/**
* @file libavcodec/atrac1.c
* Atrac 1 compatible decoder.
- * This decoder handles raw ATRAC1 data.
+ * This decoder handles raw ATRAC1 data and probably SDDS data.
*/
/* Many thanks to Tim Craig for all the help! */
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
+#include "fft.h"
#include "atrac.h"
#include "atrac1data.h"
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
+ 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;
/**
*/
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, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
- DECLARE_ALIGNED_16(float, low[256]);
- DECLARE_ALIGNED_16(float, mid[256]);
- DECLARE_ALIGNED_16(float, high[512]);
+ DECLARE_ALIGNED(16, float, low)[256];
+ DECLARE_ALIGNED(16, float, mid)[256];
+ DECLARE_ALIGNED(16, float, high)[512];
float* bands[3];
- DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
- MDCTContext mdct_ctx[3];
+ DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+ FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
} AT1Ctx;
-DECLARE_ALIGNED_16(static float, short_window[32]);
-
/** 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;
+ FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
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++)
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;
+ 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];
/* 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;
+ 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;
+ /* 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 (nbits != 5 && nbits != 7 && nbits != 8)
+ return -1;
+ } else {
+ block_size = 32;
+ nbits = 5;
+ }
- if (num_blocks == 1) {
- /* long blocks */
- 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
+ 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 long blocks */
- q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16],
- &su->spectrum[0][ref_pos], short_window, 0, 16);
- memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
- } else {
- /* short blocks */
- float *prev_buf;
- start_pos = 0;
- prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
- for (; num_blocks != 0; num_blocks--) {
- at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num);
-
- /* overlap and window between short blocks */
- q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
- &su->spectrum[0][ref_pos + start_pos], short_window, 0, 16);
-
- prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
- start_pos += 32; // use hardcoded block_size
- pos += 32;
- }
+ /* overlap and window */
+ q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
+ &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 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;
}
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)];
/* get word length index (idwl) for each BFU */
for (i = 0; i < su->num_bfus; i++)
- su->idwls[i] = get_bits(gb, 4);
+ 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);
+ 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;
+ 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++) {
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 */
+ int word_len = !!idwls[bfu_num] + idwls[bfu_num];
+ float scale_factor = 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)
}
-void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
+static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
{
float temp[256];
float iqmf_temp[512 + 46];
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));
- ff_sine_window_init(short_window, 32);
+ ff_init_ff_sine_windows(5);
atrac_generate_tables();
return 0;
}
+
+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;
+}
+
+
AVCodec atrac1_decoder = {
.name = "atrac1",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
.init = atrac1_decode_init,
- .close = NULL,
+ .close = atrac1_decode_end,
.decode = atrac1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
projects / ffmpeg / blobdiff