@@ -960,6 +960,7 @@ CONFIG_LIST="

     rtpdec

     runtime_cpudetect

     shared

+    sinewin

     small

     sram

     static

@@ -1238,8 +1239,8 @@ mdct_select="fft"

 rdft_select="fft"

 # decoders / encoders / hardware accelerators

-aac_decoder_select="mdct rdft"

-aac_encoder_select="mdct"

+aac_decoder_select="mdct rdft sinewin"

+aac_encoder_select="mdct sinewin"

 aac_latm_decoder_select="aac_decoder aac_latm_parser"

 ac3_decoder_select="mdct ac3_parser"

 ac3_encoder_select="mdct ac3dsp"

@@ -1247,12 +1248,12 @@ ac3_fixed_encoder_select="ac3dsp"

 alac_encoder_select="lpc"

 amrnb_decoder_select="lsp"

 amrwb_decoder_select="lsp"

-atrac1_decoder_select="mdct"

+atrac1_decoder_select="mdct sinewin"

 atrac3_decoder_select="mdct"

 binkaudio_dct_decoder_select="mdct rdft dct"

 binkaudio_rdft_decoder_select="mdct rdft"

 cavs_decoder_select="golomb"

-cook_decoder_select="mdct"

+cook_decoder_select="mdct sinewin"

 cscd_decoder_suggest="zlib"

 dca_decoder_select="mdct"

 dnxhd_encoder_select="aandct"

@@ -1315,8 +1316,8 @@ msmpeg4v2_decoder_select="h263_decoder"

 msmpeg4v2_encoder_select="h263_encoder"

 msmpeg4v3_decoder_select="h263_decoder"

 msmpeg4v3_encoder_select="h263_encoder"

-nellymoser_decoder_select="mdct"

-nellymoser_encoder_select="mdct"

+nellymoser_decoder_select="mdct sinewin"

+nellymoser_encoder_select="mdct sinewin"

 png_decoder_select="zlib"

 png_encoder_select="zlib"

 qcelp_decoder_select="lsp"

@@ -1343,7 +1344,7 @@ tiff_decoder_suggest="zlib"

 tiff_encoder_suggest="zlib"

 truehd_decoder_select="mlp_decoder"

 tscc_decoder_select="zlib"

-twinvq_decoder_select="mdct lsp"

+twinvq_decoder_select="mdct lsp sinewin"

 vc1_decoder_select="h263_decoder"

 vc1_crystalhd_decoder_select="crystalhd"

 vc1_dxva2_hwaccel_deps="dxva2api_h DXVA_PictureParameters_wDecodedPictureIndex"

@@ -1356,12 +1357,12 @@ vp6_decoder_select="huffman"

 vp6a_decoder_select="vp6_decoder"

 vp6f_decoder_select="vp6_decoder"

 vp8_decoder_select="h264pred"

-wmapro_decoder_select="mdct"

-wmav1_decoder_select="mdct"

-wmav1_encoder_select="mdct"

-wmav2_decoder_select="mdct"

-wmav2_encoder_select="mdct"

-wmavoice_decoder_select="lsp rdft dct mdct"

+wmapro_decoder_select="mdct sinewin"

+wmav1_decoder_select="mdct sinewin"

+wmav1_encoder_select="mdct sinewin"

+wmav2_decoder_select="mdct sinewin"

+wmav2_encoder_select="mdct sinewin"

+wmavoice_decoder_select="lsp rdft dct mdct sinewin"

 wmv1_decoder_select="h263_decoder"

 wmv1_encoder_select="h263_encoder"

 wmv2_decoder_select="h263_decoder"

@@ -622,11 +622,43 @@ Synchronize read on input.

 @section Advanced options

 @table @option

-@item -map @var{input_stream_id}[:@var{sync_stream_id}]

-Set stream mapping from input streams to output streams.

-Just enumerate the input streams in the order you want them in the output.

-@var{sync_stream_id} if specified sets the input stream to sync

-against.

+@item -map @var{input_file_id}.@var{input_stream_id}[:@var{sync_file_id}.@var{sync_stream_id}]

+

+Designate an input stream as a source for the output file. Each input

+stream is identified by the input file index @var{input_file_id} and

+the input stream index @var{input_stream_id} within the input

+file. Both indexes start at 0. If specified,

+@var{sync_file_id}.@var{sync_stream_id} sets which input stream

+is used as a presentation sync reference.

+

+The @code{-map} options must be specified just after the output file.

+If any @code{-map} options are used, the number of @code{-map} options

+on the command line must match the number of streams in the output

+file. The first @code{-map} option on the command line specifies the

+source for output stream 0, the second @code{-map} option specifies

+the source for output stream 1, etc.

+

+For example, if you have two audio streams in the first input file,

+these streams are identified by "0.0" and "0.1". You can use

+@code{-map} to select which stream to place in an output file. For

+example:

+@example

+ffmpeg -i INPUT out.wav -map 0.1

+@end example

+will map the input stream in @file{INPUT} identified by "0.1" to

+the (single) output stream in @file{out.wav}.

+

+For example, to select the stream with index 2 from input file

+@file{a.mov} (specified by the identifier "0.2"), and stream with

+index 6 from input @file{b.mov} (specified by the identifier "1.6"),

+and copy them to the output file @file{out.mov}:

+@example

+ffmpeg -i a.mov -i b.mov -vcodec copy -acodec copy out.mov -map 0.2 -map 1.6

+@end example

+

+To add more streams to the output file, you can use the

+@code{-newaudio}, @code{-newvideo}, @code{-newsubtitle} options.

+

 @item -map_meta_data @var{outfile}[,@var{metadata}]:@var{infile}[,@var{metadata}]

 Deprecated, use @var{-map_metadata} instead.

@@ -4214,7 +4214,7 @@ static const OptionDef options[] = {

     { "f", HAS_ARG, {(void*)opt_format}, "force format", "fmt" },

     { "i", HAS_ARG, {(void*)opt_input_file}, "input file name", "filename" },

     { "y", OPT_BOOL, {(void*)&file_overwrite}, "overwrite output files" },

-    { "map", HAS_ARG | OPT_EXPERT, {(void*)opt_map}, "set input stream mapping", "file:stream[:syncfile:syncstream]" },

+    { "map", HAS_ARG | OPT_EXPERT, {(void*)opt_map}, "set input stream mapping", "file.stream[:syncfile.syncstream]" },

     { "map_meta_data", HAS_ARG | OPT_EXPERT, {(void*)opt_map_meta_data}, "DEPRECATED set meta data information of outfile from infile",

       "outfile[,metadata]:infile[,metadata]" },

     { "map_metadata", HAS_ARG | OPT_EXPERT, {(void*)opt_map_metadata}, "set metadata information of outfile from infile",

@@ -43,6 +43,7 @@ OBJS-$(CONFIG_LSP)                     += lsp.o

 OBJS-$(CONFIG_MDCT)                    += mdct.o

 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o

 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)

+OBJS-$(CONFIG_SINEWIN)                 += sinewin.o

 OBJS-$(CONFIG_VAAPI)                   += vaapi.o

 OBJS-$(CONFIG_VDPAU)                   += vdpau.o

@@ -50,14 +51,14 @@ OBJS-$(CONFIG_VDPAU)                   += vdpau.o

 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o

 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o

 OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \

-                                          aacadtsdec.o mpeg4audio.o

+                                          aacadtsdec.o mpeg4audio.o kbdwin.o

 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \

                                           aacpsy.o aactab.o      \

                                           psymodel.o iirfilter.o \

-                                          mpeg4audio.o

+                                          mpeg4audio.o kbdwin.o

 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o

 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o

-OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3tab.o ac3.o

+OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3tab.o ac3.o kbdwin.o

 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3tab.o ac3.o

 OBJS-$(CONFIG_ALAC_DECODER)            += alac.o

 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o

@@ -694,7 +695,7 @@ $(SUBDIR)%_tablegen$(HOSTEXESUF): $(SUBDIR)%_tablegen.c $(SUBDIR)%_tablegen.h $(

 	$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $(filter %.c,$^) $(HOSTLIBS)

 GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h     \

-              mdct_tables.h mpegaudio_tables.h motionpixels_tables.h    \

+              sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \

               pcm_tables.h qdm2_tables.h

 GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))

@@ -706,7 +707,7 @@ $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h

 $(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h

 $(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h

 $(SUBDIR)dv.o: $(SUBDIR)dv_tables.h

-$(SUBDIR)mdct.o: $(SUBDIR)mdct_tables.h

+$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h

 $(SUBDIR)mpegaudiodec.o: $(SUBDIR)mpegaudio_tables.h

 $(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h

 $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h

@@ -87,6 +87,8 @@

 #include "fft.h"

 #include "fmtconvert.h"

 #include "lpc.h"

+#include "kbdwin.h"

+#include "sinewin.h"

 #include "aac.h"

 #include "aactab.h"

@@ -1750,7 +1752,7 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,

         ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);

         memset(in + 1024 + 576, 0, 448 * sizeof(float));

     }

-    ff_mdct_calc(&ac->mdct_ltp, out, in);

+    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);

 }

 /**

@@ -1839,9 +1841,9 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)

     // imdct

     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {

         for (i = 0; i < 1024; i += 128)

-            ff_imdct_half(&ac->mdct_small, buf + i, in + i);

+            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);

     } else

-        ff_imdct_half(&ac->mdct, buf, in);

+        ac->mdct.imdct_half(&ac->mdct, buf, in);

     /* window overlapping

      * NOTE: To simplify the overlapping code, all 'meaningless' short to long

@@ -34,6 +34,8 @@

 #include "put_bits.h"

 #include "dsputil.h"

 #include "mpeg4audio.h"

+#include "kbdwin.h"

+#include "sinewin.h"

 #include "aac.h"

 #include "aactab.h"

@@ -250,7 +252,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,

             for (i = 0; i < 1024; i++)

                 sce->saved[i] = audio[i * chans];

         }

-        ff_mdct_calc(&s->mdct1024, sce->coeffs, output);

+        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);

     } else {

         for (k = 0; k < 1024; k += 128) {

             for (i = 448 + k; i < 448 + k + 256; i++)

@@ -259,7 +261,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,

                                          : audio[(i-1024)*chans];

             s->dsp.vector_fmul        (output,     output, k ?  swindow : pwindow, 128);

             s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);

-            ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);

+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);

         }

         for (i = 0; i < 1024; i++)

             sce->saved[i] = audio[i * chans];

@@ -1155,7 +1155,7 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in,

         }

         z[64+63] = z[32];

-        ff_imdct_half(mdct, z, z+64);

+        mdct->imdct_half(mdct, z, z+64);

         for (k = 0; k < 32; k++) {

             W[1][i][k][0] = -z[63-k];

             W[1][i][k][1] = z[k];

@@ -1190,7 +1190,7 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,

                 X[0][i][   n] = -X[0][i][n];

                 X[0][i][32+n] =  X[1][i][31-n];

             }

-            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);

+            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);

             for (n = 0; n < 32; n++) {

                 v[     n] =  mdct_buf[0][63 - 2*n];

                 v[63 - n] = -mdct_buf[0][62 - 2*n];

@@ -1199,8 +1199,8 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,

             for (n = 1; n < 64; n+=2) {

                 X[1][i][n] = -X[1][i][n];

             }

-            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);

-            ff_imdct_half(mdct, mdct_buf[1], X[1][i]);

+            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);

+            mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);

             for (n = 0; n < 64; n++) {

                 v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];

                 v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];

@@ -35,6 +35,7 @@

 #include "ac3_parser.h"

 #include "ac3dec.h"

 #include "ac3dec_data.h"

+#include "kbdwin.h"

 /** Large enough for maximum possible frame size when the specification limit is ignored */

 #define AC3_FRAME_BUFFER_SIZE 32768

@@ -621,13 +622,13 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)

             float *x = s->tmp_output+128;

             for(i=0; i<128; i++)

                 x[i] = s->transform_coeffs[ch][2*i];

-            ff_imdct_half(&s->imdct_256, s->tmp_output, x);

+            s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);

             s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);

             for(i=0; i<128; i++)

                 x[i] = s->transform_coeffs[ch][2*i+1];

-            ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);

+            s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x);

         } else {

-            ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);

+            s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);

             s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);

             memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));

         }

@@ -28,6 +28,7 @@

 #define CONFIG_AC3ENC_FLOAT 1

 #include "ac3enc.c"

+#include "kbdwin.h"

 /**

@@ -74,7 +75,7 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,

  */

 static void mdct512(AC3MDCTContext *mdct, float *out, float *in)

 {

-    ff_mdct_calc(&mdct->fft, out, in);

+    mdct->fft.mdct_calc(&mdct->fft, out, in);

 }

@@ -19,6 +19,7 @@

  */

 #include "libavcodec/fft.h"

+#include "libavcodec/rdft.h"

 #include "libavcodec/synth_filter.h"

 void ff_fft_permute_neon(FFTContext *s, FFTComplex *z);

@@ -36,6 +36,7 @@

 #include "get_bits.h"

 #include "dsputil.h"

 #include "fft.h"

+#include "sinewin.h"

 #include "atrac.h"

 #include "atrac1data.h"

@@ -99,7 +100,7 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,

         for (i = 0; i < transf_size / 2; i++)

             FFSWAP(float, spec[i], spec[transf_size - 1 - i]);

     }

-    ff_imdct_half(mdct_context, out, spec);

+    mdct_context->imdct_half(mdct_context, out, spec);

 }

@@ -146,7 +146,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)

         /**

         * Reverse the odd bands before IMDCT, this is an effect of the QMF transform

         * or it gives better compression to do it this way.

-        * FIXME: It should be possible to handle this in ff_imdct_calc

+        * FIXME: It should be possible to handle this in imdct_calc

         * for that to happen a modification of the prerotation step of

         * all SIMD code and C code is needed.

         * Or fix the functions before so they generate a pre reversed spectrum.

@@ -156,7 +156,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)

             FFSWAP(float, pInput[i], pInput[255-i]);

     }

-    ff_imdct_calc(&q->mdct_ctx,pOutput,pInput);

+    q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);

     /* Perform windowing on the output. */

     dsp.vector_fmul(pOutput, pOutput, mdct_window, 512);

@@ -19,6 +19,8 @@

 #include "libavutil/mem.h"

 #include "avfft.h"

 #include "fft.h"

+#include "rdft.h"

+#include "dct.h"

 /* FFT */

@@ -101,7 +103,7 @@ RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans)

 void av_rdft_calc(RDFTContext *s, FFTSample *data)

 {

-    ff_rdft_calc(s, data);

+    s->rdft_calc(s, data);

 }

 void av_rdft_end(RDFTContext *s)

@@ -128,7 +130,7 @@ DCTContext *av_dct_init(int nbits, enum DCTTransformType inverse)

 void av_dct_calc(DCTContext *s, FFTSample *data)

 {

-    ff_dct_calc(s, data);

+    s->dct_calc(s, data);

 }

 void av_dct_end(DCTContext *s)

@@ -32,7 +32,8 @@

 #define ALT_BITSTREAM_READER_LE

 #include "get_bits.h"

 #include "dsputil.h"

-#include "fft.h"

+#include "dct.h"

+#include "rdft.h"

 #include "fmtconvert.h"

 #include "libavutil/intfloat_readwrite.h"

@@ -223,11 +224,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)

         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {

             coeffs[0] /= 0.5;

-            ff_dct_calc (&s->trans.dct,  coeffs);

+            s->trans.dct.dct_calc(&s->trans.dct,  coeffs);

             s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);

         }

         else if (CONFIG_BINKAUDIO_RDFT_DECODER)

-            ff_rdft_calc(&s->trans.rdft, coeffs);

+            s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);

     }

     s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,

@@ -54,6 +54,7 @@

 #include "bytestream.h"

 #include "fft.h"

 #include "libavutil/audioconvert.h"

+#include "sinewin.h"

 #include "cookdata.h"

@@ -753,7 +754,7 @@ static void imlt_gain(COOKContext *q, float *inbuffer,

     int i;

     /* Inverse modified discrete cosine transform */

-    ff_imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);

+    q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);

     q->imlt_window (q, buffer1, gains_ptr, previous_buffer);

@@ -37,7 +37,7 @@ int main(int argc, char *argv[])

     double (*func)(double) = do_sin ? sin : cos;

     printf("/* This file was generated by libavcodec/costablegen */\n");

-    printf("#include \"libavcodec/fft.h\"\n");

+    printf("#include \"libavcodec/%s\"\n", do_sin ? "rdft.h" : "fft.h");

     for (i = 4; i <= BITS; i++) {

         int m = 1 << i;

         double freq = 2*M_PI/m;

@@ -29,8 +29,7 @@

 #include <math.h>

 #include "libavutil/mathematics.h"

-#include "fft.h"

-#include "x86/fft.h"

+#include "dct.h"

 #define DCT32_FLOAT

 #include "dct32.c"

@@ -59,7 +58,7 @@ static void ff_dst_calc_I_c(DCTContext *ctx, FFTSample *data)

     }

     data[n/2] *= 2;

-    ff_rdft_calc(&ctx->rdft, data);

+    ctx->rdft.rdft_calc(&ctx->rdft, data);

     data[0] *= 0.5f;

@@ -93,7 +92,7 @@ static void ff_dct_calc_I_c(DCTContext *ctx, FFTSample *data)

         data[n - i] = tmp1 + s;

     }

-    ff_rdft_calc(&ctx->rdft, data);

+    ctx->rdft.rdft_calc(&ctx->rdft, data);

     data[n] = data[1];

     data[1] = next;

@@ -121,7 +120,7 @@ static void ff_dct_calc_III_c(DCTContext *ctx, FFTSample *data)

     data[1] = 2 * next;

-    ff_rdft_calc(&ctx->rdft, data);

+    ctx->rdft.rdft_calc(&ctx->rdft, data);

     for (i = 0; i < n / 2; i++) {

         float tmp1 = data[i        ] * inv_n;

@@ -152,7 +151,7 @@ static void ff_dct_calc_II_c(DCTContext *ctx, FFTSample *data)

         data[n-i-1] = tmp1 - s;

     }

-    ff_rdft_calc(&ctx->rdft, data);

+    ctx->rdft.rdft_calc(&ctx->rdft, data);

     next = data[1] * 0.5;

     data[1] *= -1;

@@ -176,11 +175,6 @@ static void dct32_func(DCTContext *ctx, FFTSample *data)

     ctx->dct32(data, data);

 }

-void ff_dct_calc(DCTContext *s, FFTSample *data)

-{

-    s->dct_calc(s, data);

-}

-

 av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)

 {

     int n = 1 << nbits;

new file mode 100644

@@ -0,0 +1,50 @@

+/*

+ * (I)DCT Transforms

+ * Copyright (c) 2009 Peter Ross <pross@xvid.org>

+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>

+ * Copyright (c) 2010 Vitor Sessak

+ *

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg 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.

+ *

+ * FFmpeg 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 FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

+ */

+

+#ifndef AVCODEC_DCT_H

+#define AVCODEC_DCT_H

+

+#include "rdft.h"

+

+struct DCTContext {

+    int nbits;

+    int inverse;

+    RDFTContext rdft;

+    const float *costab;

+    FFTSample *csc2;

+    void (*dct_calc)(struct DCTContext *s, FFTSample *data);

+    void (*dct32)(FFTSample *out, const FFTSample *in);

+};

+

+/**

+ * Set up DCT.

+ * @param nbits           size of the input array:

+ *                        (1 << nbits)     for DCT-II, DCT-III and DST-I

+ *                        (1 << nbits) + 1 for DCT-I

+ *

+ * @note the first element of the input of DST-I is ignored

+ */

+int  ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type);

+void ff_dct_end (DCTContext *s);

+

+#endif

@@ -27,6 +27,8 @@

 #include "libavutil/lfg.h"

 #include "libavutil/log.h"

 #include "fft.h"

+#include "dct.h"

+#include "rdft.h"

 #include <math.h>

 #include <unistd.h>

 #include <sys/time.h>

@@ -327,20 +329,20 @@ int main(int argc, char **argv)

     case TRANSFORM_MDCT:

         if (do_inverse) {

             imdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);

-            ff_imdct_calc(m, tab2, (float *)tab1);

+            m->imdct_calc(m, tab2, (float *)tab1);

             err = check_diff((float *)tab_ref, tab2, fft_size, scale);

         } else {

             mdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);

-            ff_mdct_calc(m, tab2, (float *)tab1);

+            m->mdct_calc(m, tab2, (float *)tab1);

             err = check_diff((float *)tab_ref, tab2, fft_size / 2, scale);

         }

         break;

     case TRANSFORM_FFT:

         memcpy(tab, tab1, fft_size * sizeof(FFTComplex));

-        ff_fft_permute(s, tab);

-        ff_fft_calc(s, tab);

+        s->fft_permute(s, tab);

+        s->fft_calc(s, tab);

         fft_ref(tab_ref, tab1, fft_nbits);

         err = check_diff((float *)tab_ref, (float *)tab, fft_size * 2, 1.0);

@@ -357,7 +359,7 @@ int main(int argc, char **argv)

             memcpy(tab2, tab1, fft_size * sizeof(FFTSample));

             tab2[1] = tab1[fft_size_2].re;

-            ff_rdft_calc(r, tab2);

+            r->rdft_calc(r, tab2);

             fft_ref(tab_ref, tab1, fft_nbits);

             for (i = 0; i < fft_size; i++) {

                 tab[i].re = tab2[i];

@@ -369,7 +371,7 @@ int main(int argc, char **argv)

                 tab2[i]    = tab1[i].re;

                 tab1[i].im = 0;

             }

-            ff_rdft_calc(r, tab2);

+            r->rdft_calc(r, tab2);

             fft_ref(tab_ref, tab1, fft_nbits);

             tab_ref[0].im = tab_ref[fft_size_2].re;

             err = check_diff((float *)tab_ref, (float *)tab2, fft_size, 1.0);

@@ -377,7 +379,7 @@ int main(int argc, char **argv)

         break;

     case TRANSFORM_DCT:

         memcpy(tab, tab1, fft_size * sizeof(FFTComplex));

-        ff_dct_calc(d, tab);

+        d->dct_calc(d, tab);

         if (do_inverse) {

             idct_ref(tab_ref, tab1, fft_nbits);

         } else {

@@ -402,22 +404,22 @@ int main(int argc, char **argv)

                 switch (transform) {

                 case TRANSFORM_MDCT:

                     if (do_inverse) {

-                        ff_imdct_calc(m, (float *)tab, (float *)tab1);

+                        m->imdct_calc(m, (float *)tab, (float *)tab1);

                     } else {

-                        ff_mdct_calc(m, (float *)tab, (float *)tab1);

+                        m->mdct_calc(m, (float *)tab, (float *)tab1);

                     }

                     break;

                 case TRANSFORM_FFT:

                     memcpy(tab, tab1, fft_size * sizeof(FFTComplex));

-                    ff_fft_calc(s, tab);

+                    s->fft_calc(s, tab);

                     break;

                 case TRANSFORM_RDFT:

                     memcpy(tab2, tab1, fft_size * sizeof(FFTSample));

-                    ff_rdft_calc(r, tab2);

+                    r->rdft_calc(r, tab2);

                     break;

                 case TRANSFORM_DCT:

                     memcpy(tab2, tab1, fft_size * sizeof(FFTSample));

-                    ff_dct_calc(d, tab2);

+                    d->dct_calc(d, tab2);

                     break;

                 }

             }

@@ -39,7 +39,14 @@ struct FFTContext {

     /* pre/post rotation tables */

     FFTSample *tcos;

     FFTSample *tsin;

+    /**

+     * Do the permutation needed BEFORE calling fft_calc().

+     */

     void (*fft_permute)(struct FFTContext *s, FFTComplex *z);

+    /**

+     * Do a complex FFT with the parameters defined in ff_fft_init(). The

+     * input data must be permuted before. No 1.0/sqrt(n) normalization is done.

+     */

     void (*fft_calc)(struct FFTContext *s, FFTComplex *z);

     void (*imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input);

     void (*imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input);

@@ -54,20 +61,13 @@ struct FFTContext {

 #if CONFIG_HARDCODED_TABLES

 #define COSTABLE_CONST const

-#define SINTABLE_CONST const

-#define SINETABLE_CONST const

 #else

 #define COSTABLE_CONST

-#define SINTABLE_CONST

-#define SINETABLE_CONST

 #endif

 #define COSTABLE(size) \

     COSTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_cos_##size)[size/2]

-#define SINTABLE(size) \

-    SINTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_sin_##size)[size/2]

-#define SINETABLE(size) \

-    SINETABLE_CONST DECLARE_ALIGNED(16, float, ff_sine_##size)[size]

+

 extern COSTABLE(16);

 extern COSTABLE(32);

 extern COSTABLE(64);

@@ -89,20 +89,6 @@ extern COSTABLE_CONST FFTSample* const ff_cos_tabs[17];

  */

 void ff_init_ff_cos_tabs(int index);

-extern SINTABLE(16);

-extern SINTABLE(32);

-extern SINTABLE(64);

-extern SINTABLE(128);

-extern SINTABLE(256);

-extern SINTABLE(512);

-extern SINTABLE(1024);

-extern SINTABLE(2048);

-extern SINTABLE(4096);

-extern SINTABLE(8192);

-extern SINTABLE(16384);

-extern SINTABLE(32768);

-extern SINTABLE(65536);

-

 /**

  * Set up a complex FFT.

  * @param nbits           log2 of the length of the input array

@@ -115,131 +101,12 @@ void ff_fft_init_mmx(FFTContext *s);

 void ff_fft_init_arm(FFTContext *s);

 void ff_dct_init_mmx(DCTContext *s);

-/**

- * Do the permutation needed BEFORE calling ff_fft_calc().

- */

-static inline void ff_fft_permute(FFTContext *s, FFTComplex *z)

-{

-    s->fft_permute(s, z);

-}

-/**

- * Do a complex FFT with the parameters defined in ff_fft_init(). The

- * input data must be permuted before. No 1.0/sqrt(n) normalization is done.

- */

-static inline void ff_fft_calc(FFTContext *s, FFTComplex *z)

-{

-    s->fft_calc(s, z);

-}

 void ff_fft_end(FFTContext *s);

-/* MDCT computation */

-

-static inline void ff_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input)

-{

-    s->imdct_calc(s, output, input);

-}

-static inline void ff_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input)

-{

-    s->imdct_half(s, output, input);

-}

-

-static inline void ff_mdct_calc(FFTContext *s, FFTSample *output,

-                                const FFTSample *input)

-{

-    s->mdct_calc(s, output, input);

-}

-

-/**

- * Maximum window size for ff_kbd_window_init.

- */

-#define FF_KBD_WINDOW_MAX 1024

-

-/**

- * Generate a Kaiser-Bessel Derived Window.

- * @param   window  pointer to half window

- * @param   alpha   determines window shape

- * @param   n       size of half window, max FF_KBD_WINDOW_MAX

- */

-void ff_kbd_window_init(float *window, float alpha, int n);

-

-/**

- * Generate a sine window.

- * @param   window  pointer to half window

- * @param   n       size of half window

- */

-void ff_sine_window_init(float *window, int n);

-

-/**

- * initialize the specified entry of ff_sine_windows

- */

-void ff_init_ff_sine_windows(int index);

-extern SINETABLE(  32);

-extern SINETABLE(  64);

-extern SINETABLE( 128);

-extern SINETABLE( 256);

-extern SINETABLE( 512);

-extern SINETABLE(1024);

-extern SINETABLE(2048);

-extern SINETABLE(4096);

-extern SINETABLE_CONST float * const ff_sine_windows[13];

-

 int ff_mdct_init(FFTContext *s, int nbits, int inverse, double scale);

 void ff_imdct_calc_c(FFTContext *s, FFTSample *output, const FFTSample *input);

 void ff_imdct_half_c(FFTContext *s, FFTSample *output, const FFTSample *input);

 void ff_mdct_calc_c(FFTContext *s, FFTSample *output, const FFTSample *input);

 void ff_mdct_end(FFTContext *s);

-/* Real Discrete Fourier Transform */

-

-struct RDFTContext {

-    int nbits;

-    int inverse;

-    int sign_convention;

-

-    /* pre/post rotation tables */

-    const FFTSample *tcos;

-    SINTABLE_CONST FFTSample *tsin;

-    FFTContext fft;

-    void (*rdft_calc)(struct RDFTContext *s, FFTSample *z);

-};

-

-/**

- * Set up a real FFT.

- * @param nbits           log2 of the length of the input array

- * @param trans           the type of transform

- */

-int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans);

-void ff_rdft_end(RDFTContext *s);

-

-void ff_rdft_init_arm(RDFTContext *s);

-

-static av_always_inline void ff_rdft_calc(RDFTContext *s, FFTSample *data)

-{

-    s->rdft_calc(s, data);

-}

-

-/* Discrete Cosine Transform */

-

-struct DCTContext {

-    int nbits;

-    int inverse;

-    RDFTContext rdft;

-    const float *costab;

-    FFTSample *csc2;

-    void (*dct_calc)(struct DCTContext *s, FFTSample *data);

-    void (*dct32)(FFTSample *out, const FFTSample *in);

-};

-

-/**

- * Set up DCT.

- * @param nbits           size of the input array:

- *                        (1 << nbits)     for DCT-II, DCT-III and DST-I

- *                        (1 << nbits) + 1 for DCT-I

- *

- * @note the first element of the input of DST-I is ignored

- */

-int  ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type);

-void ff_dct_calc(DCTContext *s, FFTSample *data);

-void ff_dct_end (DCTContext *s);

-

 #endif /* AVCODEC_FFT_H */

@@ -41,6 +41,7 @@

 #include "dsputil.h"

 #include "fft.h"

 #include "libavutil/audioconvert.h"

+#include "sinewin.h"

 #include "imcdata.h"

@@ -564,8 +565,8 @@ static void imc_imdct256(IMCContext *q) {

     }

     /* FFT */

-    ff_fft_permute(&q->fft, q->samples);

-    ff_fft_calc (&q->fft, q->samples);

+    q->fft.fft_permute(&q->fft, q->samples);

+    q->fft.fft_calc   (&q->fft, q->samples);

     /* postrotation, window and reorder */

     for(i = 0; i < COEFFS/2; i++){

new file mode 100644

@@ -0,0 +1,48 @@

+/*

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg 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.

+ *

+ * FFmpeg 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 FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include <assert.h>

+#include <libavutil/mathematics.h>

+#include "libavutil/attributes.h"

+#include "kbdwin.h"

+