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>

24 files changed, 470 insertions, 399 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 6723118693..db184bcdce 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -41,6 +41,9 @@ OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
 OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o
+OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
+                                          mpegaudiodsp_fixed.o          \
+                                          mpegaudiodsp_float.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
 OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
@@ -676,24 +679,23 @@ TESTPROGS = cabac dct eval fft fft-fixed h264 iirfilter rangecoder snow
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o
 
-HOSTPROGS = costablegen
+HOSTPROGS = aac_tablegen aacps_tablegen cbrt_tablegen cos_tablegen      \
+            dv_tablegen motionpixels_tablegen mpegaudio_tablegen        \
+            pcm_tablegen qdm2_tablegen sinewin_tablegen
 
 DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
 
-CLEANFILES = sin_tables.c cos_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
+CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
 include $(SUBDIR)../subdir.mak
 
 $(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o
 
-$(SUBDIR)cos_tables.c: $(SUBDIR)costablegen$(HOSTEXESUF)
-	$(M)./$< > $@
-
-$(SUBDIR)cos_fixed_tables.c: $(SUBDIR)costablegen$(HOSTEXESUF)
-	$(M)./$< cos fixed > $@
+TRIG_TABLES  = cos cos_fixed sin
+TRIG_TABLES := $(TRIG_TABLES:%=$(SUBDIR)%_tables.c)
 
-$(SUBDIR)sin_tables.c: $(SUBDIR)costablegen$(HOSTEXESUF)
-	$(M)./$< sin > $@
+$(TRIG_TABLES): $(SUBDIR)%_tables.c: $(SUBDIR)cos_tablegen$(HOSTEXESUF)
+	$(M)./$< $* > $@
 
 ifdef CONFIG_SMALL
 $(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=1
@@ -701,9 +703,6 @@ else
 $(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
 endif
 
-$(SUBDIR)%_tablegen$(HOSTEXESUF): $(SUBDIR)%_tablegen.c $(SUBDIR)%_tablegen.h $(SUBDIR)tableprint.c
-	$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $(filter %.c,$^) $(HOSTLIBS)
-
 GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h     \
               sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
               pcm_tables.h qdm2_tables.h
diff --git a/libavcodec/costablegen.c b/libavcodec/cos_tablegen.c
index 6e818252f7..1577166a46 100644
--- a/libavcodec/costablegen.c
+++ b/libavcodec/cos_tablegen.c
@@ -51,10 +51,10 @@ int main(int argc, char *argv[])
 {
     int i, j;
     int do_sin = argc > 1 && !strcmp(argv[1], "sin");
-    int fixed  = argc > 2 && !strcmp(argv[2], "fixed");
+    int fixed  = argc > 1 &&  strstr(argv[1], "fixed");
     double (*func)(double) = do_sin ? sin : cos;
 
-    printf("/* This file was generated by libavcodec/costablegen */\n");
+    printf("/* This file was automatically generated. */\n");
     printf("#define CONFIG_FFT_FLOAT %d\n", !fixed);
     printf("#include \"libavcodec/%s\"\n", do_sin ? "rdft.h" : "fft.h");
     for (i = 4; i <= BITS; i++) {
diff --git a/libavcodec/h264.c b/libavcodec/h264.c
index ae3d263535..3068db8d85 100644
--- a/libavcodec/h264.c
+++ b/libavcodec/h264.c
@@ -1002,7 +1002,7 @@ static inline void xchg_mb_border(H264Context *h, uint8_t *src_y,
                                   int linesize, int uvlinesize,
                                   int xchg, int simple, int pixel_shift){
     MpegEncContext * const s = &h->s;
-    int deblock_left;
+    int deblock_topleft;
     int deblock_top;
     int top_idx = 1;
     uint8_t *top_border_m1;
@@ -1018,11 +1018,11 @@ static inline void xchg_mb_border(H264Context *h, uint8_t *src_y,
     }
 
     if(h->deblocking_filter == 2) {
-        deblock_left = h->left_type[0];
-        deblock_top  = h->top_type;
+        deblock_topleft = h->slice_table[h->mb_xy - 1 - s->mb_stride] == h->slice_num;
+        deblock_top     = h->top_type;
     } else {
-        deblock_left = (s->mb_x > 0);
-        deblock_top =  (s->mb_y > !!MB_FIELD);
+        deblock_topleft = (s->mb_x > 0);
+        deblock_top     = (s->mb_y > !!MB_FIELD);
     }
 
     src_y  -=   linesize + 1 + pixel_shift;
@@ -1045,7 +1045,7 @@ if (xchg) AV_SWAP64(b,a);\
 else      AV_COPY64(b,a);
 
     if(deblock_top){
-        if(deblock_left){
+        if(deblock_topleft){
             XCHG(top_border_m1 + (8 << pixel_shift), src_y - (7 << pixel_shift), 1);
         }
         XCHG(top_border + (0 << pixel_shift), src_y + (1 << pixel_shift), xchg);
@@ -1056,7 +1056,7 @@ else      AV_COPY64(b,a);
     }
     if(simple || !CONFIG_GRAY || !(s->flags&CODEC_FLAG_GRAY)){
         if(deblock_top){
-            if(deblock_left){
+            if(deblock_topleft){
                 XCHG(top_border_m1 + (16 << pixel_shift), src_cb - (7 << pixel_shift), 1);
                 XCHG(top_border_m1 + (24 << pixel_shift), src_cr - (7 << pixel_shift), 1);
             }
@@ -2561,18 +2561,16 @@ static int fill_filter_caches(H264Context *h, int mb_type){
     return 0;
 }
 
-static void loop_filter(H264Context *h){
+static void loop_filter(H264Context *h, int start_x, int end_x){
     MpegEncContext * const s = &h->s;
     uint8_t  *dest_y, *dest_cb, *dest_cr;
     int linesize, uvlinesize, mb_x, mb_y;
     const int end_mb_y= s->mb_y + FRAME_MBAFF;
     const int old_slice_type= h->slice_type;
-    const int end_mb_x  = s->mb_x;
     const int pixel_shift = h->pixel_shift;
 
     if(h->deblocking_filter) {
-        int start_x= s->resync_mb_y == s->mb_y ? s->resync_mb_x : 0;
-        for(mb_x= start_x; mb_x<end_mb_x; mb_x++){
+        for(mb_x= start_x; mb_x<end_x; mb_x++){
             for(mb_y=end_mb_y - FRAME_MBAFF; mb_y<= end_mb_y; mb_y++){
                 int mb_xy, mb_type;
                 mb_xy = h->mb_xy = mb_x + mb_y*s->mb_stride;
@@ -2617,7 +2615,7 @@ static void loop_filter(H264Context *h){
         }
     }
     h->slice_type= old_slice_type;
-    s->mb_x= end_mb_x;
+    s->mb_x= end_x;
     s->mb_y= end_mb_y - FRAME_MBAFF;
     h->chroma_qp[0] = get_chroma_qp(h, 0, s->qscale);
     h->chroma_qp[1] = get_chroma_qp(h, 1, s->qscale);
@@ -2672,6 +2670,7 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
     H264Context *h = *(void**)arg;
     MpegEncContext * const s = &h->s;
     const int part_mask= s->partitioned_frame ? (AC_END|AC_ERROR) : 0x7F;
+    int lf_x_start = s->mb_x;
 
     s->mb_skip_run= -1;
 
@@ -2710,6 +2709,7 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
 
             if((s->workaround_bugs & FF_BUG_TRUNCATED) && h->cabac.bytestream > h->cabac.bytestream_end + 2){
                 ff_er_add_slice(s, s->resync_mb_x, s->resync_mb_y, s->mb_x-1, s->mb_y, (AC_END|DC_END|MV_END)&part_mask);
+                if (s->mb_x >= lf_x_start) loop_filter(h, lf_x_start, s->mb_x + 1);
                 return 0;
             }
             if( ret < 0 || h->cabac.bytestream > h->cabac.bytestream_end + 2) {
@@ -2719,8 +2719,8 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
             }
 
             if( ++s->mb_x >= s->mb_width ) {
-                loop_filter(h);
-                s->mb_x = 0;
+                loop_filter(h, lf_x_start, s->mb_x);
+                s->mb_x = lf_x_start = 0;
                 decode_finish_row(h);
                 ++s->mb_y;
                 if(FIELD_OR_MBAFF_PICTURE) {
@@ -2731,10 +2731,9 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
             }
 
             if( eos || s->mb_y >= s->mb_height ) {
-                if(s->mb_x)
-                    loop_filter(h);
                 tprintf(s->avctx, "slice end %d %d\n", get_bits_count(&s->gb), s->gb.size_in_bits);
                 ff_er_add_slice(s, s->resync_mb_x, s->resync_mb_y, s->mb_x-1, s->mb_y, (AC_END|DC_END|MV_END)&part_mask);
+                if (s->mb_x > lf_x_start) loop_filter(h, lf_x_start, s->mb_x);
                 return 0;
             }
         }
@@ -2756,13 +2755,12 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
             if(ret<0){
                 av_log(h->s.avctx, AV_LOG_ERROR, "error while decoding MB %d %d\n", s->mb_x, s->mb_y);
                 ff_er_add_slice(s, s->resync_mb_x, s->resync_mb_y, s->mb_x, s->mb_y, (AC_ERROR|DC_ERROR|MV_ERROR)&part_mask);
-
                 return -1;
             }
 
             if(++s->mb_x >= s->mb_width){
-                loop_filter(h);
-                s->mb_x=0;
+                loop_filter(h, lf_x_start, s->mb_x);
+                s->mb_x = lf_x_start = 0;
                 decode_finish_row(h);
                 ++s->mb_y;
                 if(FIELD_OR_MBAFF_PICTURE) {
@@ -2789,9 +2787,8 @@ static int decode_slice(struct AVCodecContext *avctx, void *arg){
             if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->mb_skip_run<=0){
                 tprintf(s->avctx, "slice end %d %d\n", get_bits_count(&s->gb), s->gb.size_in_bits);
                 if(get_bits_count(&s->gb) == s->gb.size_in_bits ){
-                    if(s->mb_x)
-                        loop_filter(h);
                     ff_er_add_slice(s, s->resync_mb_x, s->resync_mb_y, s->mb_x-1, s->mb_y, (AC_END|DC_END|MV_END)&part_mask);
+                    if (s->mb_x > lf_x_start) loop_filter(h, lf_x_start, s->mb_x);
 
                     return 0;
                 }else{
diff --git a/libavcodec/mpc.c b/libavcodec/mpc.c
index ca4c3d0dcb..c2975ec732 100644
--- a/libavcodec/mpc.c
+++ b/libavcodec/mpc.c
@@ -29,6 +29,7 @@
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
+#include "mpegaudiodsp.h"
 #include "mpegaudio.h"
 
 #include "mpc.h"
@@ -51,7 +52,8 @@ static void mpc_synth(MPCContext *c, int16_t *out, int channels)
     for(ch = 0;  ch < channels; ch++){
         samples_ptr = samples + ch;
         for(i = 0; i < SAMPLES_PER_BAND; i++) {
-            ff_mpa_synth_filter_fixed(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
+            ff_mpa_synth_filter_fixed(&c->mpadsp,
+                                c->synth_buf[ch], &(c->synth_buf_offset[ch]),
                                 ff_mpa_synth_window_fixed, &dither_state,
                                 samples_ptr, channels,
                                 c->sb_samples[ch][i]);
diff --git a/libavcodec/mpc.h b/libavcodec/mpc.h
index 2d9755a593..365580ebd0 100644
--- a/libavcodec/mpc.h
+++ b/libavcodec/mpc.h
@@ -52,6 +52,7 @@ typedef struct {
 
 typedef struct {
     DSPContext dsp;
+    MPADSPContext mpadsp;
     GetBitContext gb;
     int IS, MSS, gapless;
     int lastframelen;
diff --git a/libavcodec/mpc7.c b/libavcodec/mpc7.c
index 04657e5ff2..bb21469356 100644
--- a/libavcodec/mpc7.c
+++ b/libavcodec/mpc7.c
@@ -29,7 +29,7 @@
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
-#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
 #include "libavutil/audioconvert.h"
 
 #include "mpc.h"
@@ -68,6 +68,7 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx)
     memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
     av_lfg_init(&c->rnd, 0xDEADBEEF);
     dsputil_init(&c->dsp, avctx);
+    ff_mpadsp_init(&c->mpadsp);
     c->dsp.bswap_buf((uint32_t*)buf, (const uint32_t*)avctx->extradata, 4);
     ff_mpc_init();
     init_get_bits(&gb, buf, 128);
diff --git a/libavcodec/mpc8.c b/libavcodec/mpc8.c
index cf65ffe904..51c5509425 100644
--- a/libavcodec/mpc8.c
+++ b/libavcodec/mpc8.c
@@ -29,7 +29,7 @@
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
-#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
 #include "libavutil/audioconvert.h"
 
 #include "mpc.h"
@@ -120,6 +120,7 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx)
     memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
     av_lfg_init(&c->rnd, 0xDEADBEEF);
     dsputil_init(&c->dsp, avctx);
+    ff_mpadsp_init(&c->mpadsp);
 
     ff_mpc_init();
 
diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 005598797d..7e2ca845d1 100644
--- a/libavcodec/mpegaudio.h
+++ b/libavcodec/mpegaudio.h
@@ -31,9 +31,6 @@
 #endif
 
 #include "avcodec.h"
-#include "get_bits.h"
-#include "dsputil.h"
-#include "dct.h"
 
 /* max frame size, in samples */
 #define MPA_FRAME_SIZE 1152
@@ -50,10 +47,6 @@
 #define MPA_DUAL    2
 #define MPA_MONO    3
 
-/* header + layer + bitrate + freq + lsf/mpeg25 */
-#define SAME_HEADER_MASK \
-   (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
-
 #define MP3_MASK 0xFFFE0CCF
 
 #ifndef FRAC_BITS
@@ -66,46 +59,19 @@
 #define FIX(a)   ((int)((a) * FRAC_ONE))
 
 #if CONFIG_FLOAT
-typedef float OUT_INT;
-#else
-typedef int16_t OUT_INT;
-#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
-#endif
-
-#if CONFIG_FLOAT
 #   define INTFLOAT float
 typedef float MPA_INT;
+typedef float OUT_INT;
 #elif FRAC_BITS <= 15
 #   define INTFLOAT int
 typedef int16_t MPA_INT;
+typedef int16_t OUT_INT;
 #else
 #   define INTFLOAT int
 typedef int32_t MPA_INT;
+typedef int16_t OUT_INT;
 #endif
 
-#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;
-
 #define MPA_DECODE_HEADER \
     int frame_size; \
     int error_protection; \
@@ -122,58 +88,8 @@ typedef struct MPADecodeHeader {
   MPA_DECODE_HEADER
 } MPADecodeHeader;
 
-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;
-#if CONFIG_FLOAT
-    DCTContext dct;
-#endif
-    void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window,
-                             int *dither_state, OUT_INT *samples, int incr);
-} MPADecodeContext;
-
-/* layer 3 huffman tables */
-typedef struct HuffTable {
-    int xsize;
-    const uint8_t *bits;
-    const uint16_t *codes;
-} HuffTable;
-
 int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
 int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
-extern MPA_INT ff_mpa_synth_window_fixed[];
-void ff_mpa_synth_init_fixed(MPA_INT *window);
-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]);
-
-void ff_mpa_synth_init_float(MPA_INT *window);
-void ff_mpa_synth_filter_float(MPADecodeContext *s,
-                         MPA_INT *synth_buf_ptr, int *synth_buf_offset,
-                         MPA_INT *window, int *dither_state,
-                         OUT_INT *samples, int incr,
-                         INTFLOAT sb_samples[SBLIMIT]);
-
-void ff_mpegaudiodec_init_mmx(MPADecodeContext *s);
-void ff_mpegaudiodec_init_altivec(MPADecodeContext *s);
 
 /* fast header check for resync */
 static inline int ff_mpa_check_header(uint32_t header){
diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c
index cfd92d42eb..ee54def579 100644
--- a/libavcodec/mpegaudio_parser.c
+++ b/libavcodec/mpegaudio_parser.c
@@ -35,7 +35,6 @@ typedef struct MpegAudioParseContext {
 #define MPA_HEADER_SIZE 4
 
 /* header + layer + bitrate + freq + lsf/mpeg25 */
-#undef SAME_HEADER_MASK /* mpegaudio.h defines different version */
 #define SAME_HEADER_MASK \
    (0xffe00000 | (3 << 17) | (3 << 10) | (3 << 19))
 
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,
diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c
index 183e5540c2..26454190f5 100644
--- a/libavcodec/mpegaudiodec_float.c
+++ b/libavcodec/mpegaudiodec_float.c
@@ -22,25 +22,6 @@
 #define CONFIG_FLOAT 1
 #include "mpegaudiodec.c"
 
-void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr,
-                               int *synth_buf_offset,
-                               float *window, int *dither_state,
-                               float *samples, int incr,
-                               float sb_samples[SBLIMIT])
-{
-    float *synth_buf;
-    int offset;
-
-    offset = *synth_buf_offset;
-    synth_buf = synth_buf_ptr + offset;
-
-    s->dct.dct32(synth_buf, sb_samples);
-    s->apply_window_mp3(synth_buf, window, dither_state, samples, incr);
-
-    offset = (offset - 32) & 511;
-    *synth_buf_offset = offset;
-}
-
 static void compute_antialias_float(MPADecodeContext *s,
                               GranuleDef *g)
 {
diff --git a/libavcodec/mpegaudiodectab.h b/libavcodec/mpegaudiodectab.h
index 234a70e474..4dd8a7cfc9 100644
--- a/libavcodec/mpegaudiodectab.h
+++ b/libavcodec/mpegaudiodectab.h
@@ -33,6 +33,13 @@
 /*******************************************************/
 /* layer 3 tables */
 
+/* layer 3 huffman tables */
+typedef struct HuffTable {
+    int xsize;
+    const uint8_t *bits;
+    const uint16_t *codes;
+} HuffTable;
+
 /* layer3 scale factor size */
 static const uint8_t slen_table[2][16] = {
     { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 },
diff --git a/libavcodec/tableprint.c b/libavcodec/mpegaudiodsp.c
index 52f6ac2a7c..064acd1e74 100644
--- a/libavcodec/tableprint.c
+++ b/libavcodec/mpegaudiodsp.c
@@ -1,7 +1,5 @@
 /*
- * Generate a file for hardcoded tables
- *
- * Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ * Copyright (c) 2011 Mans Rullgard
  *
  * This file is part of FFmpeg.
  *
@@ -20,22 +18,23 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <stdio.h>
-#include <inttypes.h>
-#include "tableprint.h"
+#include "config.h"
+#include "mpegaudiodsp.h"
+#include "dct.h"
+#include "dct32.h"
+
+void ff_mpadsp_init(MPADSPContext *s)
+{
+    DCTContext dct;
+
+    ff_dct_init(&dct, 5, DCT_II);
 
-WRITE_1D_FUNC(int8_t,   "%3"PRIi8, 15)
-WRITE_1D_FUNC(uint8_t,  "0x%02"PRIx8, 15)
-WRITE_1D_FUNC(uint16_t, "0x%08"PRIx16, 7)
-WRITE_1D_FUNC(uint32_t, "0x%08"PRIx32, 7)
-WRITE_1D_FUNC(float,    "%.18e", 3)
+    s->apply_window_float = ff_mpadsp_apply_window_float;
+    s->apply_window_fixed = ff_mpadsp_apply_window_fixed;
 
-WRITE_2D_FUNC(int8_t)
-WRITE_2D_FUNC(uint8_t)
-WRITE_2D_FUNC(uint32_t)
-WRITE_2D_FUNC(float)
+    s->dct32_float = dct.dct32;
+    s->dct32_fixed = ff_dct32_fixed;
 
-void write_fileheader(void) {
-    printf("/* This file was generated by libavcodec/tableprint */\n");
-    printf("#include <stdint.h>\n");
+    if (HAVE_MMX)     ff_mpadsp_init_mmx(s);
+    if (HAVE_ALTIVEC) ff_mpadsp_init_altivec(s);
 }
diff --git a/libavcodec/mpegaudiodsp.h b/libavcodec/mpegaudiodsp.h
new file mode 100644
index 0000000000..597e2533f5
--- /dev/null
+++ b/libavcodec/mpegaudiodsp.h
@@ -0,0 +1,63 @@
+/*
+ * 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
+ */
+
+#ifndef AVCODEC_MPEGAUDIODSP_H
+#define AVCODEC_MPEGAUDIODSP_H
+
+#include <stdint.h>
+
+typedef struct MPADSPContext {
+    void (*apply_window_float)(float *synth_buf, float *window,
+                               int *dither_state, float *samples, int incr);
+    void (*apply_window_fixed)(int32_t *synth_buf, int32_t *window,
+                               int *dither_state, int16_t *samples, int incr);
+    void (*dct32_float)(float *dst, const float *src);
+    void (*dct32_fixed)(int *dst, const int *src);
+} MPADSPContext;
+
+void ff_mpadsp_init(MPADSPContext *s);
+
+extern int32_t ff_mpa_synth_window_fixed[];
+extern float   ff_mpa_synth_window_float[];
+
+void ff_mpa_synth_filter_fixed(MPADSPContext *s,
+                               int32_t *synth_buf_ptr, int *synth_buf_offset,
+                               int32_t *window, int *dither_state,
+                               int16_t *samples, int incr,
+                               int32_t *sb_samples);
+
+void ff_mpa_synth_filter_float(MPADSPContext *s,
+                               float *synth_buf_ptr, int *synth_buf_offset,
+                               float *window, int *dither_state,
+                               float *samples, int incr,
+                               float *sb_samples);
+
+void ff_mpadsp_init_mmx(MPADSPContext *s);
+void ff_mpadsp_init_altivec(MPADSPContext *s);
+
+void ff_mpa_synth_init_float(float *window);
+void ff_mpa_synth_init_fixed(int32_t *window);
+
+void ff_mpadsp_apply_window_float(float *synth_buf, float *window,
+                                  int *dither_state, float *samples,
+                                  int incr);
+void ff_mpadsp_apply_window_fixed(int32_t *synth_buf, int32_t *window,
+                                  int *dither_state, int16_t *samples,
+                                  int incr);
+
+#endif
diff --git a/libavcodec/mpegaudiodsp_fixed.c b/libavcodec/mpegaudiodsp_fixed.c
new file mode 100644
index 0000000000..3c49a568b7
--- /dev/null
+++ b/libavcodec/mpegaudiodsp_fixed.c
@@ -0,0 +1,20 @@
+/*
+ * 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
+ */
+
+#define CONFIG_FLOAT 0
+#include "mpegaudiodsp_template.c"
diff --git a/libavcodec/mpegaudiodsp_float.c b/libavcodec/mpegaudiodsp_float.c
new file mode 100644
index 0000000000..2d8d53ea26
--- /dev/null
+++ b/libavcodec/mpegaudiodsp_float.c
@@ -0,0 +1,20 @@
+/*
+ * 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
+ */
+
+#define CONFIG_FLOAT 1
+#include "mpegaudiodsp_template.c"
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];
+}
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index 515da6f670..c3d86137fa 100644
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -782,5 +782,3 @@ AVCodec ff_mp2_encoder = {
     .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
 };
-
-#undef FIX
diff --git a/libavcodec/ppc/Makefile b/libavcodec/ppc/Makefile
index 35ea0c38f8..8e37fc791d 100644
--- a/libavcodec/ppc/Makefile
+++ b/libavcodec/ppc/Makefile
@@ -7,11 +7,7 @@ ALTIVEC-OBJS-$(CONFIG_VP5_DECODER)     += ppc/vp3dsp_altivec.o
 ALTIVEC-OBJS-$(CONFIG_VP6_DECODER)     += ppc/vp3dsp_altivec.o
 ALTIVEC-OBJS-$(CONFIG_VP8_DECODER)     += ppc/vp8dsp_altivec.o
 
-ALTIVEC-OBJS-$(CONFIG_MP1FLOAT_DECODER)    += ppc/mpegaudiodec_altivec.o
-ALTIVEC-OBJS-$(CONFIG_MP2FLOAT_DECODER)    += ppc/mpegaudiodec_altivec.o
-ALTIVEC-OBJS-$(CONFIG_MP3FLOAT_DECODER)    += ppc/mpegaudiodec_altivec.o
-ALTIVEC-OBJS-$(CONFIG_MP3ON4FLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
-ALTIVEC-OBJS-$(CONFIG_MP3ADUFLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MPEGAUDIODSP)    += ppc/mpegaudiodec_altivec.o
 
 FFT-OBJS-$(HAVE_GNU_AS)                += ppc/fft_altivec_s.o           \
 
diff --git a/libavcodec/ppc/mpegaudiodec_altivec.c b/libavcodec/ppc/mpegaudiodec_altivec.c
index e087d4add1..2de5dd133a 100644
--- a/libavcodec/ppc/mpegaudiodec_altivec.c
+++ b/libavcodec/ppc/mpegaudiodec_altivec.c
@@ -21,9 +21,8 @@
 
 #include "dsputil_altivec.h"
 #include "util_altivec.h"
-
-#define CONFIG_FLOAT 1
-#include "libavcodec/mpegaudio.h"
+#include "libavcodec/dsputil.h"
+#include "libavcodec/mpegaudiodsp.h"
 
 #define MACS(rt, ra, rb) rt+=(ra)*(rb)
 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
@@ -124,7 +123,7 @@ static void apply_window_mp3(float *in, float *win, int *unused, float *out,
     *out = sum;
 }
 
-void ff_mpegaudiodec_init_altivec(MPADecodeContext *s)
+void ff_mpadsp_init_altivec(MPADSPContext *s)
 {
-    s->apply_window_mp3 = apply_window_mp3;
+    s->apply_window_float = apply_window_mp3;
 }
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index e1165074f7..b9252bab40 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -39,6 +39,7 @@
 #include "get_bits.h"
 #include "dsputil.h"
 #include "rdft.h"
+#include "mpegaudiodsp.h"
 #include "mpegaudio.h"
 
 #include "qdm2data.h"
@@ -170,9 +171,10 @@ typedef struct {
     float output_buffer[1024];
 
     /// Synthesis filter
-    DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
+    MPADSPContext mpadsp;
+    DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
     int synth_buf_offset[MPA_MAX_CHANNELS];
-    DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+    DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
 
     /// Mixed temporary data used in decoding
     float tone_level[MPA_MAX_CHANNELS][30][64];
@@ -329,11 +331,6 @@ static av_cold void qdm2_init_vlc(void)
     }
 }
 
-
-/* for floating point to fixed point conversion */
-static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
 {
     int value;
@@ -482,8 +479,8 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb)
 
     for (ch = 0; ch < q->nb_channels; ch++)
         for (j = 0; j < 64; j++) {
-            q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
-            q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
+            q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+            q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
         }
 }
 
@@ -923,11 +920,11 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
                     for (chs = 0; chs < q->nb_channels; chs++)
                         for (k = 0; k < run; k++)
                             if ((j + k) < 128)
-                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
+                                q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
                 } else {
                     for (k = 0; k < run; k++)
                         if ((j + k) < 128)
-                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
+                            q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
                 }
 
                 j += run;
@@ -1601,7 +1598,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
  */
 static void qdm2_synthesis_filter (QDM2Context *q, int index)
 {
-    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
+    float samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
 
     /* copy sb_samples */
@@ -1613,11 +1610,12 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
                 q->sb_samples[ch][(8 * index) + i][k] = 0;
 
     for (ch = 0; ch < q->nb_channels; ch++) {
-        OUT_INT *samples_ptr = samples + ch;
+        float *samples_ptr = samples + ch;
 
         for (i = 0; i < 8; i++) {
-            ff_mpa_synth_filter_fixed(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
-                ff_mpa_synth_window_fixed, &dither_state,
+            ff_mpa_synth_filter_float(&q->mpadsp,
+                q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+                ff_mpa_synth_window_float, &dither_state,
                 samples_ptr, q->nb_channels,
                 q->sb_samples[ch][(8 * index) + i]);
             samples_ptr += 32 * q->nb_channels;
@@ -1629,7 +1627,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
 
     for (ch = 0; ch < q->channels; ch++)
         for (i = 0; i < q->frame_size; i++)
-            q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
+            q->output_buffer[q->channels * i + ch] += (1 << 23) * samples[q->nb_channels * sub_sampling * i + ch];
 }
 
 
@@ -1646,7 +1644,7 @@ static av_cold void qdm2_init(QDM2Context *q) {
     initialized = 1;
 
     qdm2_init_vlc();
-    ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
+    ff_mpa_synth_init_float(ff_mpa_synth_window_float);
     softclip_table_init();
     rnd_table_init();
     init_noise_samples();
@@ -1863,6 +1861,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     }
 
     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+    ff_mpadsp_init(&s->mpadsp);
 
     qdm2_init(s);
 
diff --git a/libavcodec/tableprint.h b/libavcodec/tableprint.h
index 97a667db33..d81b9a387b 100644
--- a/libavcodec/tableprint.h
+++ b/libavcodec/tableprint.h
@@ -23,8 +23,9 @@
 #ifndef AVCODEC_TABLEPRINT_H
 #define AVCODEC_TABLEPRINT_H
 
-#include <stdint.h>
+#include <inttypes.h>
 #include <stdio.h>
+
 #include "libavutil/common.h"
 
 #define WRITE_1D_FUNC_ARGV(type, linebrk, fmtstr, ...)\
@@ -70,9 +71,6 @@ void write_uint32_t_2d_array(const void *, int, int);
 void write_float_2d_array   (const void *, int, int);
 /** \} */ // end of printfuncs group
 
-/** Write a standard file header */
-void write_fileheader(void);
-
 #define WRITE_ARRAY(prefix, type, name)                 \
     do {                                                \
         const size_t array_size = FF_ARRAY_ELEMS(name); \
@@ -92,4 +90,22 @@ void write_fileheader(void);
         printf("};\n");                                                 \
     } while(0)
 
+
+WRITE_1D_FUNC(int8_t,   "%3"PRIi8, 15)
+WRITE_1D_FUNC(uint8_t,  "0x%02"PRIx8, 15)
+WRITE_1D_FUNC(uint16_t, "0x%08"PRIx16, 7)
+WRITE_1D_FUNC(uint32_t, "0x%08"PRIx32, 7)
+WRITE_1D_FUNC(float,    "%.18e", 3)
+
+WRITE_2D_FUNC(int8_t)
+WRITE_2D_FUNC(uint8_t)
+WRITE_2D_FUNC(uint32_t)
+WRITE_2D_FUNC(float)
+
+static inline void write_fileheader(void)
+{
+    printf("/* This file was automatically generated. */\n");
+    printf("#include <stdint.h>\n");
+}
+
 #endif /* AVCODEC_TABLEPRINT_H */
diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile
index 4775853697..578ce04d77 100644
--- a/libavcodec/x86/Makefile
+++ b/libavcodec/x86/Makefile
@@ -21,11 +21,7 @@ YASM-OBJS-$(CONFIG_VC1_DECODER)        += x86/vc1dsp_yasm.o
 MMX-OBJS-$(CONFIG_AC3DSP)              += x86/ac3dsp_mmx.o
 YASM-OBJS-$(CONFIG_AC3DSP)             += x86/ac3dsp.o
 MMX-OBJS-$(CONFIG_CAVS_DECODER)        += x86/cavsdsp_mmx.o
-MMX-OBJS-$(CONFIG_MP1FLOAT_DECODER)    += x86/mpegaudiodec_mmx.o
-MMX-OBJS-$(CONFIG_MP2FLOAT_DECODER)    += x86/mpegaudiodec_mmx.o
-MMX-OBJS-$(CONFIG_MP3FLOAT_DECODER)    += x86/mpegaudiodec_mmx.o
-MMX-OBJS-$(CONFIG_MP3ON4FLOAT_DECODER) += x86/mpegaudiodec_mmx.o
-MMX-OBJS-$(CONFIG_MP3ADUFLOAT_DECODER) += x86/mpegaudiodec_mmx.o
+MMX-OBJS-$(CONFIG_MPEGAUDIODSP)        += x86/mpegaudiodec_mmx.o
 MMX-OBJS-$(CONFIG_PNG_DECODER)         += x86/png_mmx.o
 MMX-OBJS-$(CONFIG_ENCODERS)            += x86/dsputilenc_mmx.o
 YASM-OBJS-$(CONFIG_ENCODERS)           += x86/dsputilenc_yasm.o
diff --git a/libavcodec/x86/mpegaudiodec_mmx.c b/libavcodec/x86/mpegaudiodec_mmx.c
index 2f34281510..d7f8a0a142 100644
--- a/libavcodec/x86/mpegaudiodec_mmx.c
+++ b/libavcodec/x86/mpegaudiodec_mmx.c
@@ -21,9 +21,8 @@
 
 #include "libavutil/cpu.h"
 #include "libavutil/x86_cpu.h"
-
-#define CONFIG_FLOAT 1
-#include "libavcodec/mpegaudio.h"
+#include "libavcodec/dsputil.h"
+#include "libavcodec/mpegaudiodsp.h"
 
 #define MACS(rt, ra, rb) rt+=(ra)*(rb)
 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
@@ -148,11 +147,11 @@ static void apply_window_mp3(float *in, float *win, int *unused, float *out,
     *out = sum;
 }
 
-void ff_mpegaudiodec_init_mmx(MPADecodeContext *s)
+void ff_mpadsp_init_mmx(MPADSPContext *s)
 {
     int mm_flags = av_get_cpu_flags();
 
     if (mm_flags & AV_CPU_FLAG_SSE2) {
-        s->apply_window_mp3 = apply_window_mp3;
+        s->apply_window_float = apply_window_mp3;
     }
 }