Merge remote-tracking branch 'qatar/master'

* qatar/master:
  qdm2: Use floating point synthesis filter.
  h264: correct border check.
  h264: fix loopfilter with threading at slice boundaries.
  Fix ff_mpa_synth_filter_fixed() prototype
  Rename costablegen.c ---> cos_tablegen.c.
  Collapse tableprint.c into tableprint.h.
  Simplify trig table rules
  Remove potentially unstable filenames from comments in generated files.
  Ignore generated tables and generated table generator programs.
  Simplify CLEANFILES make variable by using wildcards.
  Remove silly insults from avformat_version() Doxygen documentation.
  mpegaudiodsp: fix x86 and ppc makefiles
  configure: Adjust AVX assembler check.
  mpegaudio: remove unused version of SAME_HEADER_MASK
  mpegaudio: remove useless #undef at end of file
  asfdec: add missing #include for av_bswap32()
  mpegaudio: merge two #if CONFIG_FLOAT blocks
  mpegaudio: move some struct definitions from mpegaudio.h
  Move some mpegaudio functions to new mpegaudiodsp subsystem

Conflicts:
	libavcodec/h264.c
	libavcodec/x86/Makefile

Merged-by: Michael Niedermayer <michaelni@gmx.at>

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,