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/* |
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* ATRAC1 compatible decoder |
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* 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
*/
/** |
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* @file |
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* ATRAC1 compatible decoder. |
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* This decoder handles raw ATRAC1 data and probably SDDS data. |
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*/
/* Many thanks to Tim Craig for all the help! */
#include <math.h>
#include <stddef.h>
#include <stdio.h>
|
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#include "libavutil/float_dsp.h" |
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#include "avcodec.h"
#include "get_bits.h" |
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#include "fft.h" |
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#include "internal.h" |
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#include "sinewin.h" |
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#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]; |
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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 |
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} AT1SUCtx;
/**
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit |
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DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer |
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|
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DECLARE_ALIGNED(32, float, low)[256];
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512]; |
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float* bands[3]; |
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FFTContext mdct_ctx[3]; |
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AVFloatDSPContext fdsp; |
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} 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};
|
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
int rev_spec) |
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{ |
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FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
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int transf_size = 1 << nbits;
if (rev_spec) {
int i; |
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for (i = 0; i < transf_size / 2; i++) |
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FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
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} |
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mdct_context->imdct_half(mdct_context, out, spec); |
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}
static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
{ |
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int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
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unsigned int start_pos, ref_pos = 0, pos = 0; |
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|
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for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
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float *prev_buf;
int j;
|
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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;
|
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if (num_blocks == 1) { |
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/* 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; |
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|
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/* calc transform size in bits according to the block_size_mode */
nbits = mdct_long_nbits[band_num] - log2_block_count; |
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if (nbits != 5 && nbits != 7 && nbits != 8) |
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return AVERROR_INVALIDDATA; |
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} else { |
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block_size = 32;
nbits = 5;
}
|
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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); |
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|
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/* overlap and window */ |
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q->fdsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16); |
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|
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prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += block_size;
pos += block_size;
} |
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if (num_blocks == 1)
memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
|
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ref_pos += band_samples;
}
/* Swap buffers so the mdct overlap works */
FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
return 0;
}
|
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/**
* Parse the block size mode byte
*/ |
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|
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
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{
int log2_block_count_tmp, i;
|
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for (i = 0; i < 2; i++) { |
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/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1) |
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return AVERROR_INVALIDDATA; |
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log2_block_cnt[i] = 2 - log2_block_count_tmp; |
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}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) |
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return AVERROR_INVALIDDATA; |
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log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
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skip_bits(gb, 2);
return 0;
}
|
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
float spec[AT1_SU_SAMPLES]) |
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{
int bits_used, band_num, bfu_num, i; |
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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 |
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/* 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 */ |
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for (i = 0; i < su->num_bfus; i++) |
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idwls[i] = get_bits(gb, 4); |
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/* get scalefactor index (idsf) for each BFU */ |
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for (i = 0; i < su->num_bfus; i++) |
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idsfs[i] = get_bits(gb, 6); |
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/* zero idwl/idsf for empty BFUs */
for (i = su->num_bfus; i < AT1_MAX_BFU; i++) |
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idwls[i] = idsfs[i] = 0; |
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/* read in the spectral data and reconstruct MDCT spectrum of this channel */ |
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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++) { |
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int pos;
int num_specs = specs_per_bfu[bfu_num]; |
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int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
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float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]]; |
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bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
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/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS) |
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return AVERROR_INVALIDDATA; |
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/* 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) { |
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float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
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|
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for (i = 0; i < num_specs; i++) { |
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/* 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;
} |
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} else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */ |
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memset(&spec[pos], 0, num_specs * sizeof(float)); |
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}
}
}
return 0;
}
|
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static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) |
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{ |
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float temp[256];
float iqmf_temp[512 + 46]; |
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/* combine low and middle bands */ |
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ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); |
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/* delay the signal of the high band by 23 samples */ |
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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); |
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/* combine (low + middle) and high bands */ |
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ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); |
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}
|
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static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
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int *got_frame_ptr, AVPacket *avpkt) |
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{ |
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AVFrame *frame = data; |
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const uint8_t *buf = avpkt->data; |
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int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data; |
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int ch, ret; |
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GetBitContext gb;
|
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if (buf_size < 212 * avctx->channels) { |
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av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); |
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return AVERROR_INVALIDDATA; |
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}
|
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/* get output buffer */ |
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frame->nb_samples = AT1_SU_SAMPLES; |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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|
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for (ch = 0; ch < avctx->channels; ch++) { |
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AT1SUCtx* su = &q->SUs[ch];
|
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init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
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/* parse block_size_mode, 1st byte */ |
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ret = at1_parse_bsm(&gb, su->log2_block_count); |
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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; |
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at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]); |
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}
|
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*got_frame_ptr = 1; |
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|
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return avctx->block_align;
}
|
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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;
}
|
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static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data; |
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int ret; |
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|
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
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|
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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);
} |
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|
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if (avctx->block_align <= 0) { |
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av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); |
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return AVERROR_PATCHWELCOME;
}
|
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/* Init the mdct transforms */ |
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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;
} |
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|
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ff_init_ff_sine_windows(5); |
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|
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ff_atrac_generate_tables(); |
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|
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avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); |
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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;
}
|
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|
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AVCodec ff_atrac1_decoder = { |
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.name = "atrac1", |
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.long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_ATRAC1, |
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.priv_data_size = sizeof(AT1Ctx), |
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.init = atrac1_decode_init,
.close = atrac1_decode_end,
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1, |
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE }, |
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}; |