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, 205 insertions, 0 deletions

diff --git a/libavcodec/mpegaudiodsp_template.c b/libavcodec/mpegaudiodsp_template.c
new file mode 100644
index 0000000000..5561c46135
--- /dev/null
+++ b/libavcodec/mpegaudiodsp_template.c
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2001, 2002 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "libavutil/mem.h"
+#include "dct32.h"
+#include "mathops.h"
+#include "mpegaudiodsp.h"
+#include "mpegaudio.h"
+#include "mpegaudiodata.h"
+
+#if CONFIG_FLOAT
+#define RENAME(n) n##_float
+
+static inline float round_sample(float *sum)
+{
+    float sum1=*sum;
+    *sum = 0;
+    return sum1;
+}
+
+#define MACS(rt, ra, rb) rt+=(ra)*(rb)
+#define MULS(ra, rb) ((ra)*(rb))
+#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
+
+#else
+
+#define RENAME(n) n##_fixed
+#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
+
+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
+
+DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
+
+#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 RENAME(ff_mpadsp_apply_window)(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. */
+void RENAME(ff_mpa_synth_filter)(MPADSPContext *s, MPA_INT *synth_buf_ptr,
+                                 int *synth_buf_offset,
+                                 MPA_INT *window, int *dither_state,
+                                 OUT_INT *samples, int incr,
+                                 MPA_INT *sb_samples)
+{
+    MPA_INT *synth_buf;
+    int offset;
+
+    offset = *synth_buf_offset;
+    synth_buf = synth_buf_ptr + offset;
+
+    s->RENAME(dct32)(synth_buf, sb_samples);
+    s->RENAME(apply_window)(synth_buf, window, dither_state, samples, incr);
+
+    offset = (offset - 32) & 511;
+    *synth_buf_offset = offset;
+}
+
+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];
+}