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FFmpeg/libavcodec/atrac3.c

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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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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 "avcodec.h"
#include "bytestream.h"
#include "fft.h"
#include "fmtconvert.h"
#include "get_bits.h"
#include "internal.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 {
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);
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_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[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:
assert(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)
{
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) {
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 **)frame->extended_data);
if (ret) {
av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
return ret;
}
*got_frame_ptr = 1;
return avctx->block_align;
}
static void atrac3_init_static_data(AVCodec *codec)
{
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] = powf(2.0, (4 - i));
for (i = -15; i < 16; i++)
gain_tab2[i + 15] = powf(2.0, i * -0.125);
}
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
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);
}
/* 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);
}
/* 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);
}
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,
.init_static_data = atrac3_init_static_data,
.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 },
};