diff options
Diffstat (limited to 'libavcodec/mpegaudiodec.c')
| -rw-r--r-- | libavcodec/mpegaudiodec.c | 243 |
1 files changed, 50 insertions, 193 deletions
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index c7d830fe21..decb23e665 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -29,7 +29,7 @@ #include "get_bits.h" #include "dsputil.h" #include "mathops.h" -#include "dct32.h" +#include "mpegaudiodsp.h" /* * TODO: @@ -39,6 +39,52 @@ #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)) @@ -68,8 +114,6 @@ #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]; @@ -119,8 +163,6 @@ static const int32_t scale_factor_mult2[3][3] = { 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. @@ -259,14 +301,8 @@ static av_cold int decode_init(AVCodecContext * avctx) 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; @@ -461,183 +497,6 @@ static av_cold int decode_init(AVCodecContext * avctx) 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) */ @@ -1914,9 +1773,7 @@ static int mp_decode_frame(MPADecodeContext *s, 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, |