/* * Copyright (c) 2017 Paul B Mahol * * 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 "libavutil/tx.h" #include "avfilter.h" #include "formats.h" #include "internal.h" #include "audio.h" #undef ctype #undef ftype #undef SQRT #undef HYPOT #undef SAMPLE_FORMAT #undef TX_TYPE #if DEPTH == 32 #define SAMPLE_FORMAT float #define SQRT sqrtf #define HYPOT hypotf #define ctype AVComplexFloat #define ftype float #define TX_TYPE AV_TX_FLOAT_RDFT #else #define SAMPLE_FORMAT double #define SQRT sqrt #define HYPOT hypot #define ctype AVComplexDouble #define ftype double #define TX_TYPE AV_TX_DOUBLE_RDFT #endif #define fn3(a,b) a##_##b #define fn2(a,b) fn3(a,b) #define fn(a) fn2(a, SAMPLE_FORMAT) static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out) { AudioFIRContext *s = ctx->priv; ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN; ftype min_delay = FLT_MAX, max_delay = FLT_MIN; int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1; char text[32]; int channel, i, x; for (int y = 0; y < s->h; y++) memset(out->data[0] + y * out->linesize[0], 0, s->w * 4); phase = av_malloc_array(s->w, sizeof(*phase)); mag = av_malloc_array(s->w, sizeof(*mag)); delay = av_malloc_array(s->w, sizeof(*delay)); if (!mag || !phase || !delay) goto end; channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1); for (i = 0; i < s->w; i++) { const ftype *src = (const ftype *)s->ir[s->selir]->extended_data[channel]; double w = i * M_PI / (s->w - 1); double div, real_num = 0., imag_num = 0., real = 0., imag = 0.; for (x = 0; x < s->nb_taps[s->selir]; x++) { real += cos(-x * w) * src[x]; imag += sin(-x * w) * src[x]; real_num += cos(-x * w) * src[x] * x; imag_num += sin(-x * w) * src[x] * x; } mag[i] = hypot(real, imag); phase[i] = atan2(imag, real); div = real * real + imag * imag; delay[i] = (real_num * real + imag_num * imag) / div; min = fminf(min, mag[i]); max = fmaxf(max, mag[i]); min_delay = fminf(min_delay, delay[i]); max_delay = fmaxf(max_delay, delay[i]); } for (i = 0; i < s->w; i++) { int ymag = mag[i] / max * (s->h - 1); int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1); int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1); ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1); yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1); ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1); if (prev_ymag < 0) prev_ymag = ymag; if (prev_yphase < 0) prev_yphase = yphase; if (prev_ydelay < 0) prev_ydelay = ydelay; draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF); draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00); draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF); prev_ymag = ymag; prev_yphase = yphase; prev_ydelay = ydelay; } if (s->w > 400 && s->h > 100) { drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD); snprintf(text, sizeof(text), "%.2f", max); drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD); drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD); snprintf(text, sizeof(text), "%.2f", min); drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD); drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD); snprintf(text, sizeof(text), "%.2f", max_delay); drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD); drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD); snprintf(text, sizeof(text), "%.2f", min_delay); drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD); } end: av_free(delay); av_free(phase); av_free(mag); } static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps, int ch, ftype *time) { ftype ch_gain = 1; switch (s->gtype) { case -1: ch_gain = 1; break; case 0: { ftype sum = 0; for (int i = 0; i < cur_nb_taps; i++) sum += FFABS(time[i]); ch_gain = 1. / sum; } break; case 1: { ftype sum = 0; for (int i = 0; i < cur_nb_taps; i++) sum += time[i]; ch_gain = 1. / sum; } break; case 2: { ftype sum = 0; for (int i = 0; i < cur_nb_taps; i++) sum += time[i] * time[i]; ch_gain = 1. / SQRT(sum); } break; case 3: case 4: { ftype *inc, *outc, scale, power; AVTXContext *tx; av_tx_fn tx_fn; int ret, size; size = 1 << av_ceil_log2_c(cur_nb_taps); inc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT)); outc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT)); if (!inc || !outc) { av_free(outc); av_free(inc); break; } scale = 1.; ret = av_tx_init(&tx, &tx_fn, TX_TYPE, 0, size, &scale, 0); if (ret < 0) { av_free(outc); av_free(inc); break; } { memcpy(inc, time, cur_nb_taps * sizeof(SAMPLE_FORMAT)); tx_fn(tx, outc, inc, sizeof(SAMPLE_FORMAT)); power = 0; if (s->gtype == 3) { for (int i = 0; i < size / 2 + 1; i++) power = FFMAX(power, HYPOT(outc[i * 2], outc[i * 2 + 1])); } else { ftype sum = 0; for (int i = 0; i < size / 2 + 1; i++) sum += HYPOT(outc[i * 2], outc[i * 2 + 1]); power = SQRT(sum / (size / 2 + 1)); } ch_gain = 1. / power; } av_tx_uninit(&tx); av_free(outc); av_free(inc); } break; default: return AVERROR_BUG; } if (ch_gain != 1. || s->ir_gain != 1.) { ftype gain = ch_gain * s->ir_gain; av_log(ctx, AV_LOG_DEBUG, "ch%d gain %f\n", ch, gain); #if DEPTH == 32 s->fdsp->vector_fmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 4)); #else s->fdsp->vector_dmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 8)); #endif } return 0; } static void fn(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch, AudioFIRSegment *seg, int coeff_partition, int selir) { const int coffset = coeff_partition * seg->coeff_size; const int nb_taps = s->nb_taps[selir]; ftype *time = (ftype *)s->norm_ir[selir]->extended_data[ch]; ftype *tempin = (ftype *)seg->tempin->extended_data[ch]; ftype *tempout = (ftype *)seg->tempout->extended_data[ch]; ctype *coeff = (ctype *)seg->coeff->extended_data[ch]; const int remaining = nb_taps - (seg->input_offset + coeff_partition * seg->part_size); const int size = remaining >= seg->part_size ? seg->part_size : remaining; memset(tempin + size, 0, sizeof(*tempin) * (seg->block_size - size)); memcpy(tempin, time + seg->input_offset + coeff_partition * seg->part_size, size * sizeof(*tempin)); seg->ctx_fn(seg->ctx[ch], tempout, tempin, sizeof(*tempin)); memcpy(coeff + coffset, tempout, seg->coeff_size * sizeof(*coeff)); av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch); av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions); av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size); av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size); av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length); av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size); av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size); av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset); } static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples) { if ((nb_samples & 15) == 0 && nb_samples >= 8) { #if DEPTH == 32 s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples); #else s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples); #endif } else { for (int n = 0; n < nb_samples; n++) dst[n] += src[n]; } } static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int ioffset, int offset, int selir) { AudioFIRContext *s = ctx->priv; const ftype *in = (const ftype *)s->in->extended_data[ch] + ioffset; ftype *blockout, *ptr = (ftype *)out->extended_data[ch] + offset; const int min_part_size = s->min_part_size; const int nb_samples = FFMIN(min_part_size, out->nb_samples - offset); const int nb_segments = s->nb_segments[selir]; const float dry_gain = s->dry_gain; const float wet_gain = s->wet_gain; for (int segment = 0; segment < nb_segments; segment++) { AudioFIRSegment *seg = &s->seg[selir][segment]; ftype *src = (ftype *)seg->input->extended_data[ch]; ftype *dst = (ftype *)seg->output->extended_data[ch]; ftype *sumin = (ftype *)seg->sumin->extended_data[ch]; ftype *sumout = (ftype *)seg->sumout->extended_data[ch]; ftype *tempin = (ftype *)seg->tempin->extended_data[ch]; ftype *buf = (ftype *)seg->buffer->extended_data[ch]; int *output_offset = &seg->output_offset[ch]; const int nb_partitions = seg->nb_partitions; const int input_offset = seg->input_offset; const int part_size = seg->part_size; int j; seg->part_index[ch] = seg->part_index[ch] % nb_partitions; if (dry_gain == 1.f) { memcpy(src + input_offset, in, nb_samples * sizeof(*src)); } else if (min_part_size >= 8) { #if DEPTH == 32 s->fdsp->vector_fmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 4)); #else s->fdsp->vector_dmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 8)); #endif emms_c(); } else { ftype *src2 = src + input_offset; for (int n = 0; n < nb_samples; n++) src2[n] = in[n] * dry_gain; } output_offset[0] += min_part_size; if (output_offset[0] >= part_size) { output_offset[0] = 0; } else { memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src)); dst += output_offset[0]; fn(fir_fadd)(s, ptr, dst, nb_samples); continue; } memset(sumin, 0, sizeof(*sumin) * seg->fft_length); blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size; memset(tempin + part_size, 0, sizeof(*tempin) * (seg->block_size - part_size)); memcpy(tempin, src, sizeof(*src) * part_size); seg->tx_fn(seg->tx[ch], blockout, tempin, sizeof(ftype)); j = seg->part_index[ch]; for (int i = 0; i < nb_partitions; i++) { const int input_partition = j; const int coeff_partition = i; const int coffset = coeff_partition * seg->coeff_size; const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + input_partition * seg->block_size; const ctype *coeff = ((const ctype *)seg->coeff->extended_data[ch]) + coffset; if (j == 0) j = nb_partitions; j--; #if DEPTH == 32 s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, part_size); #else s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, part_size); #endif } seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ctype)); fn(fir_fadd)(s, buf, sumout, part_size); memcpy(dst, buf, part_size * sizeof(*dst)); memcpy(buf, sumout + part_size, part_size * sizeof(*buf)); fn(fir_fadd)(s, ptr, dst, nb_samples); if (part_size != min_part_size) memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src)); seg->part_index[ch] = (seg->part_index[ch] + 1) % nb_partitions; } if (wet_gain == 1.f) return 0; if (min_part_size >= 8) { #if DEPTH == 32 s->fdsp->vector_fmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 4)); #else s->fdsp->vector_dmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 8)); #endif emms_c(); } else { for (int n = 0; n < nb_samples; n++) ptr[n] *= wet_gain; } return 0; }