[FFmpeg-cvslog] r19811 - in trunk/libavcodec: atrac1.c atrac1data.h
banan
subversion
Thu Sep 10 20:47:02 CEST 2009
Author: banan
Date: Thu Sep 10 20:47:02 2009
New Revision: 19811
Log:
Initial commit of the atrac1 decoder, not hooked up yet
Added:
trunk/libavcodec/atrac1.c
trunk/libavcodec/atrac1data.h
Added: trunk/libavcodec/atrac1.c
==============================================================================
--- /dev/null 00:00:00 1970 (empty, because file is newly added)
+++ trunk/libavcodec/atrac1.c Thu Sep 10 20:47:02 2009 (r19811)
@@ -0,0 +1,402 @@
+/*
+ * 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.
+ */
+
+/* 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
+ int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
+ int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
+ 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,short_buf[64]); ///< buffer for the short mode
+ DECLARE_ALIGNED_16(float, low[256]);
+ DECLARE_ALIGNED_16(float, mid[256]);
+ DECLARE_ALIGNED_16(float,high[512]);
+ float* bands[3];
+ float out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+ MDCTContext mdct_ctx[3];
+ int channels;
+ DSPContext dsp;
+} AT1Ctx;
+
+static float *short_window;
+static float *mid_window;
+DECLARE_ALIGNED_16(static float, long_window[256]);
+static float *window_per_band[3];
+
+/** 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)
+{
+ MDCTContext* mdct_context;
+ int transf_size = 1 << nbits;
+
+ mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
+
+ 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) {
+ 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
+ } else {
+ /* calc start position for the 1st short block: 96(128) or 112(256) */
+ start_pos = (band_samples * (num_blocks - 1)) >> (log2_block_count + 1);
+ memset(&su->spectrum[0][ref_pos], 0, sizeof(float) * (band_samples * 2));
+
+ for (; num_blocks!=0 ; num_blocks--) {
+ /* use hardcoded nbits for the short mode */
+ at1_imdct(q, &q->spec[pos], q->short_buf, 5, band_num);
+
+ /* overlap and window between short blocks */
+ q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos],
+ &su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16);
+ start_pos += 32; // use hardcoded block_size
+ pos += 32;
+ }
+ }
+
+ /* overlap and window with the previous frame and output the result */
+ q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples/2],
+ &su->spectrum[0][ref_pos], window_per_band[band_num], 0, band_samples/2);
+
+ ref_pos += band_samples;
+ }
+
+ /* Swap buffers so the mdct overlap works */
+ FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
+
+ return 0;
+}
+
+
+static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[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_count[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_count[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;
+
+ /* 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++)
+ su->idwls[i] = get_bits(gb, 4);
+
+ /* get scalefactor index (idsf) for each BFU */
+ for (i=0 ; i<su->num_bfus ; i++)
+ su->idsfs[i] = get_bits(gb, 6);
+
+ /* zero idwl/idsf for empty BFUs */
+ for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
+ su->idwls[i] = su->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 = !!su->idwls[bfu_num] + su->idwls[bfu_num];
+ float scale_factor = sf_table[su->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_block_size_mode(&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./(1<<15), 32700./(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./(1<<15), 32700./(1<<15));
+ samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(1<<15));
+ }
+ }
+
+ *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
+ return avctx->block_align;
+}
+
+
+static av_cold void init_mdct_windows(void)
+{
+ int i;
+
+ /** The mid and long windows uses the same sine window splitted
+ * in the middle and wrapped into zero/one regions as follows:
+ *
+ * region of "ones"
+ * -------------
+ * /
+ * / 1st half
+ * / of the sine
+ * / window
+ * ---------/
+ * zero region
+ *
+ * The mid and short windows are subsets of the long window.
+ */
+
+ /* Build "zero" region */
+ memset(long_window, 0, sizeof(long_window));
+ /* Build sine window region */
+ short_window = &long_window[112];
+ ff_sine_window_init(short_window,32);
+ /* Build "ones" region */
+ for (i = 0; i < 112; i++)
+ long_window[144 + i] = 1.0f;
+ /* Save the mid window subset start */
+ mid_window = &long_window[64];
+
+ /* Prepare the window table */
+ window_per_band[0] = mid_window;
+ window_per_band[1] = mid_window;
+ window_per_band[2] = long_window;
+}
+
+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));
+ init_mdct_windows();
+
+ 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;
+}
+
+AVCodec atrac1_decoder = {
+ .name = "atrac1",
+ .type = CODEC_TYPE_AUDIO,
+ .id = CODEC_ID_ATRAC1,
+ .priv_data_size = sizeof(AT1Ctx),
+ .init = atrac1_decode_init,
+ .close = NULL,
+ .decode = atrac1_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
+};
Added: trunk/libavcodec/atrac1data.h
==============================================================================
--- /dev/null 00:00:00 1970 (empty, because file is newly added)
+++ trunk/libavcodec/atrac1data.h Thu Sep 10 20:47:02 2009 (r19811)
@@ -0,0 +1,62 @@
+/*
+ * Atrac 1 compatible decoder data
+ * 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/atrac1data.h
+ * Atrac 1 compatible decoder data
+ */
+
+#ifndef AVCODEC_ATRAC1DATA_H
+#define AVCODEC_ATRAC1DATA_H
+
+static const uint8_t bfu_amount_tab1[8] = {20, 28, 32, 36, 40, 44, 48, 52};
+static const uint8_t bfu_amount_tab2[4] = { 0, 112, 176, 208};
+static const uint8_t bfu_amount_tab3[8] = { 0, 24, 36, 48, 72, 108, 132, 156};
+
+/** number of BFUs in each QMF band */
+static const uint8_t bfu_bands_t[4] = {0, 20, 36, 52};
+
+/** number of spectral lines in each BFU
+ * block floating unit = group of spectral frequencies having the
+ * same quantization parameters like word length and scale factor
+ */
+static const uint8_t specs_per_bfu[52] = {
+ 8, 8, 8, 8, 4, 4, 4, 4, 8, 8, 8, 8, 6, 6, 6, 6, 6, 6, 6, 6, // low band
+ 6, 6, 6, 6, 7, 7, 7, 7, 9, 9, 9, 9, 10, 10, 10, 10, // midle band
+ 12, 12, 12, 12, 12, 12, 12, 12, 20, 20, 20, 20, 20, 20, 20, 20 // high band
+};
+
+/** start position of each BFU in the MDCT spectrum for the long mode */
+static const uint16_t bfu_start_long[52] = {
+ 0, 8, 16, 24, 32, 36, 40, 44, 48, 56, 64, 72, 80, 86, 92, 98, 104, 110, 116, 122,
+ 128, 134, 140, 146, 152, 159, 166, 173, 180, 189, 198, 207, 216, 226, 236, 246,
+ 256, 268, 280, 292, 304, 316, 328, 340, 352, 372, 392, 412, 432, 452, 472, 492,
+};
+
+/** start position of each BFU in the MDCT spectrum for the short mode */
+static const uint16_t bfu_start_short[52] = {
+ 0, 32, 64, 96, 8, 40, 72, 104, 12, 44, 76, 108, 20, 52, 84, 116, 26, 58, 90, 122,
+ 128, 160, 192, 224, 134, 166, 198, 230, 141, 173, 205, 237, 150, 182, 214, 246,
+ 256, 288, 320, 352, 384, 416, 448, 480, 268, 300, 332, 364, 396, 428, 460, 492
+};
+
+#endif /* AVCODEC_ATRAC1DATA_H */
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