/* * 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 "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);    }}