diff options
28 files changed, 489 insertions, 408 deletions
diff --git a/.gitignore b/.gitignore index eac1758ed9..3e8ed0046e 100644 --- a/.gitignore +++ b/.gitignore @@ -16,6 +16,9 @@ ffmpeg ffplay ffprobe ffserver +libavcodec/*_tablegen +libavcodec/*_tables.c +libavcodec/*_tables.h libavcodec/libavcodec* libavcore/libavcore* libavdevice/libavdevice* @@ -956,6 +956,7 @@ CONFIG_LIST=" mdct memalign_hack mlib + mpegaudiodsp network nonfree pic @@ -1244,6 +1245,7 @@ symver_if_any="symver_asm_label symver_gnu_asm" dct_select="rdft" mdct_select="fft" rdft_select="fft" +mpegaudiodsp_select="dct" # decoders / encoders / hardware accelerators aac_decoder_select="mdct sinewin" @@ -1297,11 +1299,16 @@ ljpeg_encoder_select="aandct" loco_decoder_select="golomb" mjpeg_encoder_select="aandct" mlp_decoder_select="mlp_parser" -mp1float_decoder_select="dct" -mp2float_decoder_select="dct" -mp3adufloat_decoder_select="dct" -mp3float_decoder_select="dct" -mp3on4float_decoder_select="dct" +mp1_decoder_select="mpegaudiodsp" +mp2_decoder_select="mpegaudiodsp" +mp3adu_decoder_select="mpegaudiodsp" +mp3_decoder_select="mpegaudiodsp" +mp3on4_decoder_select="mpegaudiodsp" +mp1float_decoder_select="mpegaudiodsp" +mp2float_decoder_select="mpegaudiodsp" +mp3adufloat_decoder_select="mpegaudiodsp" +mp3float_decoder_select="mpegaudiodsp" +mp3on4float_decoder_select="mpegaudiodsp" mpeg1video_encoder_select="aandct" mpeg2video_encoder_select="aandct" mpeg4_decoder_select="h263_decoder mpeg4video_parser" @@ -1331,7 +1338,7 @@ nellymoser_encoder_select="mdct sinewin" png_decoder_select="zlib" png_encoder_select="zlib" qcelp_decoder_select="lsp" -qdm2_decoder_select="mdct rdft" +qdm2_decoder_select="mdct rdft mpegaudiodsp" ra_144_encoder_select="lpc" rv10_decoder_select="h263_decoder" rv10_encoder_select="h263_encoder" @@ -2753,7 +2760,7 @@ EOF check_yasm "pextrd [eax], xmm0, 1" && enable yasm || die "yasm not found, use --disable-yasm for a crippled build" - check_yasm "vpaddw xmm0, xmm0, xmm0" || disable avx + check_yasm "vextractf128 xmm0, ymm0, 0" || disable avx fi case "$cpu" in diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 6723118693..db184bcdce 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -41,6 +41,9 @@ OBJS-$(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; } } diff --git a/libavformat/asfdec.c b/libavformat/asfdec.c index 84881592b6..3564bf465f 100644 --- a/libavformat/asfdec.c +++ b/libavformat/asfdec.c @@ -21,6 +21,7 @@ //#define DEBUG +#include "libavutil/bswap.h" #include "libavutil/common.h" #include "libavutil/avstring.h" #include "libavcodec/mpegaudio.h" diff --git a/libavformat/avformat.h b/libavformat/avformat.h index f9091f0afd..2574bbc424 100644 --- a/libavformat/avformat.h +++ b/libavformat/avformat.h @@ -23,8 +23,7 @@ /** - * I return the LIBAVFORMAT_VERSION_INT constant. You got - * a fucking problem with that, douchebag? + * Return the LIBAVFORMAT_VERSION_INT constant. */ unsigned avformat_version(void); |