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-rw-r--r--ffmpeg/libavcodec/atrac1.c390
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diff --git a/ffmpeg/libavcodec/atrac1.c b/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 "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",
+ .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,
+ .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
+};
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