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8aa29f063c
FFmpeg/libavcodec/atrac3.c
Reimar Döffinger 8aa29f063c atrac3: do not use init_static_data to init VLC data. 
It would be called while registering the codec, which means
it needlessly wastes memory when it is never used.
Instead do the work when first opening the codec.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2012-11-28 16:41:48 +01:00

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/*
* Atrac 3 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
* Atrac 3 compatible decoder.
* This decoder handles Sony's ATRAC3 data.
*
* Container formats used to store atrac 3 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/float_dsp.h"
#include "libavutil/libm.h"
#include "avcodec.h"
#include "bytestream.h"
#include "fft.h"
#include "fmtconvert.h"
#include "get_bits.h"
#include "atrac.h"
#include "atrac3data.h"
#define JOINT_STEREO 0x12
#define STEREO 0x2
#define SAMPLES_PER_FRAME 1024
#define MDCT_SIZE 512
typedef struct GainInfo {
int num_gain_data;
int lev_code[8];
int loc_code[8];
} GainInfo;
typedef struct GainBlock {
GainInfo 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 {
AVFrame frame;
GetBitContext gb;
//@{
/** stream data */
int coding_mode;
ChannelUnit *units;
//@}
//@{
/** joint-stereo related variables */
int matrix_coeff_index_prev[4];
int matrix_coeff_index_now[4];
int matrix_coeff_index_next[4];
int weighting_delay[6];
//@}
//@{
/** data buffers */
uint8_t *decoded_bytes_buffer;
float temp_buf[1070];
//@}
//@{
/** extradata */
int scrambled_stream;
//@}
FFTContext mdct_ctx;
FmtConvertContext fmt_conv;
AVFloatDSPContext fdsp;
} ATRAC3Context;
static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC_TYPE atrac3_vlc_table[4096][2];
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
/**
* 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->fdsp.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);
c = av_be2ne32((0x537F6103 >> (off * 8)) | (0x537F6103 << (32 - (off * 8))));
bytes += 3 + off;
for (i = 0; i < bytes / 4; i++)
output[i] = c ^ buf[i];
if (off)
av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
return off;
}
static av_cold void init_atrac3_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_free(q->units);
av_free(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 i, cf, num_data;
int *level, *loc;
GainInfo *gain = block->g_block;
for (i = 0; i <= num_bands; i++) {
num_data = get_bits(gb, 3);
gain[i].num_gain_data = num_data;
level = gain[i].lev_code;
loc = gain[i].loc_code;
for (cf = 0; cf < gain[i].num_gain_data; cf++) {
level[cf] = get_bits(gb, 4);
loc [cf] = get_bits(gb, 5);
if (cf && loc[cf] <= loc[cf - 1])
return AVERROR_INVALIDDATA;
}
}
/* Clear the unused blocks. */
for (; i < 4 ; i++)
gain[i].num_gain_data = 0;
return 0;
}
/**
* Apply gain parameters and perform the MDCT overlapping part
*
* @param input input buffer
* @param prev previous buffer to perform overlap against
* @param output output buffer
* @param gain1 current band gain info
* @param gain2 next band gain info
*/
static void gain_compensate_and_overlap(float *input, float *prev,
float *output, GainInfo *gain1,
GainInfo *gain2)
{
float g1, g2, gain_inc;
int i, j, num_data, start_loc, end_loc;
if (gain2->num_gain_data == 0)
g1 = 1.0;
else
g1 = gain_tab1[gain2->lev_code[0]];
if (gain1->num_gain_data == 0) {
for (i = 0; i < 256; i++)
output[i] = input[i] * g1 + prev[i];
} else {
num_data = gain1->num_gain_data;
gain1->loc_code[num_data] = 32;
gain1->lev_code[num_data] = 4;
for (i = 0, j = 0; i < num_data; i++) {
start_loc = gain1->loc_code[i] * 8;
end_loc = start_loc + 8;
g2 = gain_tab1[gain1->lev_code[i]];
gain_inc = gain_tab2[gain1->lev_code[i + 1] -
gain1->lev_code[i ] + 15];
/* interpolate */
for (; j < start_loc; j++)
output[j] = (input[j] * g1 + prev[j]) * g2;
/* interpolation is done over eight samples */
for (; j < end_loc; j++) {
output[j] = (input[j] * g1 + prev[j]) * g2;
g2 *= gain_inc;
}
}
for (; j < 256; j++)
output[j] = input[j] * g1 + prev[j];
}
/* Delay for the overlapping part. */
memcpy(prev, &input[256], 256 * sizeof(*prev));
}
/**
* 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[i] += input[i];
}
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 regular stereo/mono)
*/
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 == 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 == -1)
return -1;
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 */
gain_compensate_and_overlap(snd->imdct_buf,
&snd->prev_frame[band * 256],
&output[band * 256],
&gain1->g_block[band],
&gain2->g_block[band]);
}
/* 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;
uint8_t *ptr1;
if (q->coding_mode == JOINT_STEREO) {
/* channel coupling mode */
/* decode Sound Unit 1 */
init_get_bits(&q->gb, databuf, avctx->block_align * 8);
ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
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 (databuf == q->decoded_bytes_buffer) {
uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
ptr1 = q->decoded_bytes_buffer;
for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
FFSWAP(uint8_t, *ptr1, *ptr2);
} else {
const uint8_t *ptr2 = databuf + avctx->block_align - 1;
for (i = 0; i < avctx->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 >= avctx->block_align)
return AVERROR_INVALIDDATA;
}
/* set the bitstream reader at the start of the second Sound Unit*/
init_get_bits(&q->gb, ptr1, avctx->block_align * 8);
/* Fill the Weighting coeffs delay buffer */
memmove(q->weighting_delay, &q->weighting_delay[2],
4 * sizeof(*q->weighting_delay));
q->weighting_delay[4] = get_bits1(&q->gb);
q->weighting_delay[5] = get_bits(&q->gb, 3);
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
}
/* Decode Sound Unit 2. */
ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
out_samples[1], 1, JOINT_STEREO);
if (ret != 0)
return ret;
/* Reconstruct the channel coefficients. */
reverse_matrixing(out_samples[0], out_samples[1],
q->matrix_coeff_index_prev,
q->matrix_coeff_index_now);
channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
} else {
/* normal stereo mode or mono */
/* 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 atrac3_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
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 */
q->frame.nb_samples = SAMPLES_PER_FRAME;
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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 **)q->frame.extended_data);
if (ret) {
av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
return ret;
}
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return avctx->block_align;
}
static void atrac3_init_static_data(void)
{
int i;
init_atrac3_window();
ff_atrac_generate_tables();
/* Initialize the VLC tables. */
for (i = 0; i < 7; i++) {
spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
atrac3_vlc_offs[i ];
init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
huff_bits[i], 1, 1,
huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
}
/* Generate gain tables */
for (i = 0; i < 16; i++)
gain_tab1[i] = exp2f (4 - i);
for (i = -15; i < 16; i++)
gain_tab2[i + 15] = exp2f (i * -0.125);
}
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
static int static_init_done;
int i, ret;
int version, delay, samples_per_frame, frame_factor;
const uint8_t *edata_ptr = avctx->extradata;
ATRAC3Context *q = avctx->priv_data;
if (avctx->channels <= 0 || avctx->channels > 2) {
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->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 : STEREO;
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 == 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(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
avctx->extradata_size);
return AVERROR(EINVAL);
}
if (q->coding_mode == JOINT_STEREO && avctx->channels < 2) {
av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
return AVERROR_INVALIDDATA;
}
/* 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 &&
samples_per_frame != SAMPLES_PER_FRAME * 2) {
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 == STEREO)
av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
else if (q->coding_mode == JOINT_STEREO)
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 >= UINT_MAX / 2)
return AVERROR(EINVAL);
q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
FF_INPUT_BUFFER_PADDING_SIZE);
if (q->decoded_bytes_buffer == NULL)
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");
av_freep(&q->decoded_bytes_buffer);
return ret;
}
/* init the joint-stereo decoding data */
q->weighting_delay[0] = 0;
q->weighting_delay[1] = 7;
q->weighting_delay[2] = 0;
q->weighting_delay[3] = 7;
q->weighting_delay[4] = 0;
q->weighting_delay[5] = 7;
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[i] = 3;
q->matrix_coeff_index_now[i] = 3;
q->matrix_coeff_index_next[i] = 3;
}
avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->units = av_mallocz(sizeof(*q->units) * avctx->channels);
if (!q->units) {
atrac3_decode_close(avctx);
return AVERROR(ENOMEM);
}
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
return 0;
}
AVCodec ff_atrac3_decoder = {
.name = "atrac3",
.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 = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
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