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ffmpeg.wasm-core
Commit 90ac9f40 authored by Paul B Mahol's avatar Paul B Mahol
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avcodec: add QDMC decoder 

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent 49633f9f
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Showing with 789 additions and 2 deletions
Changelog
View file @ 90ac9f40
...@@ -12,6 +12,7 @@ version <next>: ...@@ -12,6 +12,7 @@ version <next>:
- 16.8 floating point pcm decoder - 16.8 floating point pcm decoder
- 24.0 floating point pcm decoder - 24.0 floating point pcm decoder
- Apple Pixlet decoder - Apple Pixlet decoder
- QDMC audio decoder
version 3.2: version 3.2:
- libopenmpt demuxer - libopenmpt demuxer
......
doc/general.texi
View file @ 90ac9f40
...@@ -1048,6 +1048,7 @@ following image formats are supported: ...@@ -1048,6 +1048,7 @@ following image formats are supported:
@item PCM unsigned 32-bit little-endian @tab X @tab X @item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab @tab X @item PCM Zork @tab @tab X
@item QCELP / PureVoice @tab @tab X @item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 1 @tab @tab X
@item QDesign Music Codec 2 @tab @tab X @item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions. @tab There are still some distortions.
@item RealAudio 1.0 (14.4K) @tab X @tab X @item RealAudio 1.0 (14.4K) @tab X @tab X
......
libavcodec/Makefile
View file @ 90ac9f40
...@@ -466,6 +466,7 @@ OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o \ ...@@ -466,6 +466,7 @@ OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o \
celp_filters.o acelp_vectors.o \ celp_filters.o acelp_vectors.o \
acelp_filters.o acelp_filters.o
OBJS-$(CONFIG_QDM2_DECODER) += qdm2.o OBJS-$(CONFIG_QDM2_DECODER) += qdm2.o
OBJS-$(CONFIG_QDMC_DECODER) += qdmc.o
OBJS-$(CONFIG_QDRAW_DECODER) += qdrw.o OBJS-$(CONFIG_QDRAW_DECODER) += qdrw.o
OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o
OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o
......
libavcodec/allcodecs.c
View file @ 90ac9f40
...@@ -449,6 +449,7 @@ void avcodec_register_all(void) ...@@ -449,6 +449,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(PAF_AUDIO, paf_audio); REGISTER_DECODER(PAF_AUDIO, paf_audio);
REGISTER_DECODER(QCELP, qcelp); REGISTER_DECODER(QCELP, qcelp);
REGISTER_DECODER(QDM2, qdm2); REGISTER_DECODER(QDM2, qdm2);
REGISTER_DECODER(QDMC, qdmc);
REGISTER_ENCDEC (RA_144, ra_144); REGISTER_ENCDEC (RA_144, ra_144);
REGISTER_DECODER(RA_288, ra_288); REGISTER_DECODER(RA_288, ra_288);
REGISTER_DECODER(RALF, ralf); REGISTER_DECODER(RALF, ralf);
......
libavcodec/qdmc.c 0 → 100644
View file @ 90ac9f40
/*
* 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 "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][16];
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 noise_value_bits[] = {
12, 7, 9, 7, 10, 9, 11, 9, 9, 2, 9, 9, 9, 9,
9, 3, 9, 10, 10, 12, 2, 3, 3, 5, 5, 6, 7,
};
static const uint8_t noise_value_symbols[] = {
0, 10, 11, 12, 13, 14, 15, 16, 18, 1, 20, 22, 24,
26, 28, 2, 30, 32, 34, 36, 3, 4, 5, 6, 7, 8, 9,
};
static const uint16_t noise_value_codes[] = {
0xC7A, 0x002, 0x0FA, 0x03A, 0x35A, 0x1C2, 0x07A, 0x1FA,
0x17A, 0x000, 0x0DA, 0x142, 0x0C2, 0x042, 0x1DA, 0x001,
0x05A, 0x15A, 0x27A, 0x47A, 0x003, 0x005, 0x006, 0x012,
0x00A, 0x022, 0x01A,
};
static const uint8_t noise_segment_length_bits[] = {
10, 8, 5, 1, 2, 4, 4, 4, 6, 7, 9, 10,
};
static const uint8_t noise_segment_length_symbols[] = {
0, 13, 17, 1, 2, 3, 4, 5, 6, 7, 8, 9,
};
static const uint16_t noise_segment_length_codes[] = {
0x30B, 0x8B, 0x1B, 0x0, 0x1, 0x3, 0x7, 0xF, 0x2b, 0x4B, 0xB, 0x10B,
};
static const uint8_t freq_diff_bits[] = {
18, 2, 4, 4, 5, 4, 4, 5, 5, 4, 5, 5, 5, 5, 6, 6, 6, 6, 6, 7, 7, 6,
7, 6, 6, 6, 7, 7, 7, 7, 7, 8, 9, 9, 8, 9, 11, 11, 12, 12, 13, 12,
14, 15, 18, 16, 17,
};
static const uint32_t freq_diff_codes[] = {
0x2AD46, 0x1, 0x0, 0x3, 0xC, 0xA, 0x7, 0x18, 0x12, 0xE, 0x4, 0x16,
0xF, 0x1C, 0x8, 0x22, 0x26, 0x2, 0x3B, 0x34, 0x74, 0x1F, 0x14, 0x2B,
0x1B, 0x3F, 0x28, 0x54, 0x6, 0x4B, 0xB, 0x68, 0xE8, 0x46, 0xC6, 0x1E8,
0x146, 0x346, 0x546, 0x746, 0x1D46, 0xF46, 0xD46, 0x6D46, 0xAD46, 0x2D46,
0x1AD46,
};
static const uint8_t amplitude_bits[] = {
13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6,
5, 4, 3, 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13,
};
static const uint16_t amplitude_codes[] = {
0x1EC6, 0x6, 0xC2, 0x142, 0x242, 0x246, 0xC6, 0x46, 0x42, 0x146, 0xA2,
0x62, 0x26, 0x16, 0xE, 0x5, 0x4, 0x3, 0x0, 0x1, 0xA, 0x12, 0x2, 0x22,
0x1C6, 0x2C6, 0x6C6, 0xEC6,
};
static const uint8_t amplitude_diff_bits[] = {
8, 2, 1, 3, 4, 5, 6, 7, 8,
};
static const uint8_t amplitude_diff_codes[] = {
0xFE, 0x0, 0x1, 0x2, 0x6, 0xE, 0x1E, 0x3E, 0x7E,
};
static const uint8_t phase_diff_bits[] = {
6, 2, 2, 4, 4, 6, 5, 4, 2,
};
static const uint8_t phase_diff_codes[] = {
0x35, 0x2, 0x0, 0x1, 0xD, 0x15, 0x5, 0x9, 0x3,
};
#define INIT_VLC_STATIC_LE(vlc, nb_bits, nb_codes, \
bits, bits_wrap, bits_size, \
codes, codes_wrap, codes_size, \
symbols, symbols_wrap, symbols_size, \
static_size) \
do { \
static VLC_TYPE table[static_size][2]; \
(vlc)->table = table; \
(vlc)->table_allocated = static_size; \
ff_init_vlc_sparse(vlc, nb_bits, nb_codes, \
bits, bits_wrap, bits_size, \
codes, codes_wrap, codes_size, \
symbols, symbols_wrap, symbols_size, \
INIT_VLC_LE | INIT_VLC_USE_NEW_STATIC); \
} while (0)
static av_cold void qdmc_init_static_data(AVCodec *codec)
{
int i;
INIT_VLC_STATIC_LE(&vtable[0], 12, FF_ARRAY_ELEMS(noise_value_bits),
noise_value_bits, 1, 1, noise_value_codes, 2, 2, noise_value_symbols, 1, 1, 4096);
INIT_VLC_STATIC_LE(&vtable[1], 10, FF_ARRAY_ELEMS(noise_segment_length_bits),
noise_segment_length_bits, 1, 1, noise_segment_length_codes, 2, 2,
noise_segment_length_symbols, 1, 1, 1024);
INIT_VLC_STATIC_LE(&vtable[2], 13, FF_ARRAY_ELEMS(amplitude_bits),
amplitude_bits, 1, 1, amplitude_codes, 2, 2, NULL, 0, 0, 8192);
INIT_VLC_STATIC_LE(&vtable[3], 18, FF_ARRAY_ELEMS(freq_diff_bits),
freq_diff_bits, 1, 1, freq_diff_codes, 4, 4, NULL, 0, 0, 262144);
INIT_VLC_STATIC_LE(&vtable[4], 8, FF_ARRAY_ELEMS(amplitude_diff_bits),
amplitude_diff_bits, 1, 1, amplitude_diff_codes, 1, 1, NULL, 0, 0, 256);
INIT_VLC_STATIC_LE(&vtable[5], 6, FF_ARRAY_ELEMS(phase_diff_bits),
phase_diff_bits, 1, 1, phase_diff_codes, 1, 1, NULL, 0, 0, 64);
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)
{
QDMCContext *s = avctx->priv_data;
int fft_size, fft_order, size, g, j, x;
GetByteContext b;
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;
}
ff_fft_init(&s->fft_ctx, fft_order, 1);
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;
v = get_vlc2(gb, table->table, table->bits, 1);
if (v < 0)
return AVERROR_INVALIDDATA;
if (v)
v = v - 1;
else
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 = 214013 * s->rndval + 2531011;
rnd_im = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];
s->rndval = 214013 * 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;
}
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,
.init_static_data = qdmc_init_static_data,
.close = qdmc_decode_close,
.decode = qdmc_decode_frame,
.flush = qdmc_flush,
.capabilities = AV_CODEC_CAP_DR1,
};
libavcodec/version.h
View file @ 90ac9f40
...@@ -28,8 +28,8 @@ ...@@ -28,8 +28,8 @@
#include "libavutil/version.h" #include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 57 #define LIBAVCODEC_VERSION_MAJOR 57
#define LIBAVCODEC_VERSION_MINOR 70 #define LIBAVCODEC_VERSION_MINOR 71
#define LIBAVCODEC_VERSION_MICRO 101 #define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \ LIBAVCODEC_VERSION_MINOR, \
......
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