0; g--) {
for (j = 0; j < (1 << g) - 1; j++)
s->alt_sin[5-g][j] = sin_table[(((j+1) << (8 - g)) & 0x1FF)];
}
make_noises(s);
return 0;
}
static av_cold int qdmc_decode_close(AVCodecContext *avctx)
{
QDMCContext *s = avctx->priv_data;
ff_fft_end(&s->fft_ctx);
return 0;
}
static int qdmc_get_vlc(GetBitContext *gb, VLC *table, int flag)
{
int v;
if (get_bits_left(gb) < 1)
return AVERROR_INVALIDDATA;
v = get_vlc2(gb, table->table, table->bits, 2);
if (v < 0)
v = get_bits(gb, get_bits(gb, 3) + 1);
if (flag) {
if (v >= FF_ARRAY_ELEMS(code_prefix))
return AVERROR_INVALIDDATA;
v = code_prefix[v] + get_bitsz(gb, v >> 2);
}
return v;
}
static int skip_label(QDMCContext *s, GetBitContext *gb)
{
uint32_t label = get_bits_long(gb, 32);
uint16_t sum = 226, checksum = get_bits(gb, 16);
const uint8_t *ptr = gb->buffer + 6;
int i;
if (label != MKTAG('Q', 'M', 'C', 1))
return AVERROR_INVALIDDATA;
for (i = 0; i < s->checksum_size - 6; i++)
sum += ptr[i];
return sum != checksum;
}
static int read_noise_data(QDMCContext *s, GetBitContext *gb)
{
int ch, j, k, v, idx, band, lastval, newval, len;
for (ch = 0; ch < s->nb_channels; ch++) {
for (band = 0; band < noise_bands_size[s->band_index]; band++) {
v = qdmc_get_vlc(gb, &vtable[0], 0);
if (v < 0)
return AVERROR_INVALIDDATA;
if (v & 1)
v = v + 1;
else
v = -v;
lastval = v / 2;
s->noise[ch][band][0] = lastval - 1;
for (j = 0; j < 15;) {
len = qdmc_get_vlc(gb, &vtable[1], 1);
if (len < 0)
return AVERROR_INVALIDDATA;
len += 1;
v = qdmc_get_vlc(gb, &vtable[0], 0);
if (v < 0)
return AVERROR_INVALIDDATA;
if (v & 1)
newval = lastval + (v + 1) / 2;
else
newval = lastval - v / 2;
idx = j + 1;
if (len + idx > 16)
return AVERROR_INVALIDDATA;
for (k = 1; idx <= j + len; k++, idx++)
s->noise[ch][band][idx] = lastval + k * (newval - lastval) / len - 1;
lastval = newval;
j += len;
}
}
}
return 0;
}
static void add_tone(QDMCContext *s, int group, int offset, int freq, int stereo_mode, int amplitude, int phase)
{
const int index = s->nb_tones[group];
if (index >= FF_ARRAY_ELEMS(s->tones[group])) {
av_log(s->avctx, AV_LOG_WARNING, "Too many tones already in buffer, ignoring tone!\n");
return;
}
s->tones[group][index].offset = offset;
s->tones[group][index].freq = freq;
s->tones[group][index].mode = stereo_mode;
s->tones[group][index].amplitude = amplitude;
s->tones[group][index].phase = phase;
s->nb_tones[group]++;
}
static int read_wave_data(QDMCContext *s, GetBitContext *gb)
{
int amp, phase, stereo_mode = 0, i, group, freq, group_size, group_bits;
int amp2, phase2, pos2, off;
for (group = 0; group < 5; group++) {
group_size = 1 << (s->frame_bits - group - 1);
group_bits = 4 - group;
pos2 = 0;
off = 0;
for (i = 1; ; i = freq + 1) {
int v;
v = qdmc_get_vlc(gb, &vtable[3], 1);
if (v < 0)
return AVERROR_INVALIDDATA;
freq = i + v;
while (freq >= group_size - 1) {
freq += 2 - group_size;
pos2 += group_size;
off += 1 << group_bits;
}
if (pos2 >= s->frame_size)
break;
if (s->nb_channels > 1)
stereo_mode = get_bits(gb, 2);
amp = qdmc_get_vlc(gb, &vtable[2], 0);
if (amp < 0)
return AVERROR_INVALIDDATA;
phase = get_bits(gb, 3);
if (stereo_mode > 1) {
amp2 = qdmc_get_vlc(gb, &vtable[4], 0);
if (amp2 < 0)
return AVERROR_INVALIDDATA;
amp2 = amp - amp2;
phase2 = qdmc_get_vlc(gb, &vtable[5], 0);
if (phase2 < 0)
return AVERROR_INVALIDDATA;
phase2 = phase - phase2;
if (phase2 < 0)
phase2 += 8;
}
if ((freq >> group_bits) + 1 < s->subframe_size) {
add_tone(s, group, off, freq, stereo_mode & 1, amp, phase);
if (stereo_mode > 1)
add_tone(s, group, off, freq, ~stereo_mode & 1, amp2, phase2);
}
}
}
return 0;
}
static void lin_calc(QDMCContext *s, float amplitude, int node1, int node2, int index)
{
int subframe_size, i, j, k, length;
float scale, *noise_ptr;
scale = 0.5 * amplitude;
subframe_size = s->subframe_size;
if (subframe_size >= node2)
subframe_size = node2;
length = (subframe_size - node1) & 0xFFFC;
j = node1;
noise_ptr = &s->noise_buffer[256 * index];
for (i = 0; i < length; i += 4, j+= 4, noise_ptr += 4) {
s->noise2_buffer[j ] += scale * noise_ptr[0];
s->noise2_buffer[j + 1] += scale * noise_ptr[1];
s->noise2_buffer[j + 2] += scale * noise_ptr[2];
s->noise2_buffer[j + 3] += scale * noise_ptr[3];
}
k = length + node1;
noise_ptr = s->noise_buffer + length + (index << 8);
for (i = length; i < subframe_size - node1; i++, k++, noise_ptr++)
s->noise2_buffer[k] += scale * noise_ptr[0];
}
static void add_noise(QDMCContext *s, int ch, int current_subframe)
{
int i, j, aindex;
float amplitude;
float *im = &s->fft_buffer[0 + ch][s->fft_offset + s->subframe_size * current_subframe];
float *re = &s->fft_buffer[2 + ch][s->fft_offset + s->subframe_size * current_subframe];
memset(s->noise2_buffer, 0, 4 * s->subframe_size);
for (i = 0; i < noise_bands_size[s->band_index]; i++) {
if (qdmc_nodes[i + 21 * s->band_index] > s->subframe_size - 1)
break;
aindex = s->noise[ch][i][current_subframe / 2];
amplitude = aindex > 0 ? amplitude_tab[aindex & 0x3F] : 0.0f;
lin_calc(s, amplitude, qdmc_nodes[21 * s->band_index + i],
qdmc_nodes[21 * s->band_index + i + 2], i);
}
for (j = 2; j < s->subframe_size - 1; j++) {
float rnd_re, rnd_im;
s->rndval = 214013U * s->rndval + 2531011;
rnd_im = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];
s->rndval = 214013U * s->rndval + 2531011;
rnd_re = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];
im[j ] += rnd_im;
re[j ] += rnd_re;
im[j+1] -= rnd_im;
re[j+1] -= rnd_re;
}
}
static void add_wave(QDMCContext *s, int offset, int freqs, int group, int stereo_mode, int amp, int phase)
{
int j, group_bits, pos, pindex;
float im, re, amplitude, level, *imptr, *reptr;
if (s->nb_channels == 1)
stereo_mode = 0;
group_bits = 4 - group;
pos = freqs >> (4 - group);
amplitude = amplitude_tab[amp & 0x3F];
imptr = &s->fft_buffer[ stereo_mode][s->fft_offset + s->subframe_size * offset + pos];
reptr = &s->fft_buffer[2 + stereo_mode][s->fft_offset + s->subframe_size * offset + pos];
pindex = (phase << 6) - ((2 * (freqs >> (4 - group)) + 1) << 7);
for (j = 0; j < (1 << (group_bits + 1)) - 1; j++) {
pindex += (2 * freqs + 1) << (7 - group_bits);
level = amplitude * s->alt_sin[group][j];
im = level * sin_table[ pindex & 0x1FF];
re = level * sin_table[(pindex + 128) & 0x1FF];
imptr[0] += im;
imptr[1] -= im;
reptr[0] += re;
reptr[1] -= re;
imptr += s->subframe_size;
reptr += s->subframe_size;
if (imptr >= &s->fft_buffer[stereo_mode][2 * s->frame_size]) {
imptr = &s->fft_buffer[0 + stereo_mode][pos];
reptr = &s->fft_buffer[2 + stereo_mode][pos];
}
}
}
static void add_wave0(QDMCContext *s, int offset, int freqs, int stereo_mode, int amp, int phase)
{
float level, im, re;
int pos;
if (s->nb_channels == 1)
stereo_mode = 0;
level = amplitude_tab[amp & 0x3F];
im = level * sin_table[ (phase << 6) & 0x1FF];
re = level * sin_table[((phase << 6) + 128) & 0x1FF];
pos = s->fft_offset + freqs + s->subframe_size * offset;
s->fft_buffer[ stereo_mode][pos ] += im;
s->fft_buffer[2 + stereo_mode][pos ] += re;
s->fft_buffer[ stereo_mode][pos + 1] -= im;
s->fft_buffer[2 + stereo_mode][pos + 1] -= re;
}
static void add_waves(QDMCContext *s, int current_subframe)
{
int w, g;
for (g = 0; g < 4; g++) {
for (w = s->cur_tone[g]; w < s->nb_tones[g]; w++) {
QDMCTone *t = &s->tones[g][w];
if (current_subframe < t->offset)
break;
add_wave(s, t->offset, t->freq, g, t->mode, t->amplitude, t->phase);
}
s->cur_tone[g] = w;
}
for (w = s->cur_tone[4]; w < s->nb_tones[4]; w++) {
QDMCTone *t = &s->tones[4][w];
if (current_subframe < t->offset)
break;
add_wave0(s, t->offset, t->freq, t->mode, t->amplitude, t->phase);
}
s->cur_tone[4] = w;
}
static int decode_frame(QDMCContext *s, GetBitContext *gb, int16_t *out)
{
int ret, ch, i, n;
if (skip_label(s, gb))
return AVERROR_INVALIDDATA;
s->fft_offset = s->frame_size - s->fft_offset;
s->buffer_ptr = &s->buffer[s->nb_channels * s->buffer_offset];
ret = read_noise_data(s, gb);
if (ret < 0)
return ret;
ret = read_wave_data(s, gb);
if (ret < 0)
return ret;
for (n = 0; n < 32; n++) {
float *r;
for (ch = 0; ch < s->nb_channels; ch++)
add_noise(s, ch, n);
add_waves(s, n);
for (ch = 0; ch < s->nb_channels; ch++) {
for (i = 0; i < s->subframe_size; i++) {
s->cmplx[ch][i].re = s->fft_buffer[ch + 2][s->fft_offset + n * s->subframe_size + i];
s->cmplx[ch][i].im = s->fft_buffer[ch + 0][s->fft_offset + n * s->subframe_size + i];
s->cmplx[ch][s->subframe_size + i].re = 0;
s->cmplx[ch][s->subframe_size + i].im = 0;
}
}
for (ch = 0; ch < s->nb_channels; ch++) {
s->fft_ctx.fft_permute(&s->fft_ctx, s->cmplx[ch]);
s->fft_ctx.fft_calc(&s->fft_ctx, s->cmplx[ch]);
}
r = &s->buffer_ptr[s->nb_channels * n * s->subframe_size];
for (i = 0; i < 2 * s->subframe_size; i++) {
for (ch = 0; ch < s->nb_channels; ch++) {
*r++ += s->cmplx[ch][i].re;
}
}
r = &s->buffer_ptr[n * s->subframe_size * s->nb_channels];
for (i = 0; i < s->nb_channels * s->subframe_size; i++) {
out[i] = av_clipf(r[i], INT16_MIN, INT16_MAX);
}
out += s->subframe_size * s->nb_channels;
for (ch = 0; ch < s->nb_channels; ch++) {
memset(s->fft_buffer[ch+0] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);
memset(s->fft_buffer[ch+2] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);
}
memset(s->buffer + s->nb_channels * (n * s->subframe_size + s->frame_size + s->buffer_offset), 0, 4 * s->subframe_size * s->nb_channels);
}
s->buffer_offset += s->frame_size;
if (s->buffer_offset >= 32768 - s->frame_size) {
memcpy(s->buffer, &s->buffer[s->nb_channels * s->buffer_offset], 4 * s->frame_size * s->nb_channels);
s->buffer_offset = 0;
}
return 0;
}
static av_cold void qdmc_flush(AVCodecContext *avctx)
{
QDMCContext *s = avctx->priv_data;
memset(s->buffer, 0, sizeof(s->buffer));
memset(s->fft_buffer, 0, sizeof(s->fft_buffer));
s->fft_offset = 0;
s->buffer_offset = 0;
}
static int qdmc_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
QDMCContext *s = avctx->priv_data;
AVFrame *frame = data;
GetBitContext gb;
int ret;
if (!avpkt->data)
return 0;
if (avpkt->size < s->checksum_size)
return AVERROR_INVALIDDATA;
s->avctx = avctx;
frame->nb_samples = s->frame_size;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
if ((ret = init_get_bits8(&gb, avpkt->data, s->checksum_size)) < 0)
return ret;
memset(s->nb_tones, 0, sizeof(s->nb_tones));
memset(s->cur_tone, 0, sizeof(s->cur_tone));
ret = decode_frame(s, &gb, (int16_t *)frame->data[0]);
if (ret >= 0) {
*got_frame_ptr = 1;
return s->checksum_size;
}
qdmc_flush(avctx);
return ret;
}
const AVCodec ff_qdmc_decoder = {
.name = "qdmc",
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 1"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_QDMC,
.priv_data_size = sizeof(QDMCContext),
.init = qdmc_decode_init,
.close = qdmc_decode_close,
.decode = qdmc_decode_frame,
.flush = qdmc_flush,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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