libavcodec/atrac1.c - third_party/ffmpeg - Git at Google

 /*
  * ATRAC1 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
  * ATRAC1 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 "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "fft.h"
 #include "internal.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 AT1SUCtx {
     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+39];    ///< delay line for the last stacked QMF filter
 } AT1SUCtx;

 /**
  * The atrac1 context, holds all needed parameters for decoding
  */
 typedef struct AT1Ctx {
     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];
     FFTContext          mdct_ctx[3];
     AVFloatDSPContext   *fdsp;
 } 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->fdsp->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 empty 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 */
     ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);

     /* delay the signal of the high band by 39 samples */
     memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  39);
     memcpy(&su->last_qmf_delay[39], q->bands[2],             sizeof(float) * 256);

     /* combine (low + middle) and high bands */
     ff_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)
 {
     AVFrame *frame     = data;
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     AT1Ctx *q          = avctx->priv_data;
     int ch, ret;
     GetBitContext gb;


     if (buf_size < 212 * avctx->channels) {
         av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
         return AVERROR_INVALIDDATA;
     }

     /* get output buffer */
     frame->nb_samples = AT1_SU_SAMPLES;
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;

     for (ch = 0; ch < avctx->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, (float *)frame->extended_data[ch]);
     }

     *got_frame_ptr = 1;

     return avctx->block_align;
 }


 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]);

     av_freep(&q->fdsp);

     return 0;
 }


 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
 {
     AT1Ctx *q = avctx->priv_data;
     int ret;

     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

     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);
     }

     if (avctx->block_align <= 0) {
         av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
         return AVERROR_PATCHWELCOME;
     }

     /* 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);

     ff_atrac_generate_tables();

     q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);

     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 ff_atrac1_decoder = {
     .name           = "atrac1",
     .long_name      = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_ATRAC1,
     .priv_data_size = sizeof(AT1Ctx),
     .init           = atrac1_decode_init,
     .close          = atrac1_decode_end,
     .decode         = atrac1_decode_frame,
     .capabilities   = AV_CODEC_CAP_DR1,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
 };
	/*
	* ATRAC1 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
	* ATRAC1 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 "libavutil/float_dsp.h"
	#include "avcodec.h"
	#include "get_bits.h"
	#include "fft.h"
	#include "internal.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 AT1SUCtx {
	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+39]; ///< delay line for the last stacked QMF filter
	} AT1SUCtx;

	/**
	* The atrac1 context, holds all needed parameters for decoding
	*/
	typedef struct AT1Ctx {
	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];
	FFTContext mdct_ctx[3];
	AVFloatDSPContext *fdsp;
	} 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->fdsp->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 empty 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 */
	ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);

	/* delay the signal of the high band by 39 samples */
	memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39);
	memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256);

	/* combine (low + middle) and high bands */
	ff_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)
	{
	AVFrame *frame = data;
	const uint8_t *buf = avpkt->data;
	int buf_size = avpkt->size;
	AT1Ctx *q = avctx->priv_data;
	int ch, ret;
	GetBitContext gb;


	if (buf_size < 212 * avctx->channels) {
	av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
	return AVERROR_INVALIDDATA;
	}

	/* get output buffer */
	frame->nb_samples = AT1_SU_SAMPLES;
	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
	return ret;

	for (ch = 0; ch < avctx->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, (float *)frame->extended_data[ch]);
	}

	*got_frame_ptr = 1;

	return avctx->block_align;
	}


	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]);

	av_freep(&q->fdsp);

	return 0;
	}


	static av_cold int atrac1_decode_init(AVCodecContext *avctx)
	{
	AT1Ctx *q = avctx->priv_data;
	int ret;

	avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

	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);
	}

	if (avctx->block_align <= 0) {
	av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
	return AVERROR_PATCHWELCOME;
	}

	/* 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);

	ff_atrac_generate_tables();

	q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);

	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 ff_atrac1_decoder = {
	.name = "atrac1",
	.long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
	.type = AVMEDIA_TYPE_AUDIO,
	.id = AV_CODEC_ID_ATRAC1,
	.priv_data_size = sizeof(AT1Ctx),
	.init = atrac1_decode_init,
	.close = atrac1_decode_end,
	.decode = atrac1_decode_frame,
	.capabilities = AV_CODEC_CAP_DR1,
	.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
	AV_SAMPLE_FMT_NONE },
	};