X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fac3enc_fixed.c;h=2bb82ef3b67c3842f96da661319616ec01694d03;hb=b7a4127a45b780d76e6b09427a3d0197c4bc1cdb;hp=3de00ee484646905111a54ad1de170905ae6688d;hpb=fbb6b49dabc3398440c6dfa838aa090a7a6ebc0d;p=ffmpeg diff --git a/libavcodec/ac3enc_fixed.c b/libavcodec/ac3enc_fixed.c index 3de00ee4846..2bb82ef3b67 100644 --- a/libavcodec/ac3enc_fixed.c +++ b/libavcodec/ac3enc_fixed.c @@ -4,20 +4,20 @@ * Copyright (c) 2006-2010 Justin Ruggles * Copyright (c) 2006-2010 Prakash Punnoor * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -26,405 +26,124 @@ * fixed-point AC-3 encoder. */ +#define FFT_FLOAT 0 #undef CONFIG_AC3ENC_FLOAT -#include "ac3enc.c" +#include "internal.h" +#include "audiodsp.h" +#include "ac3enc.h" +#include "eac3enc.h" +#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED +#include "ac3enc_opts_template.c" +static const AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name, + ac3_options, LIBAVUTIL_VERSION_INT }; -/** Scale a float value by 2^15, convert to an integer, and clip to range -32767..32767. */ -#define FIX15(a) av_clip(SCALE_FLOAT(a, 15), -32767, 32767) +#include "ac3enc_template.c" /** * Finalize MDCT and free allocated memory. + * + * @param s AC-3 encoder private context */ -static av_cold void mdct_end(AC3MDCTContext *mdct) -{ - mdct->nbits = 0; - av_freep(&mdct->costab); - av_freep(&mdct->sintab); - av_freep(&mdct->xcos1); - av_freep(&mdct->xsin1); - av_freep(&mdct->rot_tmp); - av_freep(&mdct->cplx_tmp); -} - - -/** - * Initialize FFT tables. - * @param ln log2(FFT size) - */ -static av_cold int fft_init(AVCodecContext *avctx, AC3MDCTContext *mdct, int ln) +av_cold void AC3_NAME(mdct_end)(AC3EncodeContext *s) { - int i, n, n2; - float alpha; - - n = 1 << ln; - n2 = n >> 1; - - FF_ALLOC_OR_GOTO(avctx, mdct->costab, n2 * sizeof(*mdct->costab), fft_alloc_fail); - FF_ALLOC_OR_GOTO(avctx, mdct->sintab, n2 * sizeof(*mdct->sintab), fft_alloc_fail); - - for (i = 0; i < n2; i++) { - alpha = 2.0 * M_PI * i / n; - mdct->costab[i] = FIX15(cos(alpha)); - mdct->sintab[i] = FIX15(sin(alpha)); - } - - return 0; -fft_alloc_fail: - mdct_end(mdct); - return AVERROR(ENOMEM); + ff_mdct_end(&s->mdct); } /** * Initialize MDCT tables. - * @param nbits log2(MDCT size) - */ -static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct, - int nbits) -{ - int i, n, n4, ret; - - n = 1 << nbits; - n4 = n >> 2; - - mdct->nbits = nbits; - - ret = fft_init(avctx, mdct, nbits - 2); - if (ret) - return ret; - - mdct->window = ff_ac3_window; - - FF_ALLOC_OR_GOTO(avctx, mdct->xcos1, n4 * sizeof(*mdct->xcos1), mdct_alloc_fail); - FF_ALLOC_OR_GOTO(avctx, mdct->xsin1, n4 * sizeof(*mdct->xsin1), mdct_alloc_fail); - FF_ALLOC_OR_GOTO(avctx, mdct->rot_tmp, n * sizeof(*mdct->rot_tmp), mdct_alloc_fail); - FF_ALLOC_OR_GOTO(avctx, mdct->cplx_tmp, n4 * sizeof(*mdct->cplx_tmp), mdct_alloc_fail); - - for (i = 0; i < n4; i++) { - float alpha = 2.0 * M_PI * (i + 1.0 / 8.0) / n; - mdct->xcos1[i] = FIX15(-cos(alpha)); - mdct->xsin1[i] = FIX15(-sin(alpha)); - } - - return 0; -mdct_alloc_fail: - mdct_end(mdct); - return AVERROR(ENOMEM); -} - - -/** Butterfly op */ -#define BF(pre, pim, qre, qim, pre1, pim1, qre1, qim1) \ -{ \ - int ax, ay, bx, by; \ - bx = pre1; \ - by = pim1; \ - ax = qre1; \ - ay = qim1; \ - pre = (bx + ax) >> 1; \ - pim = (by + ay) >> 1; \ - qre = (bx - ax) >> 1; \ - qim = (by - ay) >> 1; \ -} - - -/** Complex multiply */ -#define CMUL(pre, pim, are, aim, bre, bim) \ -{ \ - pre = (MUL16(are, bre) - MUL16(aim, bim)) >> 15; \ - pim = (MUL16(are, bim) + MUL16(bre, aim)) >> 15; \ -} - - -/** - * Calculate a 2^n point complex FFT on 2^ln points. - * @param z complex input/output samples - * @param ln log2(FFT size) - */ -static void fft(AC3MDCTContext *mdct, IComplex *z, int ln) -{ - int j, l, np, np2; - int nblocks, nloops; - register IComplex *p,*q; - int tmp_re, tmp_im; - - np = 1 << ln; - - /* reverse */ - for (j = 0; j < np; j++) { - int k = av_reverse[j] >> (8 - ln); - if (k < j) - FFSWAP(IComplex, z[k], z[j]); - } - - /* pass 0 */ - - p = &z[0]; - j = np >> 1; - do { - BF(p[0].re, p[0].im, p[1].re, p[1].im, - p[0].re, p[0].im, p[1].re, p[1].im); - p += 2; - } while (--j); - - /* pass 1 */ - - p = &z[0]; - j = np >> 2; - do { - BF(p[0].re, p[0].im, p[2].re, p[2].im, - p[0].re, p[0].im, p[2].re, p[2].im); - BF(p[1].re, p[1].im, p[3].re, p[3].im, - p[1].re, p[1].im, p[3].im, -p[3].re); - p+=4; - } while (--j); - - /* pass 2 .. ln-1 */ - - nblocks = np >> 3; - nloops = 1 << 2; - np2 = np >> 1; - do { - p = z; - q = z + nloops; - for (j = 0; j < nblocks; j++) { - BF(p->re, p->im, q->re, q->im, - p->re, p->im, q->re, q->im); - p++; - q++; - for(l = nblocks; l < np2; l += nblocks) { - CMUL(tmp_re, tmp_im, mdct->costab[l], -mdct->sintab[l], q->re, q->im); - BF(p->re, p->im, q->re, q->im, - p->re, p->im, tmp_re, tmp_im); - p++; - q++; - } - p += nloops; - q += nloops; - } - nblocks = nblocks >> 1; - nloops = nloops << 1; - } while (nblocks); -} - - -/** - * Calculate a 512-point MDCT - * @param out 256 output frequency coefficients - * @param in 512 windowed input audio samples - */ -static void mdct512(AC3MDCTContext *mdct, int32_t *out, int16_t *in) -{ - int i, re, im, n, n2, n4; - int16_t *rot = mdct->rot_tmp; - IComplex *x = mdct->cplx_tmp; - - n = 1 << mdct->nbits; - n2 = n >> 1; - n4 = n >> 2; - - /* shift to simplify computations */ - for (i = 0; i > 1; - im = -((int)rot[n2+2*i] - (int)rot[n2-1-2*i]) >> 1; - CMUL(x[i].re, x[i].im, re, im, -mdct->xcos1[i], mdct->xsin1[i]); - } - - fft(mdct, x, mdct->nbits - 2); - - /* post rotation */ - for (i = 0; i < n4; i++) { - re = x[i].re; - im = x[i].im; - CMUL(out[n2-1-2*i], out[2*i], re, im, mdct->xsin1[i], mdct->xcos1[i]); - } -} - - -/** - * Apply KBD window to input samples prior to MDCT. - */ -static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input, - const int16_t *window, int n) -{ - int i; - int n2 = n >> 1; - - for (i = 0; i < n2; i++) { - output[i] = MUL16(input[i], window[i]) >> 15; - output[n-i-1] = MUL16(input[n-i-1], window[i]) >> 15; - } -} - - -/** - * Calculate the log2() of the maximum absolute value in an array. - * @param tab input array - * @param n number of values in the array - * @return log2(max(abs(tab[]))) - */ -static int log2_tab(AC3EncodeContext *s, int16_t *src, int len) -{ - int v = s->ac3dsp.ac3_max_msb_abs_int16(src, len); - return av_log2(v); -} - - -/** - * Left-shift each value in an array by a specified amount. - * @param tab input array - * @param n number of values in the array - * @param lshift left shift amount + * + * @param s AC-3 encoder private context + * @return 0 on success, negative error code on failure */ -static void lshift_tab(int16_t *tab, int n, unsigned int lshift) +av_cold int AC3_NAME(mdct_init)(AC3EncodeContext *s) { - int i; - - if (lshift > 0) { - for (i = 0; i < n; i++) - tab[i] <<= lshift; - } + int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0); + s->mdct_window = ff_ac3_window; + return ret; } -/** +/* * Normalize the input samples to use the maximum available precision. - * This assumes signed 16-bit input samples. Exponents are reduced by 9 to - * match the 24-bit internal precision for MDCT coefficients. - * - * @return exponent shift + * This assumes signed 16-bit input samples. */ static int normalize_samples(AC3EncodeContext *s) { - int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE); - lshift_tab(s->windowed_samples, AC3_WINDOW_SIZE, v); - return v - 9; + int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE); + v = 14 - av_log2(v); + if (v > 0) + s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v); + /* +6 to right-shift from 31-bit to 25-bit */ + return v + 6; } -/** - * Scale MDCT coefficients from float to fixed-point. +/* + * Scale MDCT coefficients to 25-bit signed fixed-point. */ static void scale_coefficients(AC3EncodeContext *s) { - /* scaling/conversion is obviously not needed for the fixed-point encoder - since the coefficients are already fixed-point. */ - return; -} - - -#ifdef TEST -/*************************************************************************/ -/* TEST */ - -#include "libavutil/lfg.h" - -#define MDCT_NBITS 9 -#define MDCT_SAMPLES (1 << MDCT_NBITS) -#define FN (MDCT_SAMPLES/4) + int blk, ch; - -static void fft_test(AC3MDCTContext *mdct, AVLFG *lfg) -{ - IComplex in[FN], in1[FN]; - int k, n, i; - float sum_re, sum_im, a; - - for (i = 0; i < FN; i++) { - in[i].re = av_lfg_get(lfg) % 65535 - 32767; - in[i].im = av_lfg_get(lfg) % 65535 - 32767; - in1[i] = in[i]; - } - fft(mdct, in, 7); - - /* do it by hand */ - for (k = 0; k < FN; k++) { - sum_re = 0; - sum_im = 0; - for (n = 0; n < FN; n++) { - a = -2 * M_PI * (n * k) / FN; - sum_re += in1[n].re * cos(a) - in1[n].im * sin(a); - sum_im += in1[n].re * sin(a) + in1[n].im * cos(a); + for (blk = 0; blk < s->num_blocks; blk++) { + AC3Block *block = &s->blocks[blk]; + for (ch = 1; ch <= s->channels; ch++) { + s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS, + block->coeff_shift[ch]); } - av_log(NULL, AV_LOG_DEBUG, "%3d: %6d,%6d %6.0f,%6.0f\n", - k, in[k].re, in[k].im, sum_re / FN, sum_im / FN); } } -static void mdct_test(AC3MDCTContext *mdct, AVLFG *lfg) +/* + * Clip MDCT coefficients to allowable range. + */ +static void clip_coefficients(AudioDSPContext *adsp, int32_t *coef, + unsigned int len) { - int16_t input[MDCT_SAMPLES]; - int32_t output[AC3_MAX_COEFS]; - float input1[MDCT_SAMPLES]; - float output1[AC3_MAX_COEFS]; - float s, a, err, e, emax; - int i, k, n; - - for (i = 0; i < MDCT_SAMPLES; i++) { - input[i] = (av_lfg_get(lfg) % 65535 - 32767) * 9 / 10; - input1[i] = input[i]; - } - - mdct512(mdct, output, input); + adsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len); +} - /* do it by hand */ - for (k = 0; k < AC3_MAX_COEFS; k++) { - s = 0; - for (n = 0; n < MDCT_SAMPLES; n++) { - a = (2*M_PI*(2*n+1+MDCT_SAMPLES/2)*(2*k+1) / (4 * MDCT_SAMPLES)); - s += input1[n] * cos(a); - } - output1[k] = -2 * s / MDCT_SAMPLES; - } - err = 0; - emax = 0; - for (i = 0; i < AC3_MAX_COEFS; i++) { - av_log(NULL, AV_LOG_DEBUG, "%3d: %7d %7.0f\n", i, output[i], output1[i]); - e = output[i] - output1[i]; - if (e > emax) - emax = e; - err += e * e; +/* + * Calculate a single coupling coordinate. + */ +static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl) +{ + if (energy_cpl <= COEF_MAX) { + return 1048576; + } else { + uint64_t coord = energy_ch / (energy_cpl >> 24); + uint32_t coord32 = FFMIN(coord, 1073741824); + coord32 = ff_sqrt(coord32) << 9; + return FFMIN(coord32, COEF_MAX); } - av_log(NULL, AV_LOG_DEBUG, "err2=%f emax=%f\n", err / AC3_MAX_COEFS, emax); } -int main(void) +static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx) { - AVLFG lfg; - AC3MDCTContext mdct; - - mdct.avctx = NULL; - av_log_set_level(AV_LOG_DEBUG); - mdct_init(&mdct, 9); - - fft_test(&mdct, &lfg); - mdct_test(&mdct, &lfg); - - return 0; + AC3EncodeContext *s = avctx->priv_data; + s->fixed_point = 1; + return ff_ac3_encode_init(avctx); } -#endif /* TEST */ AVCodec ff_ac3_fixed_encoder = { - "ac3_fixed", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_AC3, - sizeof(AC3EncodeContext), - ac3_encode_init, - ac3_encode_frame, - ac3_encode_close, - NULL, - .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, - .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), - .channel_layouts = ac3_channel_layouts, + .name = "ac3_fixed", + .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_AC3, + .priv_data_size = sizeof(AC3EncodeContext), + .init = ac3_fixed_encode_init, + .encode2 = ff_ac3_fixed_encode_frame, + .close = ff_ac3_encode_close, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_NONE }, + .priv_class = &ac3enc_class, + .channel_layouts = ff_ac3_channel_layouts, + .defaults = ac3_defaults, };