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FFmpeg/libavcodec/atrac1.c

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/*
* Atrac 1 compatible decoder
* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 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 libavcodec/atrac1.c
* Atrac 1 compatible decoder.
* This decoder handles raw ATRAC1 data.
*/
/* Many thanks to Tim Craig for all the help! */
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "atrac.h"
#include "atrac1data.h"
#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
#define AT1_FRAME_SIZE AT1_SU_SIZE * 2
#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
#define AT1_MAX_CHANNELS 2
#define AT1_QMF_BANDS 3
#define IDX_LOW_BAND 0
#define IDX_MID_BAND 1
#define IDX_HIGH_BAND 2
/**
* Sound unit struct, one unit is used per channel
*/
typedef struct {
int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
int num_bfus; ///< number of Block Floating Units
int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
float* spectrum[2];
DECLARE_ALIGNED_16(float,spec1[AT1_SU_SAMPLES]); ///< mdct buffer
DECLARE_ALIGNED_16(float,spec2[AT1_SU_SAMPLES]); ///< mdct buffer
DECLARE_ALIGNED_16(float,fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
DECLARE_ALIGNED_16(float,snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
DECLARE_ALIGNED_16(float,last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
} AT1SUCtx;
/**
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED_16(float,spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
DECLARE_ALIGNED_16(float,short_buf[64]); ///< buffer for the short mode
DECLARE_ALIGNED_16(float, low[256]);
DECLARE_ALIGNED_16(float, mid[256]);
DECLARE_ALIGNED_16(float,high[512]);
float* bands[3];
float out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
MDCTContext mdct_ctx[3];
int channels;
DSPContext dsp;
} AT1Ctx;
static float *short_window;
static float *mid_window;
DECLARE_ALIGNED_16(static float, long_window[256]);
static float *window_per_band[3];
/** size of the transform in samples in the long mode for each QMF band */
static const uint16_t samples_per_band[3] = {128, 128, 256};
static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
int rev_spec)
{
MDCTContext* mdct_context;
int transf_size = 1 << nbits;
mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
if (rev_spec) {
int i;
for (i=0 ; i<transf_size/2 ; i++)
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
}
ff_imdct_half(mdct_context, out, spec);
}
static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
{
int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
unsigned int start_pos, ref_pos=0, pos = 0;
for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
band_samples = samples_per_band[band_num];
log2_block_count = su->log2_block_count[band_num];
/* number of mdct blocks in the current QMF band: 1 - for long mode */
/* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
num_blocks = 1 << log2_block_count;
/* mdct block size in samples: 128 (long mode, low & mid bands), */
/* 256 (long mode, high band) and 32 (short mode, all bands) */
block_size = band_samples >> log2_block_count;
/* calc transform size in bits according to the block_size_mode */
nbits = mdct_long_nbits[band_num] - log2_block_count;
if (nbits!=5 && nbits!=7 && nbits!=8)
return -1;
if (num_blocks == 1) {
at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num);
pos += block_size; // move to the next mdct block in the spectrum
} else {
/* calc start position for the 1st short block: 96(128) or 112(256) */
start_pos = (band_samples * (num_blocks - 1)) >> (log2_block_count + 1);
memset(&su->spectrum[0][ref_pos], 0, sizeof(float) * (band_samples * 2));
for (; num_blocks!=0 ; num_blocks--) {
/* use hardcoded nbits for the short mode */
at1_imdct(q, &q->spec[pos], q->short_buf, 5, band_num);
/* overlap and window between short blocks */
q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos],
&su->spectrum[0][ref_pos+start_pos],
q->short_buf,short_window, 0, 16);
start_pos += 32; // use hardcoded block_size
pos += 32;
}
}
/* overlap and window with the previous frame and output the result */
q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples/2],
&su->spectrum[0][ref_pos], window_per_band[band_num], 0, band_samples/2);
ref_pos += band_samples;
}
/* Swap buffers so the mdct overlap works */
FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
return 0;
}
/**
* Parse the block size mode byte
*/
static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
{
int log2_block_count_tmp, i;
for(i=0 ; i<2 ; i++) {
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
return -1;
log2_block_cnt[i] = 2 - log2_block_count_tmp;
}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
return -1;
log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
skip_bits(gb, 2);
return 0;
}
static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
float spec[AT1_SU_SAMPLES])
{
int bits_used, band_num, bfu_num, i;
/* parse the info byte (2nd byte) telling how much BFUs were coded */
su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
/* calc number of consumed bits:
num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
bits_used = su->num_bfus * 10 + 32 +
bfu_amount_tab2[get_bits(gb, 2)] +
(bfu_amount_tab3[get_bits(gb, 3)] << 1);
/* get word length index (idwl) for each BFU */
for (i=0 ; i<su->num_bfus ; i++)
su->idwls[i] = get_bits(gb, 4);
/* get scalefactor index (idsf) for each BFU */
for (i=0 ; i<su->num_bfus ; i++)
su->idsfs[i] = get_bits(gb, 6);
/* zero idwl/idsf for empty BFUs */
for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
su->idwls[i] = su->idsfs[i] = 0;
/* read in the spectral data and reconstruct MDCT spectrum of this channel */
for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
for (bfu_num=bfu_bands_t[band_num] ; bfu_num<bfu_bands_t[band_num+1] ; bfu_num++) {
int pos;
int num_specs = specs_per_bfu[bfu_num];
int word_len = !!su->idwls[bfu_num] + su->idwls[bfu_num];
float scale_factor = sf_table[su->idsfs[bfu_num]];
bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS)
return -1;
/* get the position of the 1st spec according to the block size mode */
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
if (word_len) {
float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
for (i=0 ; i<num_specs ; i++) {
/* read in a quantized spec and convert it to
* signed int and then inverse quantization
*/
spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
}
} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
memset(&spec[pos], 0, num_specs*sizeof(float));
}
}
}
return 0;
}
void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
{
float temp[256];
float iqmf_temp[512 + 46];
/* combine low and middle bands */
atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
/* delay the signal of the high band by 23 samples */
memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23);
memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256);
/* combine (low + middle) and high bands */
atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
}
static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret, i;
GetBitContext gb;
float* samples = data;
if (buf_size < 212 * q->channels) {
av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
return -1;
}
for (ch=0 ; ch<q->channels ; ch++) {
AT1SUCtx* su = &q->SUs[ch];
init_get_bits(&gb, &buf[212*ch], 212*8);
/* parse block_size_mode, 1st byte */
ret = at1_parse_bsm(&gb, su->log2_block_count);
if (ret < 0)
return ret;
ret = at1_unpack_dequant(&gb, su, q->spec);
if (ret < 0)
return ret;
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
at1_subband_synthesis(q, su, q->out_samples[ch]);
}
/* round, convert to 16bit and interleave */
if (q->channels == 1) {
/* mono */
q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1<<15),
32700.0 / (1<<15), AT1_SU_SAMPLES);
} else {
/* stereo */
for (i = 0; i < AT1_SU_SAMPLES; i++) {
samples[i*2] = av_clipf(q->out_samples[0][i], -32700.0 / (1<<15),
32700.0 / (1<<15));
samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700.0 / (1<<15),
32700.0 / (1<<15));
}
}
*data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
return avctx->block_align;
}
static av_cold void init_mdct_windows(void)
{
int i;
/** The mid and long windows uses the same sine window splitted
* in the middle and wrapped into zero/one regions as follows:
*
* region of "ones"
* -------------
* /
* / 1st half
* / of the sine
* / window
* ---------/
* zero region
*
* The mid and short windows are subsets of the long window.
*/
/* Build "zero" region */
memset(long_window, 0, sizeof(long_window));
/* Build sine window region */
short_window = &long_window[112];
ff_sine_window_init(short_window,32);
/* Build "ones" region */
for (i = 0; i < 112; i++)
long_window[144 + i] = 1.0f;
/* Save the mid window subset start */
mid_window = &long_window[64];
/* Prepare the window table */
window_per_band[0] = mid_window;
window_per_band[1] = mid_window;
window_per_band[2] = long_window;
}
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
avctx->sample_fmt = SAMPLE_FMT_FLT;
q->channels = avctx->channels;
/* Init the mdct transforms */
ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1<<15));
ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1<<15));
ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1<<15));
init_mdct_windows();
atrac_generate_tables();
dsputil_init(&q->dsp, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
q->bands[2] = q->high;
/* Prepare the mdct overlap buffers */
q->SUs[0].spectrum[0] = q->SUs[0].spec1;
q->SUs[0].spectrum[1] = q->SUs[0].spec2;
q->SUs[1].spectrum[0] = q->SUs[1].spec1;
q->SUs[1].spectrum[1] = q->SUs[1].spec2;
return 0;
}
AVCodec atrac1_decoder = {
.name = "atrac1",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
.init = atrac1_decode_init,
.close = NULL,
.decode = atrac1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};