#include "get_bits.h"
#include "dsputil.h"
#include "mathops.h"
-#include "dct32.h"
+#include "mpegaudiodsp.h"
/*
* TODO:
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+
+/* layer 3 "granule" */
+typedef struct GranuleDef {
+ uint8_t scfsi;
+ int part2_3_length;
+ int big_values;
+ int global_gain;
+ int scalefac_compress;
+ uint8_t block_type;
+ uint8_t switch_point;
+ int table_select[3];
+ int subblock_gain[3];
+ uint8_t scalefac_scale;
+ uint8_t count1table_select;
+ int region_size[3]; /* number of huffman codes in each region */
+ int preflag;
+ int short_start, long_end; /* long/short band indexes */
+ uint8_t scale_factors[40];
+ INTFLOAT sb_hybrid[SBLIMIT * 18]; /* 576 samples */
+} GranuleDef;
+
+typedef struct MPADecodeContext {
+ MPA_DECODE_HEADER
+ uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES];
+ int last_buf_size;
+ /* next header (used in free format parsing) */
+ uint32_t free_format_next_header;
+ GetBitContext gb;
+ GetBitContext in_gb;
+ DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
+ int synth_buf_offset[MPA_MAX_CHANNELS];
+ DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
+ INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+ GranuleDef granules[2][2]; /* Used in Layer 3 */
+#ifdef DEBUG
+ int frame_count;
+#endif
+ int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+ int dither_state;
+ int error_recognition;
+ AVCodecContext* avctx;
+ MPADSPContext mpadsp;
+} MPADecodeContext;
+
#if CONFIG_FLOAT
# define SHR(a,b) ((a)*(1.0f/(1<<(b))))
# define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
#include "mpegaudiodectab.h"
static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g);
-static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
- int *dither_state, OUT_INT *samples, int incr);
/* vlc structure for decoding layer 3 huffman tables */
static VLC huff_vlc[16];
SCALE_GEN(4.0 / 9.0), /* 9 steps */
};
-DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
-
/**
* Convert region offsets to region sizes and truncate
* size to big_values.
int i, j, k;
s->avctx = avctx;
- s->apply_window_mp3 = apply_window_mp3_c;
-#if HAVE_MMX && CONFIG_FLOAT
- ff_mpegaudiodec_init_mmx(s);
-#endif
-#if CONFIG_FLOAT
- ff_dct_init(&s->dct, 5, DCT_II);
-#endif
- if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
+
+ ff_mpadsp_init(&s->mpadsp);
avctx->sample_fmt= OUT_FMT;
s->error_recognition= avctx->error_recognition;
return 0;
}
-
-#if CONFIG_FLOAT
-static inline float round_sample(float *sum)
-{
- float sum1=*sum;
- *sum = 0;
- return sum1;
-}
-
-/* signed 16x16 -> 32 multiply add accumulate */
-#define MACS(rt, ra, rb) rt+=(ra)*(rb)
-
-/* signed 16x16 -> 32 multiply */
-#define MULS(ra, rb) ((ra)*(rb))
-
-#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
-
-#else
-
-static inline int round_sample(int64_t *sum)
-{
- int sum1;
- sum1 = (int)((*sum) >> OUT_SHIFT);
- *sum &= (1<<OUT_SHIFT)-1;
- return av_clip_int16(sum1);
-}
-
-# define MULS(ra, rb) MUL64(ra, rb)
-# define MACS(rt, ra, rb) MAC64(rt, ra, rb)
-# define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
-#endif
-
-#define SUM8(op, sum, w, p) \
-{ \
- op(sum, (w)[0 * 64], (p)[0 * 64]); \
- op(sum, (w)[1 * 64], (p)[1 * 64]); \
- op(sum, (w)[2 * 64], (p)[2 * 64]); \
- op(sum, (w)[3 * 64], (p)[3 * 64]); \
- op(sum, (w)[4 * 64], (p)[4 * 64]); \
- op(sum, (w)[5 * 64], (p)[5 * 64]); \
- op(sum, (w)[6 * 64], (p)[6 * 64]); \
- op(sum, (w)[7 * 64], (p)[7 * 64]); \
-}
-
-#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
-{ \
- INTFLOAT tmp;\
- tmp = p[0 * 64];\
- op1(sum1, (w1)[0 * 64], tmp);\
- op2(sum2, (w2)[0 * 64], tmp);\
- tmp = p[1 * 64];\
- op1(sum1, (w1)[1 * 64], tmp);\
- op2(sum2, (w2)[1 * 64], tmp);\
- tmp = p[2 * 64];\
- op1(sum1, (w1)[2 * 64], tmp);\
- op2(sum2, (w2)[2 * 64], tmp);\
- tmp = p[3 * 64];\
- op1(sum1, (w1)[3 * 64], tmp);\
- op2(sum2, (w2)[3 * 64], tmp);\
- tmp = p[4 * 64];\
- op1(sum1, (w1)[4 * 64], tmp);\
- op2(sum2, (w2)[4 * 64], tmp);\
- tmp = p[5 * 64];\
- op1(sum1, (w1)[5 * 64], tmp);\
- op2(sum2, (w2)[5 * 64], tmp);\
- tmp = p[6 * 64];\
- op1(sum1, (w1)[6 * 64], tmp);\
- op2(sum2, (w2)[6 * 64], tmp);\
- tmp = p[7 * 64];\
- op1(sum1, (w1)[7 * 64], tmp);\
- op2(sum2, (w2)[7 * 64], tmp);\
-}
-
-void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
-{
- int i, j;
-
- /* max = 18760, max sum over all 16 coefs : 44736 */
- for(i=0;i<257;i++) {
- INTFLOAT v;
- v = ff_mpa_enwindow[i];
-#if CONFIG_FLOAT
- v *= 1.0 / (1LL<<(16 + FRAC_BITS));
-#endif
- window[i] = v;
- if ((i & 63) != 0)
- v = -v;
- if (i != 0)
- window[512 - i] = v;
- }
-
- // Needed for avoiding shuffles in ASM implementations
- for(i=0; i < 8; i++)
- for(j=0; j < 16; j++)
- window[512+16*i+j] = window[64*i+32-j];
-
- for(i=0; i < 8; i++)
- for(j=0; j < 16; j++)
- window[512+128+16*i+j] = window[64*i+48-j];
-}
-
-static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
- int *dither_state, OUT_INT *samples, int incr)
-{
- register const MPA_INT *w, *w2, *p;
- int j;
- OUT_INT *samples2;
-#if CONFIG_FLOAT
- float sum, sum2;
-#else
- int64_t sum, sum2;
-#endif
-
- /* copy to avoid wrap */
- memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
-
- samples2 = samples + 31 * incr;
- w = window;
- w2 = window + 31;
-
- sum = *dither_state;
- p = synth_buf + 16;
- SUM8(MACS, sum, w, p);
- p = synth_buf + 48;
- SUM8(MLSS, sum, w + 32, p);
- *samples = round_sample(&sum);
- samples += incr;
- w++;
-
- /* we calculate two samples at the same time to avoid one memory
- access per two sample */
- for(j=1;j<16;j++) {
- sum2 = 0;
- p = synth_buf + 16 + j;
- SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
- p = synth_buf + 48 - j;
- SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
-
- *samples = round_sample(&sum);
- samples += incr;
- sum += sum2;
- *samples2 = round_sample(&sum);
- samples2 -= incr;
- w++;
- w2--;
- }
-
- p = synth_buf + 32;
- SUM8(MLSS, sum, w + 32, p);
- *samples = round_sample(&sum);
- *dither_state= sum;
-}
-
-
-/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
- 32 samples. */
-/* XXX: optimize by avoiding ring buffer usage */
-#if !CONFIG_FLOAT
-void ff_mpa_synth_filter_fixed(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
- MPA_INT *window, int *dither_state,
- OUT_INT *samples, int incr,
- INTFLOAT sb_samples[SBLIMIT])
-{
- register MPA_INT *synth_buf;
- int offset;
-
- offset = *synth_buf_offset;
- synth_buf = synth_buf_ptr + offset;
-
- ff_dct32_fixed(synth_buf, sb_samples);
- apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
-
- offset = (offset - 32) & 511;
- *synth_buf_offset = offset;
-}
-#endif
-
#define C3 FIXHR(0.86602540378443864676/2)
/* 0.5 / cos(pi*(2*i+1)/36) */
samples_ptr = samples + ch;
for(i=0;i<nb_frames;i++) {
RENAME(ff_mpa_synth_filter)(
-#if CONFIG_FLOAT
- s,
-#endif
+ &s->mpadsp,
s->synth_buf[ch], &(s->synth_buf_offset[ch]),
RENAME(ff_mpa_synth_window), &s->dither_state,
samples_ptr, s->nb_channels,
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