diff options
Diffstat (limited to 'libavcodec')
41 files changed, 406 insertions, 256 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index eefe60c772..8343d92bc6 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -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 diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 6317e429e2..1399eda126 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -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 diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 0ca390e72b..8fc8f70726 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -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]; diff --git a/libavcodec/aacsbr.c b/libavcodec/aacsbr.c index 237d51a391..117aa98875 100644 --- a/libavcodec/aacsbr.c +++ b/libavcodec/aacsbr.c @@ -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 ]; diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 4df0cf8a03..094b2615ff 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -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)); } diff --git a/libavcodec/ac3enc_float.c b/libavcodec/ac3enc_float.c index 4b13e4c723..e0783f9aa3 100644 --- a/libavcodec/ac3enc_float.c +++ b/libavcodec/ac3enc_float.c @@ -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); } diff --git a/libavcodec/arm/fft_init_arm.c b/libavcodec/arm/fft_init_arm.c index dff0689566..3e0e41ec3b 100644 --- a/libavcodec/arm/fft_init_arm.c +++ b/libavcodec/arm/fft_init_arm.c @@ -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); diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index be78445b8f..8d8267556a 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -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); } diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index 3d1f990164..3bf514c8b3 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -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); diff --git a/libavcodec/avfft.c b/libavcodec/avfft.c index 7d5d08390f..9e0ddaa627 100644 --- a/libavcodec/avfft.c +++ b/libavcodec/avfft.c @@ -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) diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index a00d6578e2..d879efc2e0 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -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, diff --git a/libavcodec/cook.c b/libavcodec/cook.c index da31e9f520..3f7776bed0 100644 --- a/libavcodec/cook.c +++ b/libavcodec/cook.c @@ -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); diff --git a/libavcodec/costablegen.c b/libavcodec/costablegen.c index bfcd6356a0..545fe7f998 100644 --- a/libavcodec/costablegen.c +++ b/libavcodec/costablegen.c @@ -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; diff --git a/libavcodec/dct.c b/libavcodec/dct.c index dab94c30a0..83cf1b4896 100644 --- a/libavcodec/dct.c +++ b/libavcodec/dct.c @@ -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; diff --git a/libavcodec/dct.h b/libavcodec/dct.h new file mode 100644 index 0000000000..44cf04d41d --- /dev/null +++ b/libavcodec/dct.h @@ -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 diff --git a/libavcodec/fft-test.c b/libavcodec/fft-test.c index 85282f531a..9ad8793c2b 100644 --- a/libavcodec/fft-test.c +++ b/libavcodec/fft-test.c @@ -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; } } diff --git a/libavcodec/fft.h b/libavcodec/fft.h index 58a7f30ad1..5a0f41ff6e 100644 --- a/libavcodec/fft.h +++ b/libavcodec/fft.h @@ -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 */ diff --git a/libavcodec/imc.c b/libavcodec/imc.c index 4cbb287a57..cb0c7102d8 100644 --- a/libavcodec/imc.c +++ b/libavcodec/imc.c @@ -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++){ diff --git a/libavcodec/kbdwin.c b/libavcodec/kbdwin.c new file mode 100644 index 0000000000..8b33861f71 --- /dev/null +++ b/libavcodec/kbdwin.c @@ -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); +} + diff --git a/libavcodec/kbdwin.h b/libavcodec/kbdwin.h new file mode 100644 index 0000000000..e762105628 --- /dev/null +++ b/libavcodec/kbdwin.h @@ -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 diff --git a/libavcodec/mdct.c b/libavcodec/mdct.c index 31e2216dd3..ca686b4ccb 100644 --- a/libavcodec/mdct.c +++ b/libavcodec/mdct.c @@ -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++) { diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 97c7855f06..4d78566463 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -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 diff --git a/libavcodec/nellymoserdec.c b/libavcodec/nellymoserdec.c index fd8568d5ab..5ad49abceb 100644 --- a/libavcodec/nellymoserdec.c +++ b/libavcodec/nellymoserdec.c @@ -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); diff --git a/libavcodec/nellymoserenc.c b/libavcodec/nellymoserenc.c index 9b792ff2ac..81a1d43e4e 100644 --- a/libavcodec/nellymoserenc.c +++ b/libavcodec/nellymoserenc.c @@ -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) diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 3779b8e3aa..0d5eeac0ce 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -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; diff --git a/libavcodec/rdft.c b/libavcodec/rdft.c index 59b4624716..ebddd8b56b 100644 --- a/libavcodec/rdft.c +++ b/libavcodec/rdft.c @@ -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); } } diff --git a/libavcodec/rdft.h b/libavcodec/rdft.h new file mode 100644 index 0000000000..cf50aba39c --- /dev/null +++ b/libavcodec/rdft.h @@ -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 diff --git a/libavcodec/sinewin.c b/libavcodec/sinewin.c new file mode 100644 index 0000000000..1fa0e953f0 --- /dev/null +++ b/libavcodec/sinewin.c @@ -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" diff --git a/libavcodec/sinewin.h b/libavcodec/sinewin.h new file mode 100644 index 0000000000..33f7d68b40 --- /dev/null +++ b/libavcodec/sinewin.h @@ -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 diff --git a/libavcodec/mdct_tablegen.c b/libavcodec/sinewin_tablegen.c index 6205f06e3c..48eb771e48 100644 --- a/libavcodec/mdct_tablegen.c +++ b/libavcodec/sinewin_tablegen.c @@ -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) diff --git a/libavcodec/mdct_tablegen.h b/libavcodec/sinewin_tablegen.h index 51a0094221..1c85d79f2f 100644 --- a/libavcodec/mdct_tablegen.h +++ b/libavcodec/sinewin_tablegen.h @@ -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[] = { diff --git a/libavcodec/synth_filter.c b/libavcodec/synth_filter.c index 4af496d372..5f10530974 100644 --- a/libavcodec/synth_filter.c +++ b/libavcodec/synth_filter.c @@ -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 ]; diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c index bd43104e6e..857ca5bae5 100644 --- a/libavcodec/twinvq.c +++ b/libavcodec/twinvq.c @@ -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, diff --git a/libavcodec/vorbis_dec.c b/libavcodec/vorbis_dec.c index deaae7576b..637f510df4 100644 --- a/libavcodec/vorbis_dec.c +++ b/libavcodec/vorbis_dec.c @@ -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 diff --git a/libavcodec/vorbis_enc.c b/libavcodec/vorbis_enc.c index 93683ac422..7dae4cb2b0 100644 --- a/libavcodec/vorbis_enc.c +++ b/libavcodec/vorbis_enc.c @@ -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) { diff --git a/libavcodec/wma.c b/libavcodec/wma.c index a7eacb8c78..67599b7eab 100644 --- a/libavcodec/wma.c +++ b/libavcodec/wma.c @@ -20,6 +20,7 @@ */ #include "avcodec.h" +#include "sinewin.h" #include "wma.h" #include "wmadata.h" diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c index 83f8dea8bb..5382999cb0 100644 --- a/libavcodec/wmadec.c +++ b/libavcodec/wmadec.c @@ -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)); diff --git a/libavcodec/wmaenc.c b/libavcodec/wmaenc.c index 4e54a70161..9112918382 100644 --- a/libavcodec/wmaenc.c +++ b/libavcodec/wmaenc.c @@ -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); } } diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index d248ae4ac0..b207e18b2f 100644 --- a/libavcodec/wmaprodec.c +++ b/libavcodec/wmaprodec.c @@ -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); } } diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 13011ec234..dfdfabbbb3 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -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 */ diff --git a/libavcodec/x86/fft.c b/libavcodec/x86/fft.c index 5ca341d5e4..37081d67fc 100644 --- a/libavcodec/x86/fft.c +++ b/libavcodec/x86/fft.c @@ -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) |