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FFmpeg/libavcodec/atrac1.c
Benjamin Larsson e704b012f2 Mention SDDS so search engines will pick it up for when someone 
needs to decode the SDDS tracks found on 35 mm movies.

Originally committed as revision 19968 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-09-22 17:05:19 +00:00

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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 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 "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_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, low[256]);
DECLARE_ALIGNED_16(float, mid[256]);
DECLARE_ALIGNED_16(float, high[512]);
float* bands[3];
DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
} 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]);
}
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) {
/* long blocks */
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
/* overlap and window long blocks */
q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16],
&su->spectrum[0][ref_pos], ff_sine_32, 0, 16);
memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
} else {
/* short blocks */
float *prev_buf;
start_pos = 0;
prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
for (; num_blocks != 0; num_blocks--) {
at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num);
/* overlap and window between short blocks */
q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += 32; // use hardcoded block_size
pos += 32;
}
}
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;
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 = 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 -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 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));
ff_sine_window_init(ff_sine_32, 32);
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;
}
static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
AT1Ctx *q = avctx->priv_data;
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
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 = atrac1_decode_end,
.decode = atrac1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
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