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e4eeb851fd
FFmpeg/libavcodec/atrac3.c
Andreas Rheinhardt e4eeb851fd avcodec/atrac3: Don't use too big VLC tables 
The longest code of any of the VLC tables used is eight bits long, so
using nine bits long VLC tables is wasteful. Furthermore, there are only
seven VLC tables used, yet the code up until now made it look like there
should be eight. This has been corrected, too.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-10-27 10:19:38 +01:00

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/*
* ATRAC3 compatible decoder
* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 Benjamin Larsson
*
* 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
* ATRAC3 compatible decoder.
* This decoder handles Sony's ATRAC3 data.
*
* Container formats used to store ATRAC3 data:
* RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
*
* To use this decoder, a calling application must supply the extradata
* bytes provided in the containers above.
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "libavutil/attributes.h"
#include "libavutil/float_dsp.h"
#include "libavutil/libm.h"
#include "avcodec.h"
#include "bytestream.h"
#include "fft.h"
#include "get_bits.h"
#include "internal.h"
#include "atrac.h"
#include "atrac3data.h"
#define MIN_CHANNELS 1
#define MAX_CHANNELS 8
#define MAX_JS_PAIRS 8 / 2
#define JOINT_STEREO 0x12
#define SINGLE 0x2
#define SAMPLES_PER_FRAME 1024
#define MDCT_SIZE 512
#define ATRAC3_VLC_BITS 8
typedef struct GainBlock {
AtracGainInfo g_block[4];
} GainBlock;
typedef struct TonalComponent {
int pos;
int num_coefs;
float coef[8];
} TonalComponent;
typedef struct ChannelUnit {
int bands_coded;
int num_components;
float prev_frame[SAMPLES_PER_FRAME];
int gc_blk_switch;
TonalComponent components[64];
GainBlock gain_block[2];
DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
float delay_buf1[46]; ///<qmf delay buffers
float delay_buf2[46];
float delay_buf3[46];
} ChannelUnit;
typedef struct ATRAC3Context {
GetBitContext gb;
//@{
/** stream data */
int coding_mode;
ChannelUnit *units;
//@}
//@{
/** joint-stereo related variables */
int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
int matrix_coeff_index_now[MAX_JS_PAIRS][4];
int matrix_coeff_index_next[MAX_JS_PAIRS][4];
int weighting_delay[MAX_JS_PAIRS][6];
//@}
//@{
/** data buffers */
uint8_t *decoded_bytes_buffer;
float temp_buf[1070];
//@}
//@{
/** extradata */
int scrambled_stream;
//@}
AtracGCContext gainc_ctx;
FFTContext mdct_ctx;
void (*vector_fmul)(float *dst, const float *src0, const float *src1,
int len);
} ATRAC3Context;
static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC_TYPE atrac3_vlc_table[7 * 1 << ATRAC3_VLC_BITS][2];
static VLC spectral_coeff_tab[7];
/**
* Regular 512 points IMDCT without overlapping, with the exception of the
* swapping of odd bands caused by the reverse spectra of the QMF.
*
* @param odd_band 1 if the band is an odd band
*/
static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
{
int i;
if (odd_band) {
/**
* Reverse the odd bands before IMDCT, this is an effect of the QMF
* transform or it gives better compression to do it this way.
* FIXME: It should be possible to handle this in imdct_calc
* for that to happen a modification of the prerotation step of
* all SIMD code and C code is needed.
* Or fix the functions before so they generate a pre reversed spectrum.
*/
for (i = 0; i < 128; i++)
FFSWAP(float, input[i], input[255 - i]);
}
q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
/* Perform windowing on the output. */
q->vector_fmul(output, output, mdct_window, MDCT_SIZE);
}
/*
* indata descrambling, only used for data coming from the rm container
*/
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
{
int i, off;
uint32_t c;
const uint32_t *buf;
uint32_t *output = (uint32_t *)out;
off = (intptr_t)input & 3;
buf = (const uint32_t *)(input - off);
if (off)
c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
else
c = av_be2ne32(0x537F6103U);
bytes += 3 + off;
for (i = 0; i < bytes / 4; i++)
output[i] = c ^ buf[i];
if (off)
avpriv_request_sample(NULL, "Offset of %d", off);
return off;
}
static av_cold void init_imdct_window(void)
{
int i, j;
/* generate the mdct window, for details see
* http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
for (i = 0, j = 255; i < 128; i++, j--) {
float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
float w = 0.5 * (wi * wi + wj * wj);
mdct_window[i] = mdct_window[511 - i] = wi / w;
mdct_window[j] = mdct_window[511 - j] = wj / w;
}
}
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
{
ATRAC3Context *q = avctx->priv_data;
av_freep(&q->units);
av_freep(&q->decoded_bytes_buffer);
ff_mdct_end(&q->mdct_ctx);
return 0;
}
/**
* Mantissa decoding
*
* @param selector which table the output values are coded with
* @param coding_flag constant length coding or variable length coding
* @param mantissas mantissa output table
* @param num_codes number of values to get
*/
static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
int coding_flag, int *mantissas,
int num_codes)
{
int i, code, huff_symb;
if (selector == 1)
num_codes /= 2;
if (coding_flag != 0) {
/* constant length coding (CLC) */
int num_bits = clc_length_tab[selector];
if (selector > 1) {
for (i = 0; i < num_codes; i++) {
if (num_bits)
code = get_sbits(gb, num_bits);
else
code = 0;
mantissas[i] = code;
}
} else {
for (i = 0; i < num_codes; i++) {
if (num_bits)
code = get_bits(gb, num_bits); // num_bits is always 4 in this case
else
code = 0;
mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
}
}
} else {
/* variable length coding (VLC) */
if (selector != 1) {
for (i = 0; i < num_codes; i++) {
huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
spectral_coeff_tab[selector-1].bits, 3);
huff_symb += 1;
code = huff_symb >> 1;
if (huff_symb & 1)
code = -code;
mantissas[i] = code;
}
} else {
for (i = 0; i < num_codes; i++) {
huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
spectral_coeff_tab[selector - 1].bits, 3);
mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
}
}
}
}
/**
* Restore the quantized band spectrum coefficients
*
* @return subband count, fix for broken specification/files
*/
static int decode_spectrum(GetBitContext *gb, float *output)
{
int num_subbands, coding_mode, i, j, first, last, subband_size;
int subband_vlc_index[32], sf_index[32];
int mantissas[128];
float scale_factor;
num_subbands = get_bits(gb, 5); // number of coded subbands
coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
/* get the VLC selector table for the subbands, 0 means not coded */
for (i = 0; i <= num_subbands; i++)
subband_vlc_index[i] = get_bits(gb, 3);
/* read the scale factor indexes from the stream */
for (i = 0; i <= num_subbands; i++) {
if (subband_vlc_index[i] != 0)
sf_index[i] = get_bits(gb, 6);
}
for (i = 0; i <= num_subbands; i++) {
first = subband_tab[i ];
last = subband_tab[i + 1];
subband_size = last - first;
if (subband_vlc_index[i] != 0) {
/* decode spectral coefficients for this subband */
/* TODO: This can be done faster is several blocks share the
* same VLC selector (subband_vlc_index) */
read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
mantissas, subband_size);
/* decode the scale factor for this subband */
scale_factor = ff_atrac_sf_table[sf_index[i]] *
inv_max_quant[subband_vlc_index[i]];
/* inverse quantize the coefficients */
for (j = 0; first < last; first++, j++)
output[first] = mantissas[j] * scale_factor;
} else {
/* this subband was not coded, so zero the entire subband */
memset(output + first, 0, subband_size * sizeof(*output));
}
}
/* clear the subbands that were not coded */
first = subband_tab[i];
memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
return num_subbands;
}
/**
* Restore the quantized tonal components
*
* @param components tonal components
* @param num_bands number of coded bands
*/
static int decode_tonal_components(GetBitContext *gb,
TonalComponent *components, int num_bands)
{
int i, b, c, m;
int nb_components, coding_mode_selector, coding_mode;
int band_flags[4], mantissa[8];
int component_count = 0;
nb_components = get_bits(gb, 5);
/* no tonal components */
if (nb_components == 0)
return 0;
coding_mode_selector = get_bits(gb, 2);
if (coding_mode_selector == 2)
return AVERROR_INVALIDDATA;
coding_mode = coding_mode_selector & 1;
for (i = 0; i < nb_components; i++) {
int coded_values_per_component, quant_step_index;
for (b = 0; b <= num_bands; b++)
band_flags[b] = get_bits1(gb);
coded_values_per_component = get_bits(gb, 3);
quant_step_index = get_bits(gb, 3);
if (quant_step_index <= 1)
return AVERROR_INVALIDDATA;
if (coding_mode_selector == 3)
coding_mode = get_bits1(gb);
for (b = 0; b < (num_bands + 1) * 4; b++) {
int coded_components;
if (band_flags[b >> 2] == 0)
continue;
coded_components = get_bits(gb, 3);
for (c = 0; c < coded_components; c++) {
TonalComponent *cmp = &components[component_count];
int sf_index, coded_values, max_coded_values;
float scale_factor;
sf_index = get_bits(gb, 6);
if (component_count >= 64)
return AVERROR_INVALIDDATA;
cmp->pos = b * 64 + get_bits(gb, 6);
max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values, coded_values);
scale_factor = ff_atrac_sf_table[sf_index] *
inv_max_quant[quant_step_index];
read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
mantissa, coded_values);
cmp->num_coefs = coded_values;
/* inverse quant */
for (m = 0; m < coded_values; m++)
cmp->coef[m] = mantissa[m] * scale_factor;
component_count++;
}
}
}
return component_count;
}
/**
* Decode gain parameters for the coded bands
*
* @param block the gainblock for the current band
* @param num_bands amount of coded bands
*/
static int decode_gain_control(GetBitContext *gb, GainBlock *block,
int num_bands)
{
int b, j;
int *level, *loc;
AtracGainInfo *gain = block->g_block;
for (b = 0; b <= num_bands; b++) {
gain[b].num_points = get_bits(gb, 3);
level = gain[b].lev_code;
loc = gain[b].loc_code;
for (j = 0; j < gain[b].num_points; j++) {
level[j] = get_bits(gb, 4);
loc[j] = get_bits(gb, 5);
if (j && loc[j] <= loc[j - 1])
return AVERROR_INVALIDDATA;
}
}
/* Clear the unused blocks. */
for (; b < 4 ; b++)
gain[b].num_points = 0;
return 0;
}
/**
* Combine the tonal band spectrum and regular band spectrum
*
* @param spectrum output spectrum buffer
* @param num_components number of tonal components
* @param components tonal components for this band
* @return position of the last tonal coefficient
*/
static int add_tonal_components(float *spectrum, int num_components,
TonalComponent *components)
{
int i, j, last_pos = -1;
float *input, *output;
for (i = 0; i < num_components; i++) {
last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
input = components[i].coef;
output = &spectrum[components[i].pos];
for (j = 0; j < components[i].num_coefs; j++)
output[j] += input[j];
}
return last_pos;
}
#define INTERPOLATE(old, new, nsample) \
((old) + (nsample) * 0.125 * ((new) - (old)))
static void reverse_matrixing(float *su1, float *su2, int *prev_code,
int *curr_code)
{
int i, nsample, band;
float mc1_l, mc1_r, mc2_l, mc2_r;
for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
int s1 = prev_code[i];
int s2 = curr_code[i];
nsample = band;
if (s1 != s2) {
/* Selector value changed, interpolation needed. */
mc1_l = matrix_coeffs[s1 * 2 ];
mc1_r = matrix_coeffs[s1 * 2 + 1];
mc2_l = matrix_coeffs[s2 * 2 ];
mc2_r = matrix_coeffs[s2 * 2 + 1];
/* Interpolation is done over the first eight samples. */
for (; nsample < band + 8; nsample++) {
float c1 = su1[nsample];
float c2 = su2[nsample];
c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
su1[nsample] = c2;
su2[nsample] = c1 * 2.0 - c2;
}
}
/* Apply the matrix without interpolation. */
switch (s2) {
case 0: /* M/S decoding */
for (; nsample < band + 256; nsample++) {
float c1 = su1[nsample];
float c2 = su2[nsample];
su1[nsample] = c2 * 2.0;
su2[nsample] = (c1 - c2) * 2.0;
}
break;
case 1:
for (; nsample < band + 256; nsample++) {
float c1 = su1[nsample];
float c2 = su2[nsample];
su1[nsample] = (c1 + c2) * 2.0;
su2[nsample] = c2 * -2.0;
}
break;
case 2:
case 3:
for (; nsample < band + 256; nsample++) {
float c1 = su1[nsample];
float c2 = su2[nsample];
su1[nsample] = c1 + c2;
su2[nsample] = c1 - c2;
}
break;
default:
av_assert1(0);
}
}
}
static void get_channel_weights(int index, int flag, float ch[2])
{
if (index == 7) {
ch[0] = 1.0;
ch[1] = 1.0;
} else {
ch[0] = (index & 7) / 7.0;
ch[1] = sqrt(2 - ch[0] * ch[0]);
if (flag)
FFSWAP(float, ch[0], ch[1]);
}
}
static void channel_weighting(float *su1, float *su2, int *p3)
{
int band, nsample;
/* w[x][y] y=0 is left y=1 is right */
float w[2][2];
if (p3[1] != 7 || p3[3] != 7) {
get_channel_weights(p3[1], p3[0], w[0]);
get_channel_weights(p3[3], p3[2], w[1]);
for (band = 256; band < 4 * 256; band += 256) {
for (nsample = band; nsample < band + 8; nsample++) {
su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
}
for(; nsample < band + 256; nsample++) {
su1[nsample] *= w[1][0];
su2[nsample] *= w[1][1];
}
}
}
}
/**
* Decode a Sound Unit
*
* @param snd the channel unit to be used
* @param output the decoded samples before IQMF in float representation
* @param channel_num channel number
* @param coding_mode the coding mode (JOINT_STEREO or single channels)
*/
static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
ChannelUnit *snd, float *output,
int channel_num, int coding_mode)
{
int band, ret, num_subbands, last_tonal, num_bands;
GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
if (get_bits(gb, 2) != 3) {
av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
return AVERROR_INVALIDDATA;
}
} else {
if (get_bits(gb, 6) != 0x28) {
av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
return AVERROR_INVALIDDATA;
}
}
/* number of coded QMF bands */
snd->bands_coded = get_bits(gb, 2);
ret = decode_gain_control(gb, gain2, snd->bands_coded);
if (ret)
return ret;
snd->num_components = decode_tonal_components(gb, snd->components,
snd->bands_coded);
if (snd->num_components < 0)
return snd->num_components;
num_subbands = decode_spectrum(gb, snd->spectrum);
/* Merge the decoded spectrum and tonal components. */
last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
snd->components);
/* calculate number of used MLT/QMF bands according to the amount of coded
spectral lines */
num_bands = (subband_tab[num_subbands] - 1) >> 8;
if (last_tonal >= 0)
num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
/* Reconstruct time domain samples. */
for (band = 0; band < 4; band++) {
/* Perform the IMDCT step without overlapping. */
if (band <= num_bands)
imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
else
memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
/* gain compensation and overlapping */
ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
&snd->prev_frame[band * 256],
&gain1->g_block[band], &gain2->g_block[band],
256, &output[band * 256]);
}
/* Swap the gain control buffers for the next frame. */
snd->gc_blk_switch ^= 1;
return 0;
}
static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
float **out_samples)
{
ATRAC3Context *q = avctx->priv_data;
int ret, i, ch;
uint8_t *ptr1;
if (q->coding_mode == JOINT_STEREO) {
/* channel coupling mode */
/* Decode sound unit pairs (channels are expected to be even).
* Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
const uint8_t *js_databuf;
int js_pair, js_block_align;
js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */
for (ch = 0; ch < avctx->channels; ch = ch + 2) {
js_pair = ch/2;
js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
/* Set the bitstream reader at the start of first channel sound unit. */
init_get_bits(&q->gb,
js_databuf, js_block_align * 8);
/* decode Sound Unit 1 */
ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
out_samples[ch], ch, JOINT_STEREO);
if (ret != 0)
return ret;
/* Framedata of the su2 in the joint-stereo mode is encoded in
* reverse byte order so we need to swap it first. */
if (js_databuf == q->decoded_bytes_buffer) {
uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
ptr1 = q->decoded_bytes_buffer;
for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
FFSWAP(uint8_t, *ptr1, *ptr2);
} else {
const uint8_t *ptr2 = js_databuf + js_block_align - 1;
for (i = 0; i < js_block_align; i++)
q->decoded_bytes_buffer[i] = *ptr2--;
}
/* Skip the sync codes (0xF8). */
ptr1 = q->decoded_bytes_buffer;
for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
if (i >= js_block_align)
return AVERROR_INVALIDDATA;
}
/* set the bitstream reader at the start of the second Sound Unit */
ret = init_get_bits8(&q->gb,
ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
if (ret < 0)
return ret;
/* Fill the Weighting coeffs delay buffer */
memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
4 * sizeof(*q->weighting_delay[js_pair]));
q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
q->matrix_coeff_index_now[js_pair][i] = q->matrix_coeff_index_next[js_pair][i];
q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
}
/* Decode Sound Unit 2. */
ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
out_samples[ch+1], ch+1, JOINT_STEREO);
if (ret != 0)
return ret;
/* Reconstruct the channel coefficients. */
reverse_matrixing(out_samples[ch], out_samples[ch+1],
q->matrix_coeff_index_prev[js_pair],
q->matrix_coeff_index_now[js_pair]);
channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
}
} else {
/* single channels */
/* Decode the channel sound units. */
for (i = 0; i < avctx->channels; i++) {
/* Set the bitstream reader at the start of a channel sound unit. */
init_get_bits(&q->gb,
databuf + i * avctx->block_align / avctx->channels,
avctx->block_align * 8 / avctx->channels);
ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
out_samples[i], i, q->coding_mode);
if (ret != 0)
return ret;
}
}
/* Apply the iQMF synthesis filter. */
for (i = 0; i < avctx->channels; i++) {
float *p1 = out_samples[i];
float *p2 = p1 + 256;
float *p3 = p2 + 256;
float *p4 = p3 + 256;
ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
}
return 0;
}
static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
int size, float **out_samples)
{
ATRAC3Context *q = avctx->priv_data;
int ret, i;
/* Set the bitstream reader at the start of a channel sound unit. */
init_get_bits(&q->gb, databuf, size * 8);
/* single channels */
/* Decode the channel sound units. */
for (i = 0; i < avctx->channels; i++) {
ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
out_samples[i], i, q->coding_mode);
if (ret != 0)
return ret;
while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
skip_bits(&q->gb, 1);
}
}
/* Apply the iQMF synthesis filter. */
for (i = 0; i < avctx->channels; i++) {
float *p1 = out_samples[i];
float *p2 = p1 + 256;
float *p3 = p2 + 256;
float *p4 = p3 + 256;
ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
}
return 0;
}
static int atrac3_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;
ATRAC3Context *q = avctx->priv_data;
int ret;
const uint8_t *databuf;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
return AVERROR_INVALIDDATA;
}
/* get output buffer */
frame->nb_samples = SAMPLES_PER_FRAME;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
databuf = q->decoded_bytes_buffer;
} else {
databuf = buf;
}
ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
if (ret) {
av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
return ret;
}
*got_frame_ptr = 1;
return avctx->block_align;
}
static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
int ret;
frame->nb_samples = SAMPLES_PER_FRAME;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
(float **)frame->extended_data);
if (ret) {
av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
return ret;
}
*got_frame_ptr = 1;
return avpkt->size;
}
static av_cold void atrac3_init_static_data(void)
{
VLC_TYPE (*table)[2] = atrac3_vlc_table;
int i;
init_imdct_window();
ff_atrac_generate_tables();
/* Initialize the VLC tables. */
for (i = 0; i < 7; i++) {
spectral_coeff_tab[i].table = table;
spectral_coeff_tab[i].table_allocated = 256;
init_vlc(&spectral_coeff_tab[i], ATRAC3_VLC_BITS, huff_tab_sizes[i],
huff_bits[i], 1, 1,
huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
table += 256;
}
}
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
static int static_init_done;
int i, js_pair, ret;
int version, delay, samples_per_frame, frame_factor;
const uint8_t *edata_ptr = avctx->extradata;
ATRAC3Context *q = avctx->priv_data;
AVFloatDSPContext *fdsp;
if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
return AVERROR(EINVAL);
}
if (!static_init_done)
atrac3_init_static_data();
static_init_done = 1;
/* Take care of the codec-specific extradata. */
if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
version = 4;
samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
delay = 0x88E;
q->coding_mode = SINGLE;
} else if (avctx->extradata_size == 14) {
/* Parse the extradata, WAV format */
av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
bytestream_get_le16(&edata_ptr)); // Unknown value always 1
edata_ptr += 4; // samples per channel
q->coding_mode = bytestream_get_le16(&edata_ptr);
av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
bytestream_get_le16(&edata_ptr)); // Unknown always 0
/* setup */
samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
version = 4;
delay = 0x88E;
q->coding_mode = q->coding_mode ? JOINT_STEREO : SINGLE;
q->scrambled_stream = 0;
if (avctx->block_align != 96 * avctx->channels * frame_factor &&
avctx->block_align != 152 * avctx->channels * frame_factor &&
avctx->block_align != 192 * avctx->channels * frame_factor) {
av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
"configuration %d/%d/%d\n", avctx->block_align,
avctx->channels, frame_factor);
return AVERROR_INVALIDDATA;
}
} else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
/* Parse the extradata, RM format. */
version = bytestream_get_be32(&edata_ptr);
samples_per_frame = bytestream_get_be16(&edata_ptr);
delay = bytestream_get_be16(&edata_ptr);
q->coding_mode = bytestream_get_be16(&edata_ptr);
q->scrambled_stream = 1;
} else {
av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
avctx->extradata_size);
return AVERROR(EINVAL);
}
/* Check the extradata */
if (version != 4) {
av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
return AVERROR_INVALIDDATA;
}
if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
samples_per_frame);
return AVERROR_INVALIDDATA;
}
if (delay != 0x88E) {
av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
delay);
return AVERROR_INVALIDDATA;
}
if (q->coding_mode == SINGLE)
av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
else if (q->coding_mode == JOINT_STEREO) {
if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
return AVERROR_INVALIDDATA;
}
av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
} else {
av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
q->coding_mode);
return AVERROR_INVALIDDATA;
}
if (avctx->block_align > 1024 || avctx->block_align <= 0)
return AVERROR(EINVAL);
q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!q->decoded_bytes_buffer)
return AVERROR(ENOMEM);
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* initialize the MDCT transform */
if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
return ret;
}
/* init the joint-stereo decoding data */
for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
q->weighting_delay[js_pair][0] = 0;
q->weighting_delay[js_pair][1] = 7;
q->weighting_delay[js_pair][2] = 0;
q->weighting_delay[js_pair][3] = 7;
q->weighting_delay[js_pair][4] = 0;
q->weighting_delay[js_pair][5] = 7;
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[js_pair][i] = 3;
q->matrix_coeff_index_now[js_pair][i] = 3;
q->matrix_coeff_index_next[js_pair][i] = 3;
}
}
ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!fdsp)
return AVERROR(ENOMEM);
q->vector_fmul = fdsp->vector_fmul;
av_free(fdsp);
q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
if (!q->units)
return AVERROR(ENOMEM);
return 0;
}
AVCodec ff_atrac3_decoder = {
.name = "atrac3",
.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
AVCodec ff_atrac3al_decoder = {
.name = "atrac3al",
.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ATRAC3AL,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3al_decode_frame,
.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
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