Move some mpegaudio functions to new mpegaudiodsp subsystem

This separation allows these functions to be used in a cleaner
fashion from other codecs (e.g. qdm2) and simplifies creating
optimised versions of them.

Signed-off-by: Mans Rullgard <mans@mansr.com>

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];
+}