| /* |
| * 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 "internal.h" |
| #include "audio.h" |
| |
| #undef ctype |
| #undef ftype |
| #undef SQRT |
| #undef HYPOT |
| #undef SAMPLE_FORMAT |
| #undef TX_TYPE |
| #undef FABS |
| #undef POW |
| #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 |
| #define FABS fabsf |
| #define POW powf |
| #else |
| #define SAMPLE_FORMAT double |
| #define SQRT sqrt |
| #define HYPOT hypot |
| #define ctype AVComplexDouble |
| #define ftype double |
| #define TX_TYPE AV_TX_DOUBLE_RDFT |
| #define FABS fabs |
| #define POW pow |
| #endif |
| |
| #define fn3(a,b) a##_##b |
| #define fn2(a,b) fn3(a,b) |
| #define fn(a) fn2(a, SAMPLE_FORMAT) |
| |
| static ftype fn(ir_gain)(AVFilterContext *ctx, AudioFIRContext *s, |
| int cur_nb_taps, const ftype *time) |
| { |
| ftype ch_gain, sum = 0; |
| |
| if (s->ir_norm < 0.f) { |
| ch_gain = 1; |
| } else if (s->ir_norm == 0.f) { |
| for (int i = 0; i < cur_nb_taps; i++) |
| sum += time[i]; |
| ch_gain = 1. / sum; |
| } else { |
| ftype ir_norm = s->ir_norm; |
| |
| for (int i = 0; i < cur_nb_taps; i++) |
| sum += POW(FABS(time[i]), ir_norm); |
| ch_gain = 1. / POW(sum, 1. / ir_norm); |
| } |
| |
| return ch_gain; |
| } |
| |
| static void fn(ir_scale)(AVFilterContext *ctx, AudioFIRContext *s, |
| int cur_nb_taps, int ch, |
| ftype *time, ftype ch_gain) |
| { |
| 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 |
| } |
| } |
| |
| 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 |
| } 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 |
| } else { |
| for (int n = 0; n < nb_samples; n++) |
| ptr[n] *= wet_gain; |
| } |
| |
| return 0; |
| } |
| |
| static void fn(fir_quantums)(AVFilterContext *ctx, AudioFIRContext *s, AVFrame *out, |
| int min_part_size, int ch, int offset, |
| int prev_selir, int selir) |
| { |
| if (ctx->is_disabled || s->prev_is_disabled) { |
| const ftype *in = (const ftype *)s->in->extended_data[ch] + offset; |
| const ftype *xfade0 = (const ftype *)s->xfade[0]->extended_data[ch]; |
| const ftype *xfade1 = (const ftype *)s->xfade[1]->extended_data[ch]; |
| ftype *src0 = (ftype *)s->fadein[0]->extended_data[ch]; |
| ftype *src1 = (ftype *)s->fadein[1]->extended_data[ch]; |
| ftype *dst = ((ftype *)out->extended_data[ch]) + offset; |
| |
| if (ctx->is_disabled && !s->prev_is_disabled) { |
| memset(src0, 0, min_part_size * sizeof(ftype)); |
| fn(fir_quantum)(ctx, s->fadein[0], ch, offset, 0, selir); |
| for (int n = 0; n < min_part_size; n++) |
| dst[n] = xfade1[n] * src0[n] + xfade0[n] * in[n]; |
| } else if (!ctx->is_disabled && s->prev_is_disabled) { |
| memset(src1, 0, min_part_size * sizeof(ftype)); |
| fn(fir_quantum)(ctx, s->fadein[1], ch, offset, 0, selir); |
| for (int n = 0; n < min_part_size; n++) |
| dst[n] = xfade1[n] * in[n] + xfade0[n] * src1[n]; |
| } else { |
| memcpy(dst, in, sizeof(ftype) * min_part_size); |
| } |
| } else if (prev_selir != selir && s->loading[ch] != 0) { |
| const ftype *xfade0 = (const ftype *)s->xfade[0]->extended_data[ch]; |
| const ftype *xfade1 = (const ftype *)s->xfade[1]->extended_data[ch]; |
| ftype *src0 = (ftype *)s->fadein[0]->extended_data[ch]; |
| ftype *src1 = (ftype *)s->fadein[1]->extended_data[ch]; |
| ftype *dst = ((ftype *)out->extended_data[ch]) + offset; |
| |
| memset(src0, 0, min_part_size * sizeof(ftype)); |
| memset(src1, 0, min_part_size * sizeof(ftype)); |
| |
| fn(fir_quantum)(ctx, s->fadein[0], ch, offset, 0, prev_selir); |
| fn(fir_quantum)(ctx, s->fadein[1], ch, offset, 0, selir); |
| |
| if (s->loading[ch] > s->max_offset[selir]) { |
| for (int n = 0; n < min_part_size; n++) |
| dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n]; |
| s->loading[ch] = 0; |
| } else { |
| memcpy(dst, src0, min_part_size * sizeof(ftype)); |
| } |
| } else { |
| fn(fir_quantum)(ctx, out, ch, offset, offset, selir); |
| } |
| } |