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Merge remote-tracking branch 'newdev/master'

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Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-03-21 03:09:46 +01:00
commit d4a50a2100
44 changed files with 458 additions and 275 deletions
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27
configure vendored
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@ -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"

42
doc/ffmpeg.texi
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@ -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.

2
ffmpeg.c
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@ -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",

11
libavcodec/Makefile
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@ -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

8
libavcodec/aacdec.c
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@ -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

6
libavcodec/aacenc.c
View File

@ -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];

8
libavcodec/aacsbr.c
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@ -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 ];

7
libavcodec/ac3dec.c
View File

@ -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));
}

3
libavcodec/ac3enc_float.c
View File

@ -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);
}

1
libavcodec/arm/fft_init_arm.c
View File

@ -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);

3
libavcodec/atrac1.c
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@ -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);
}

4
libavcodec/atrac3.c
View File

@ -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);

6
libavcodec/avfft.c
View File

@ -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)

7
libavcodec/binkaudio.c
View File

@ -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,

3
libavcodec/cook.c
View File

@ -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);

2
libavcodec/costablegen.c
View File

@ -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;

16
libavcodec/dct.c
View File

@ -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;

50
libavcodec/dct.h Normal file
View File

@ -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

26
libavcodec/fft-test.c
View File

@ -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;
}
}

149
libavcodec/fft.h
View File

@ -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 */

5
libavcodec/imc.c
View File

@ -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++){

48
libavcodec/kbdwin.c Normal file
View File

@ -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"
#define BESSEL_I0_ITER 50 // default: 50 iterations of Bessel I0 approximation
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
{
int i, j;
double sum = 0.0, bessel, tmp;
double local_window[FF_KBD_WINDOW_MAX];
double alpha2 = (alpha * M_PI / n) * (alpha * M_PI / n);
assert(n <= FF_KBD_WINDOW_MAX);
for (i = 0; i < n; i++) {
tmp = i * (n - i) * alpha2;
bessel = 1.0;
for (j = BESSEL_I0_ITER; j > 0; j--)
bessel = bessel * tmp / (j * j) + 1;
sum += bessel;
local_window[i] = sum;
}
sum++;
for (i = 0; i < n; i++)
window[i] = sqrt(local_window[i] / sum);
}

35
libavcodec/kbdwin.h Normal file
View File

@ -0,0 +1,35 @@
/*
* 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
*/
#ifndef AVCODEC_KBDWIN_H
#define AVCODEC_KBDWIN_H
/**
* 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);
#endif

31
libavcodec/mdct.c
View File

@ -30,33 +30,6 @@
* MDCT/IMDCT transforms.
*/
// Generate a Kaiser-Bessel Derived Window.
#define BESSEL_I0_ITER 50 // default: 50 iterations of Bessel I0 approximation
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
{
int i, j;
double sum = 0.0, bessel, tmp;
double local_window[FF_KBD_WINDOW_MAX];
double alpha2 = (alpha * M_PI / n) * (alpha * M_PI / n);
assert(n <= FF_KBD_WINDOW_MAX);
for (i = 0; i < n; i++) {
tmp = i * (n - i) * alpha2;
bessel = 1.0;
for (j = BESSEL_I0_ITER; j > 0; j--)
bessel = bessel * tmp / (j * j) + 1;
sum += bessel;
local_window[i] = sum;
}
sum++;
for (i = 0; i < n; i++)
window[i] = sqrt(local_window[i] / sum);
}
#include "mdct_tablegen.h"
/**
* init MDCT or IMDCT computation.
*/
@ -146,7 +119,7 @@ void ff_imdct_half_c(FFTContext *s, FFTSample *output, const FFTSample *input)
in1 += 2;
in2 -= 2;
}
ff_fft_calc(s, z);
s->fft_calc(s, z);
/* post rotation + reordering */
for(k = 0; k < n8; k++) {
@ -213,7 +186,7 @@ void ff_mdct_calc_c(FFTContext *s, FFTSample *out, const FFTSample *input)
CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]);
}
ff_fft_calc(s, x);
s->fft_calc(s, x);
/* post rotation */
for(i=0;i<n8;i++) {

2
libavcodec/mpegaudio.h
View File

@ -33,7 +33,7 @@
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "dct.h"
#define CONFIG_AUDIO_NONSHORT 0

3
libavcodec/nellymoserdec.c
View File

@ -39,6 +39,7 @@
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
@ -121,7 +122,7 @@ static void nelly_decode_block(NellyMoserDecodeContext *s,
memset(&aptr[NELLY_FILL_LEN], 0,
(NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float));
ff_imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
/* XXX: overlapping and windowing should be part of a more
generic imdct function */
overlap_and_window(s, s->state, aptr, s->imdct_out);

5
libavcodec/nellymoserenc.c
View File

@ -39,6 +39,7 @@
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "sinewin.h"
#define BITSTREAM_WRITER_LE
#include "put_bits.h"
@ -116,13 +117,13 @@ static void apply_mdct(NellyMoserEncodeContext *s)
s->dsp.vector_fmul(s->in_buff, s->buf[s->bufsel], ff_sine_128, NELLY_BUF_LEN);
s->dsp.vector_fmul_reverse(s->in_buff + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN, ff_sine_128,
NELLY_BUF_LEN);
ff_mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
s->dsp.vector_fmul(s->buf[s->bufsel] + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN,
ff_sine_128, NELLY_BUF_LEN);
s->dsp.vector_fmul_reverse(s->buf[s->bufsel] + 2 * NELLY_BUF_LEN, s->buf[1 - s->bufsel], ff_sine_128,
NELLY_BUF_LEN);
ff_mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
}
static av_cold int encode_init(AVCodecContext *avctx)

2
libavcodec/qdm2.c
View File

@ -1588,7 +1588,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
/* add samples to output buffer */
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;

10
libavcodec/rdft.c
View File

@ -21,7 +21,7 @@
#include <stdlib.h>
#include <math.h>
#include "libavutil/mathematics.h"
#include "fft.h"
#include "rdft.h"
/**
* @file
@ -65,8 +65,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
const FFTSample *tsin = s->tsin;
if (!s->inverse) {
ff_fft_permute(&s->fft, (FFTComplex*)data);
ff_fft_calc(&s->fft, (FFTComplex*)data);
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
}
/* i=0 is a special case because of packing, the DC term is real, so we
are going to throw the N/2 term (also real) in with it. */
@ -91,8 +91,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
if (s->inverse) {
data[0] *= k1;
data[1] *= k1;
ff_fft_permute(&s->fft, (FFTComplex*)data);
ff_fft_calc(&s->fft, (FFTComplex*)data);
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
}
}

74
libavcodec/rdft.h Normal file
View File

@ -0,0 +1,74 @@
/*
* (I)RDFT transforms
* Copyright (c) 2009 Alex Converse <alex dot converse at gmail dot com>
*
* 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
*/
#ifndef AVCODEC_RDFT_H
#define AVCODEC_RDFT_H
#include "config.h"
#include "fft.h"
#if CONFIG_HARDCODED_TABLES
# define SINTABLE_CONST const
#else
# define SINTABLE_CONST
#endif
#define SINTABLE(size) \
SINTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_sin_##size)[size/2]
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);
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);
#endif

20
libavcodec/sinewin.c Normal file
View File

@ -0,0 +1,20 @@
/*
* 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 "sinewin.h"
#include "sinewin_tablegen.h"

59
libavcodec/sinewin.h Normal file
View File

@ -0,0 +1,59 @@
/*
* Copyright (c) 2008 Robert Swain
*
* 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
*/
#ifndef AVCODEC_SINEWIN_H
#define AVCODEC_SINEWIN_H
#include "config.h"
#include "libavutil/mem.h"
#if CONFIG_HARDCODED_TABLES
# define SINETABLE_CONST const
#else
# define SINETABLE_CONST
#endif
#define SINETABLE(size) \
SINETABLE_CONST DECLARE_ALIGNED(16, float, ff_sine_##size)[size]
/**
* 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];
#endif

4
libavcodec/mdct_tablegen.c → libavcodec/sinewin_tablegen.c
View File

@ -1,5 +1,5 @@
/*
* Generate a header file for hardcoded MDCT tables
* Generate a header file for hardcoded sine windows
*
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
@ -29,7 +29,7 @@
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
#include "mdct_tablegen.h"
#include "sinewin_tablegen.h"
#include "tableprint.h"
int main(void)

4
libavcodec/mdct_tablegen.h → libavcodec/sinewin_tablegen.h
View File

@ -1,5 +1,5 @@
/*
* Header file for hardcoded MDCT tables
* Header file for hardcoded sine windows
*
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
@ -36,7 +36,7 @@ SINETABLE(1024);
SINETABLE(2048);
SINETABLE(4096);
#else
#include "libavcodec/mdct_tables.h"
#include "libavcodec/sinewin_tables.h"
#endif
SINETABLE_CONST float * const ff_sine_windows[] = {

2
libavcodec/synth_filter.c
View File

@ -29,7 +29,7 @@ static void synth_filter_float(FFTContext *imdct,
float *synth_buf= synth_buf_ptr + *synth_buf_offset;
int i, j;
ff_imdct_half(imdct, synth_buf, in);
imdct->imdct_half(imdct, synth_buf, in);
for (i = 0; i < 16; i++){
float a= synth_buf2[i ];

4
libavcodec/twinvq.c
View File

@ -24,6 +24,7 @@
#include "dsputil.h"
#include "fft.h"
#include "lsp.h"
#include "sinewin.h"
#include <math.h>
#include <stdint.h>
@ -608,6 +609,7 @@ static void dec_lpc_spectrum_inv(TwinContext *tctx, float *lsp,
static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
float *in, float *prev, int ch)
{
FFTContext *mdct = &tctx->mdct_ctx[ftype];
const ModeTab *mtab = tctx->mtab;
int bsize = mtab->size / mtab->fmode[ftype].sub;
int size = mtab->size;
@ -640,7 +642,7 @@ static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
wsize = types_sizes[wtype_to_wsize[sub_wtype]];
ff_imdct_half(&tctx->mdct_ctx[ftype], buf1 + bsize*j, in + bsize*j);
mdct->imdct_half(mdct, buf1 + bsize*j, in + bsize*j);
tctx->dsp.vector_fmul_window(out2,
prev_buf + (bsize-wsize)/2,

6
libavcodec/vorbis_dec.c
View File

@ -1448,7 +1448,7 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
static int vorbis_parse_audio_packet(vorbis_context *vc)
{
GetBitContext *gb = &vc->gb;
FFTContext *mdct;
uint_fast8_t previous_window = vc->previous_window;
uint_fast8_t mode_number;
uint_fast8_t blockflag;
@ -1552,11 +1552,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
// Dotproduct, MDCT
mdct = &vc->mdct[blockflag];
for (j = vc->audio_channels-1;j >= 0; j--) {
ch_floor_ptr = vc->channel_floors + j * blocksize / 2;
ch_res_ptr = vc->channel_residues + res_chan[j] * blocksize / 2;
vc->dsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
ff_imdct_half(&vc->mdct[blockflag], ch_res_ptr, ch_floor_ptr);
mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
}
// Overlap/add, save data for next overlapping FPMATH

2
libavcodec/vorbis_enc.c
View File

@ -935,7 +935,7 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
}
for (channel = 0; channel < venc->channels; channel++)
ff_mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
venc->mdct[0].mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
venc->samples + channel * window_len * 2);
if (samples) {

1
libavcodec/wma.c
View File

@ -20,6 +20,7 @@
*/
#include "avcodec.h"
#include "sinewin.h"
#include "wma.h"
#include "wmadata.h"

5
libavcodec/wmadec.c
View File

@ -447,6 +447,7 @@ static int wma_decode_block(WMACodecContext *s)
int coef_nb_bits, total_gain;
int nb_coefs[MAX_CHANNELS];
float mdct_norm;
FFTContext *mdct;
#ifdef TRACE
tprintf(s->avctx, "***decode_block: %d:%d\n", s->frame_count - 1, s->block_num);
@ -742,12 +743,14 @@ static int wma_decode_block(WMACodecContext *s)
}
next:
mdct = &s->mdct_ctx[bsize];
for(ch = 0; ch < s->nb_channels; ch++) {
int n4, index;
n4 = s->block_len / 2;
if(s->channel_coded[ch]){
ff_imdct_calc(&s->mdct_ctx[bsize], s->output, s->coefs[ch]);
mdct->imdct_calc(mdct, s->output, s->coefs[ch]);
}else if(!(s->ms_stereo && ch==1))
memset(s->output, 0, sizeof(s->output));

3
libavcodec/wmaenc.c
View File

@ -77,6 +77,7 @@ static int encode_init(AVCodecContext * avctx){
static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
WMACodecContext *s = avctx->priv_data;
int window_index= s->frame_len_bits - s->block_len_bits;
FFTContext *mdct = &s->mdct_ctx[window_index];
int i, j, channel;
const float * win = s->windows[window_index];
int window_len = 1 << s->block_len_bits;
@ -89,7 +90,7 @@ static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * a
s->output[i+window_len] = audio[j] / n * win[window_len - i - 1];
s->frame_out[channel][i] = audio[j] / n * win[i];
}
ff_mdct_calc(&s->mdct_ctx[window_index], s->coefs[channel], s->output);
mdct->mdct_calc(mdct, s->coefs[channel], s->output);
}
}

7
libavcodec/wmaprodec.c
View File

@ -92,6 +92,7 @@
#include "put_bits.h"
#include "wmaprodata.h"
#include "dsputil.h"
#include "sinewin.h"
#include "wma.h"
/** current decoder limitations */
@ -1222,6 +1223,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
get_bits_count(&s->gb) - s->subframe_offset);
if (transmit_coeffs) {
FFTContext *mdct = &s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS];
/** reconstruct the per channel data */
inverse_channel_transform(s);
for (i = 0; i < s->channels_for_cur_subframe; i++) {
@ -1246,9 +1248,8 @@ static int decode_subframe(WMAProDecodeCtx *s)
quant, end - start);
}
/** apply imdct (ff_imdct_half == DCTIV with reverse) */
ff_imdct_half(&s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS],
s->channel[c].coeffs, s->tmp);
/** apply imdct (imdct_half == DCTIV with reverse) */
mdct->imdct_half(mdct, s->channel[c].coeffs, s->tmp);
}
}

19
libavcodec/wmavoice.c
View File

@ -36,8 +36,9 @@
#include "acelp_filters.h"
#include "lsp.h"
#include "libavutil/lzo.h"
#include "avfft.h"
#include "fft.h"
#include "dct.h"
#include "rdft.h"
#include "sinewin.h"
#define MAX_BLOCKS 8 ///< maximum number of blocks per frame
#define MAX_LSPS 16 ///< maximum filter order
@ -558,7 +559,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
int n, idx;
/* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
ff_rdft_calc(&s->rdft, lpcs);
s->rdft.rdft_calc(&s->rdft, lpcs);
#define log_range(var, assign) do { \
float tmp = log10f(assign); var = tmp; \
max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
@ -601,8 +602,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
* is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
* Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
* "moment" of the LPCs in this filter. */
ff_dct_calc(&s->dct, lpcs);
ff_dct_calc(&s->dst, lpcs);
s->dct.dct_calc(&s->dct, lpcs);
s->dst.dct_calc(&s->dst, lpcs);
/* Split out the coefficient indexes into phase/magnitude pairs */
idx = 255 + av_clip(lpcs[64], -255, 255);
@ -623,7 +624,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
coeffs[1] = last_coeff;
/* move into real domain */
ff_rdft_calc(&s->irdft, coeffs);
s->irdft.rdft_calc(&s->irdft, coeffs);
/* tilt correction and normalize scale */
memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
@ -693,8 +694,8 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
/* apply coefficients (in frequency spectrum domain), i.e. complex
* number multiplication */
memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
ff_rdft_calc(&s->rdft, synth_pf);
ff_rdft_calc(&s->rdft, coeffs);
s->rdft.rdft_calc(&s->rdft, synth_pf);
s->rdft.rdft_calc(&s->rdft, coeffs);
synth_pf[0] *= coeffs[0];
synth_pf[1] *= coeffs[1];
for (n = 1; n < 64; n++) {
@ -702,7 +703,7 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
}
ff_rdft_calc(&s->irdft, synth_pf);
s->irdft.rdft_calc(&s->irdft, synth_pf);
}
/* merge filter output with the history of previous runs */

1
libavcodec/x86/fft.c
View File

@ -18,6 +18,7 @@
#include "libavutil/cpu.h"
#include "libavcodec/dsputil.h"
#include "libavcodec/dct.h"
#include "fft.h"
av_cold void ff_fft_init_mmx(FFTContext *s)