[deb_ffmpeg.git] / ffmpeg / libavcodec / atrac1.c

/*
 * 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 {
    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 {
    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 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 */
    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]);

    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();

    avpriv_float_dsp_init(&q->fdsp, avctx->flags & 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   = CODEC_CAP_DR1,
    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* ATRAC1 compatible decoder
	3	* Copyright (c) 2009 Maxim Poliakovski
	4	* Copyright (c) 2009 Benjamin Larsson
	5	*
	6	* This file is part of FFmpeg.
	7	*
	8	* FFmpeg is free software; you can redistribute it and/or
	9	* modify it under the terms of the GNU Lesser General Public
	10	* License as published by the Free Software Foundation; either
	11	* version 2.1 of the License, or (at your option) any later version.
	12	*
	13	* FFmpeg is distributed in the hope that it will be useful,
	14	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	15	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	16	* Lesser General Public License for more details.
	17	*
	18	* You should have received a copy of the GNU Lesser General Public
	19	* License along with FFmpeg; if not, write to the Free Software
	20	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	21	*/
	22
	23	/**
	24	* @file
	25	* ATRAC1 compatible decoder.
	26	* This decoder handles raw ATRAC1 data and probably SDDS data.
	27	*/
	28
	29	/* Many thanks to Tim Craig for all the help! */
	30
	31	#include <math.h>
	32	#include <stddef.h>
	33	#include <stdio.h>
	34
	35	#include "libavutil/float_dsp.h"
	36	#include "avcodec.h"
	37	#include "get_bits.h"
	38	#include "fft.h"
	39	#include "internal.h"
	40	#include "sinewin.h"
	41
	42	#include "atrac.h"
	43	#include "atrac1data.h"
	44
	45	#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
	46	#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
	47	#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
	48	#define AT1_FRAME_SIZE AT1_SU_SIZE * 2
	49	#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
	50	#define AT1_MAX_CHANNELS 2
	51
	52	#define AT1_QMF_BANDS 3
	53	#define IDX_LOW_BAND 0
	54	#define IDX_MID_BAND 1
	55	#define IDX_HIGH_BAND 2
	56
	57	/**
	58	* Sound unit struct, one unit is used per channel
	59	*/
	60	typedef struct {
	61	int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
	62	int num_bfus; ///< number of Block Floating Units
	63	float* spectrum[2];
	64	DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
65	DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
66	DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
67	DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
68	DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
69	} AT1SUCtx;
70
71	/**
72	* The atrac1 context, holds all needed parameters for decoding
73	*/
74	typedef struct {
75	AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
76	DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
77
78	DECLARE_ALIGNED(32, float, low)[256];
79	DECLARE_ALIGNED(32, float, mid)[256];
80	DECLARE_ALIGNED(32, float, high)[512];
81	float* bands[3];
82	FFTContext mdct_ctx[3];
83	AVFloatDSPContext fdsp;
84	} AT1Ctx;
85
86	/** size of the transform in samples in the long mode for each QMF band */
87	static const uint16_t samples_per_band[3] = {128, 128, 256};
88	static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
89
90
91	static void at1_imdct(AT1Ctx q, float spec, float *out, int nbits,
92	int rev_spec)
93	{
94	FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
95	int transf_size = 1 << nbits;
96
97	if (rev_spec) {
98	int i;
99	for (i = 0; i < transf_size / 2; i++)
100	FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
101	}
102	mdct_context->imdct_half(mdct_context, out, spec);
103	}
104
105
106	static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
107	{
108	int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
109	unsigned int start_pos, ref_pos = 0, pos = 0;
110
111	for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
112	float *prev_buf;
113	int j;
114
115	band_samples = samples_per_band[band_num];
116	log2_block_count = su->log2_block_count[band_num];
117
118	/* number of mdct blocks in the current QMF band: 1 - for long mode */
119	/* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
120	num_blocks = 1 << log2_block_count;
121
122	if (num_blocks == 1) {
123	/* mdct block size in samples: 128 (long mode, low & mid bands), */
124	/* 256 (long mode, high band) and 32 (short mode, all bands) */
125	block_size = band_samples >> log2_block_count;
126
127	/* calc transform size in bits according to the block_size_mode */
128	nbits = mdct_long_nbits[band_num] - log2_block_count;
129
130	if (nbits != 5 && nbits != 7 && nbits != 8)
131	return AVERROR_INVALIDDATA;
132	} else {
133	block_size = 32;
134	nbits = 5;
135	}
136
137	start_pos = 0;
138	prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
139	for (j=0; j < num_blocks; j++) {
140	at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
141
142	/* overlap and window */
143	q->fdsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
144	&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
145
146	prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
147	start_pos += block_size;
148	pos += block_size;
149	}
150
151	if (num_blocks == 1)
152	memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
153
154	ref_pos += band_samples;
155	}
156
157	/* Swap buffers so the mdct overlap works */
158	FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
159
160	return 0;
161	}
162
163	/**
164	* Parse the block size mode byte
165	*/
166
167	static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
168	{
169	int log2_block_count_tmp, i;
170
171	for (i = 0; i < 2; i++) {
172	/* low and mid band */
173	log2_block_count_tmp = get_bits(gb, 2);
174	if (log2_block_count_tmp & 1)
175	return AVERROR_INVALIDDATA;
176	log2_block_cnt[i] = 2 - log2_block_count_tmp;
177	}
178
179	/* high band */
180	log2_block_count_tmp = get_bits(gb, 2);
181	if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
182	return AVERROR_INVALIDDATA;
183	log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
184
185	skip_bits(gb, 2);
186	return 0;
187	}
188
189
190	static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
191	float spec[AT1_SU_SAMPLES])
192	{
193	int bits_used, band_num, bfu_num, i;
194	uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
195	uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
196
197	/* parse the info byte (2nd byte) telling how much BFUs were coded */
198	su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
199
200	/* calc number of consumed bits:
201	num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
202	+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
203	bits_used = su->num_bfus * 10 + 32 +
204	bfu_amount_tab2[get_bits(gb, 2)] +
205	(bfu_amount_tab3[get_bits(gb, 3)] << 1);
206
207	/* get word length index (idwl) for each BFU */
208	for (i = 0; i < su->num_bfus; i++)
209	idwls[i] = get_bits(gb, 4);
210
211	/* get scalefactor index (idsf) for each BFU */
212	for (i = 0; i < su->num_bfus; i++)
213	idsfs[i] = get_bits(gb, 6);
214
215	/* zero idwl/idsf for empty BFUs */
216	for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
217	idwls[i] = idsfs[i] = 0;
218
219	/* read in the spectral data and reconstruct MDCT spectrum of this channel */
220	for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
221	for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
222	int pos;
223
224	int num_specs = specs_per_bfu[bfu_num];
225	int word_len = !!idwls[bfu_num] + idwls[bfu_num];
226	float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
227	bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
228
229	/* check for bitstream overflow */
230	if (bits_used > AT1_SU_MAX_BITS)
231	return AVERROR_INVALIDDATA;
232
233	/* get the position of the 1st spec according to the block size mode */
234	pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
235
236	if (word_len) {
237	float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
238
239	for (i = 0; i < num_specs; i++) {
240	/* read in a quantized spec and convert it to
241	* signed int and then inverse quantization
242	*/
243	spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
244	}
245	} else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */
246	memset(&spec[pos], 0, num_specs * sizeof(float));
247	}
248	}
249	}
250
251	return 0;
252	}
253
254
255	static void at1_subband_synthesis(AT1Ctx q, AT1SUCtx su, float *pOut)
256	{
257	float temp[256];
258	float iqmf_temp[512 + 46];
259
260	/* combine low and middle bands */
261	ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
262
263	/* delay the signal of the high band by 23 samples */
264	memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
265	memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
266
267	/* combine (low + middle) and high bands */
268	ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
269	}
270
271
272	static int atrac1_decode_frame(AVCodecContext avctx, void data,
273	int got_frame_ptr, AVPacket avpkt)
274	{
275	AVFrame *frame = data;
276	const uint8_t *buf = avpkt->data;
277	int buf_size = avpkt->size;
278	AT1Ctx *q = avctx->priv_data;
279	int ch, ret;
280	GetBitContext gb;
281
282
283	if (buf_size < 212 * avctx->channels) {
284	av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
285	return AVERROR_INVALIDDATA;
286	}
287
288	/* get output buffer */
289	frame->nb_samples = AT1_SU_SAMPLES;
290	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
291	return ret;
292
293	for (ch = 0; ch < avctx->channels; ch++) {
294	AT1SUCtx* su = &q->SUs[ch];
295
296	init_get_bits(&gb, &buf[212 * ch], 212 * 8);
297
298	/* parse block_size_mode, 1st byte */
299	ret = at1_parse_bsm(&gb, su->log2_block_count);
300	if (ret < 0)
301	return ret;
302
303	ret = at1_unpack_dequant(&gb, su, q->spec);
304	if (ret < 0)
305	return ret;
306
307	ret = at1_imdct_block(su, q);
308	if (ret < 0)
309	return ret;
310	at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
311	}
312
313	*got_frame_ptr = 1;
314
315	return avctx->block_align;
316	}
317
318
319	static av_cold int atrac1_decode_end(AVCodecContext * avctx)
320	{
321	AT1Ctx *q = avctx->priv_data;
322
323	ff_mdct_end(&q->mdct_ctx[0]);
324	ff_mdct_end(&q->mdct_ctx[1]);
325	ff_mdct_end(&q->mdct_ctx[2]);
326
327	return 0;
328	}
329
330
331	static av_cold int atrac1_decode_init(AVCodecContext *avctx)
332	{
333	AT1Ctx *q = avctx->priv_data;
334	int ret;
335
336	avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
337
338	if (avctx->channels < 1 \|\| avctx->channels > AT1_MAX_CHANNELS) {
339	av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
340	avctx->channels);
341	return AVERROR(EINVAL);
342	}
343
344	if (avctx->block_align <= 0) {
345	av_log(avctx, AV_LOG_ERROR, "Unsupported block align.");
346	return AVERROR_PATCHWELCOME;
347	}
348
349	/* Init the mdct transforms */
350	if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) \|\|
351	(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) \|\|
352	(ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
353	av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
354	atrac1_decode_end(avctx);
355	return ret;
356	}
357
358	ff_init_ff_sine_windows(5);
359
360	ff_atrac_generate_tables();
361
362	avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
363
364	q->bands[0] = q->low;
365	q->bands[1] = q->mid;
366	q->bands[2] = q->high;
367
368	/* Prepare the mdct overlap buffers */
369	q->SUs[0].spectrum[0] = q->SUs[0].spec1;
370	q->SUs[0].spectrum[1] = q->SUs[0].spec2;
371	q->SUs[1].spectrum[0] = q->SUs[1].spec1;
372	q->SUs[1].spectrum[1] = q->SUs[1].spec2;
373
374	return 0;
375	}
376
377
378	AVCodec ff_atrac1_decoder = {
379	.name = "atrac1",
380	.long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
381	.type = AVMEDIA_TYPE_AUDIO,
382	.id = AV_CODEC_ID_ATRAC1,
383	.priv_data_size = sizeof(AT1Ctx),
384	.init = atrac1_decode_init,
385	.close = atrac1_decode_end,
386	.decode = atrac1_decode_frame,
387	.capabilities = CODEC_CAP_DR1,
388	.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
389	AV_SAMPLE_FMT_NONE },
390	};