summaryrefslogtreecommitdiff
path: root/libavcodec/mpegaudiodsp_template.c
diff options
context:
space:
mode:
authorMans Rullgard <mans@mansr.com>2011-05-16 16:52:01 +0100
committerMans Rullgard <mans@mansr.com>2011-05-19 12:25:34 +0100
commitc4f5c2d6f4ffa3f4b56555059000208a6ba47b55 (patch)
treef46c4f0d94a1e073ac0dae24fab4d1d972bcb2c6 /libavcodec/mpegaudiodsp_template.c
parentea91e77127229015d23a046f1797d3fc6a33e54d (diff)
downloadffmpeg-c4f5c2d6f4ffa3f4b56555059000208a6ba47b55.tar.gz
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>
Diffstat (limited to 'libavcodec/mpegaudiodsp_template.c')
-rw-r--r--libavcodec/mpegaudiodsp_template.c205
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];
+}