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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
* Atrac 1 compatible decoder.
* This decoder handles raw ATRAC1 data and probably SDDS 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 "fft.h"
#include "fmtconvert.h"
#include "sinewin.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
float* spectrum[2];
DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
DECLARE_ALIGNED(32, 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 {
AVFrame frame;
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
DECLARE_ALIGNED(32, float, low)[256];
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
FmtConvertContext fmt_conv;
} AT1Ctx;
/** 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)
{
FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
int transf_size = 1 << nbits;
if (rev_spec) {
int i;
for (i = 0; i < transf_size / 2; i++)
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
}
mdct_context->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++) {
float *prev_buf;
int j;
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;
if (num_blocks == 1) {
/* 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 AVERROR_INVALIDDATA;
} else {
block_size = 32;
nbits = 5;
}
start_pos = 0;
prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
for (j=0; j < num_blocks; j++) {
at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
/* overlap and window */
q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += block_size;
pos += block_size;
}
if (num_blocks == 1)
memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
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 AVERROR_INVALIDDATA;
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 AVERROR_INVALIDDATA;
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;
uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
/* 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++)
idwls[i] = get_bits(gb, 4);
/* get scalefactor index (idsf) for each BFU */
for (i = 0; i < su->num_bfus; i++)
idsfs[i] = get_bits(gb, 6);
/* zero idwl/idsf for empty BFUs */
for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
idwls[i] = 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 = !!idwls[bfu_num] + idwls[bfu_num];
float scale_factor = ff_atrac_sf_table[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 AVERROR_INVALIDDATA;
/* 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;
}
static 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 *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret;
GetBitContext gb;
float *samples;
if (buf_size < 212 * q->channels) {
av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
q->frame.nb_samples = AT1_SU_SAMPLES;
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];
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->channels == 1 ? samples : q->out_samples[ch]);
}
/* interleave */
if (q->channels == 2) {
q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
AT1_SU_SAMPLES, 2);
}
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return avctx->block_align;
}
static av_cold int atrac1_decode_end(AVCodecContext * avctx)
{
AT1Ctx *q = avctx->priv_data;
av_freep(&q->out_samples[0]);
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
return 0;
}
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
int ret;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
avctx->channels);
return AVERROR(EINVAL);
}
q->channels = avctx->channels;
if (avctx->channels == 2) {
q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
if (!q->out_samples[0]) {
av_freep(&q->out_samples[0]);
return AVERROR(ENOMEM);
}
}
/* Init the mdct transforms */
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
atrac1_decode_end(avctx);
return ret;
}
ff_init_ff_sine_windows(5);
atrac_generate_tables();
dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, 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;
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
return 0;
}
AVCodec ff_atrac1_decoder = {
.name = "atrac1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
.init = atrac1_decode_init,
.close = atrac1_decode_end,
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};