* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
*
- * 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
*/
* fixed-point AC-3 encoder.
*/
+#define CONFIG_FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
-#include "ac3enc.c"
+#include "internal.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)
+av_cold void AC3_NAME(mdct_end)(AC3EncodeContext *s)
{
- 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)
-{
- 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)
+ *
+ * @param s AC-3 encoder private context
+ * @return 0 on success, negative error code on failure
*/
-static void fft(AC3MDCTContext *mdct, IComplex *z, int ln)
+av_cold int AC3_NAME(mdct_init)(AC3EncodeContext *s)
{
- 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);
+ int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0);
+ s->mdct_window = ff_ac3_window;
+ return ret;
}
-/**
- * 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 <n4; i++)
- rot[i] = -in[i + 3*n4];
- memcpy(&rot[n4], &in[0], 3*n4*sizeof(*in));
-
- /* pre rotation */
- for (i = 0; i < n4; i++) {
- re = ((int)rot[ 2*i] - (int)rot[ n-1-2*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(int16_t *output, const int16_t *input,
- const int16_t *window, int n)
+static void apply_window(void *dsp, int16_t *output, const int16_t *input,
+ const int16_t *window, unsigned int len)
{
- 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;
- }
+ DSPContext *dsp0 = dsp;
+ dsp0->apply_window_int16(output, input, window, len);
}
-/**
- * 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[])))
+/*
+ * Normalize the input samples to use the maximum available precision.
+ * This assumes signed 16-bit input samples.
*/
-static int log2_tab(int16_t *tab, int n)
+static int normalize_samples(AC3EncodeContext *s)
{
- int i, v;
-
- v = 0;
- for (i = 0; i < n; i++)
- v |= abs(tab[i]);
-
- return av_log2(v);
+ 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;
}
-/**
- * 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. a negative value means right shift.
+/*
+ * Scale MDCT coefficients to 25-bit signed fixed-point.
*/
-static void lshift_tab(int16_t *tab, int n, int lshift)
+static void scale_coefficients(AC3EncodeContext *s)
{
- int i;
+ int blk, ch;
- if (lshift > 0) {
- for (i = 0; i < n; i++)
- tab[i] <<= lshift;
- } else if (lshift < 0) {
- lshift = -lshift;
- for (i = 0; i < n; i++)
- tab[i] >>= lshift;
+ 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]);
+ }
}
}
-/**
- * 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
+/*
+ * Clip MDCT coefficients to allowable range.
*/
-static int normalize_samples(AC3EncodeContext *s)
+static void clip_coefficients(DSPContext *dsp, int32_t *coef, unsigned int len)
{
- int v = 14 - log2_tab(s->windowed_samples, AC3_WINDOW_SIZE);
- v = FFMAX(0, v);
- lshift_tab(s->windowed_samples, AC3_WINDOW_SIZE, v);
- return v - 9;
+ dsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len);
}
-#ifdef TEST
-/*************************************************************************/
-/* TEST */
-
-#include "libavutil/lfg.h"
-
-#define MDCT_NBITS 9
-#define MDCT_SAMPLES (1 << MDCT_NBITS)
-#define FN (MDCT_SAMPLES/4)
-
-
-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);
- }
- 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)
+/*
+ * Calculate a single coupling coordinate.
+ */
+static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
{
- 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);
-
- /* 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;
+ 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 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,
+AVCodec ff_ac3_fixed_encoder = {
+ .name = "ac3_fixed",
+ .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 },
+ .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+ .priv_class = &ac3enc_class,
+ .channel_layouts = ff_ac3_channel_layouts,
+ .defaults = ac3_defaults,
};
projects / ffmpeg / blobdiff