mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-12-26 21:47:17 +00:00
557 lines
16 KiB
C
557 lines
16 KiB
C
/*
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* Copyright (c) 2017 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* An arbitrary audio FIR filter
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "libavcodec/avfft.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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#include "af_afir.h"
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static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
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{
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int n;
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for (n = 0; n < len; n++) {
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const float cre = c[2 * n ];
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const float cim = c[2 * n + 1];
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const float tre = t[2 * n ];
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const float tim = t[2 * n + 1];
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sum[2 * n ] += tre * cre - tim * cim;
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sum[2 * n + 1] += tre * cim + tim * cre;
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}
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sum[2 * n] += t[2 * n] * c[2 * n];
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}
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static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
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{
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AudioFIRContext *s = ctx->priv;
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const float *src = (const float *)s->in[0]->extended_data[ch];
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int index1 = (s->index + 1) % 3;
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int index2 = (s->index + 2) % 3;
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float *sum = s->sum[ch];
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AVFrame *out = arg;
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float *block;
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float *dst;
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int n, i, j;
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memset(sum, 0, sizeof(*sum) * s->fft_length);
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block = s->block[ch] + s->part_index * s->block_size;
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memset(block, 0, sizeof(*block) * s->fft_length);
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s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));
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emms_c();
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av_rdft_calc(s->rdft[ch], block);
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block[2 * s->part_size] = block[1];
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block[1] = 0;
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j = s->part_index;
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for (i = 0; i < s->nb_partitions; i++) {
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const int coffset = i * s->coeff_size;
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const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
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block = s->block[ch] + j * s->block_size;
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s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
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if (j == 0)
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j = s->nb_partitions;
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j--;
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}
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sum[1] = sum[2 * s->part_size];
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av_rdft_calc(s->irdft[ch], sum);
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dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
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for (n = 0; n < s->part_size; n++) {
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dst[n] += sum[n];
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}
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dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
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memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
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dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
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if (out) {
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float *ptr = (float *)out->extended_data[ch];
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s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
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emms_c();
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}
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return 0;
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}
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static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AVFrame *out = NULL;
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int ret;
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s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
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if (!s->want_skip) {
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out = ff_get_audio_buffer(outlink, s->nb_samples);
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if (!out)
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return AVERROR(ENOMEM);
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}
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s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
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if (!s->in[0]) {
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av_frame_free(&out);
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return AVERROR(ENOMEM);
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}
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av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
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ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
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s->part_index = (s->part_index + 1) % s->nb_partitions;
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av_audio_fifo_drain(s->fifo[0], s->nb_samples);
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if (!s->want_skip) {
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out->pts = s->pts;
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if (s->pts != AV_NOPTS_VALUE)
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s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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}
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s->index++;
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if (s->index == 3)
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s->index = 0;
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av_frame_free(&s->in[0]);
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if (s->want_skip == 1) {
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s->want_skip = 0;
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ret = 0;
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} else {
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ret = ff_filter_frame(outlink, out);
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}
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return ret;
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}
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static int convert_coeffs(AVFilterContext *ctx)
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{
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AudioFIRContext *s = ctx->priv;
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int i, ch, n, N;
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s->nb_taps = av_audio_fifo_size(s->fifo[1]);
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if (s->nb_taps <= 0)
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return AVERROR(EINVAL);
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for (n = 4; (1 << n) < s->nb_taps; n++);
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N = FFMIN(n, 16);
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s->ir_length = 1 << n;
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s->fft_length = (1 << (N + 1)) + 1;
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s->part_size = 1 << (N - 1);
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s->block_size = FFALIGN(s->fft_length, 32);
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s->coeff_size = FFALIGN(s->part_size + 1, 32);
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s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
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s->nb_coeffs = s->ir_length + s->nb_partitions;
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for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
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s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
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if (!s->sum[ch])
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return AVERROR(ENOMEM);
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}
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for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
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s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
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if (!s->coeff[ch])
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return AVERROR(ENOMEM);
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}
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for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
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s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
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if (!s->block[ch])
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return AVERROR(ENOMEM);
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}
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for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
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s->rdft[ch] = av_rdft_init(N, DFT_R2C);
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s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
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if (!s->rdft[ch] || !s->irdft[ch])
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return AVERROR(ENOMEM);
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}
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s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
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if (!s->in[1])
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return AVERROR(ENOMEM);
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s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
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if (!s->buffer)
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return AVERROR(ENOMEM);
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av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
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if (s->again) {
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float power = 0;
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for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
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float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
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for (i = 0; i < s->nb_taps; i++)
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power += FFABS(time[i]);
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}
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s->gain = sqrtf(1.f / (ctx->inputs[1]->channels * power)) / (sqrtf(ctx->inputs[1]->channels));
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for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
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float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
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s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
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}
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}
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for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
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float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
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float *block = s->block[ch];
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FFTComplex *coeff = s->coeff[ch];
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for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
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time[i] = 0;
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for (i = 0; i < s->nb_partitions; i++) {
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const float scale = 1.f / s->part_size;
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const int toffset = i * s->part_size;
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const int coffset = i * s->coeff_size;
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const int boffset = s->part_size;
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const int remaining = s->nb_taps - (i * s->part_size);
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const int size = remaining >= s->part_size ? s->part_size : remaining;
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memset(block, 0, sizeof(*block) * s->fft_length);
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memcpy(block + boffset, time + toffset, size * sizeof(*block));
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av_rdft_calc(s->rdft[0], block);
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coeff[coffset].re = block[0] * scale;
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coeff[coffset].im = 0;
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for (n = 1; n < s->part_size; n++) {
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coeff[coffset + n].re = block[2 * n] * scale;
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coeff[coffset + n].im = block[2 * n + 1] * scale;
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}
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coeff[coffset + s->part_size].re = block[1] * scale;
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coeff[coffset + s->part_size].im = 0;
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}
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}
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av_frame_free(&s->in[1]);
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av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
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av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
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av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
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av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
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s->have_coeffs = 1;
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return 0;
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}
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static int read_ir(AVFilterLink *link, AVFrame *frame)
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{
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AVFilterContext *ctx = link->dst;
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AudioFIRContext *s = ctx->priv;
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int nb_taps, max_nb_taps, ret;
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ret = av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
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frame->nb_samples);
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av_frame_free(&frame);
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if (ret < 0)
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return ret;
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nb_taps = av_audio_fifo_size(s->fifo[1]);
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max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
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if (nb_taps > max_nb_taps) {
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av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
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return AVERROR(EINVAL);
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *link, AVFrame *frame)
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{
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AVFilterContext *ctx = link->dst;
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AudioFIRContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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int ret;
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ret = av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
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frame->nb_samples);
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if (ret > 0 && s->pts == AV_NOPTS_VALUE)
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s->pts = frame->pts;
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av_frame_free(&frame);
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if (ret < 0)
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return ret;
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if (!s->have_coeffs && s->eof_coeffs) {
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ret = convert_coeffs(ctx);
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if (ret < 0)
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return ret;
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}
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if (s->have_coeffs) {
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while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
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ret = fir_frame(s, outlink);
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if (ret < 0)
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return ret;
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}
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}
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return 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioFIRContext *s = ctx->priv;
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int ret;
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if (!s->eof_coeffs) {
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ret = ff_request_frame(ctx->inputs[1]);
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if (ret == AVERROR_EOF) {
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s->eof_coeffs = 1;
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ret = 0;
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}
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return ret;
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}
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ret = ff_request_frame(ctx->inputs[0]);
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if (ret == AVERROR_EOF && s->have_coeffs) {
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if (s->need_padding) {
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AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
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if (!silence)
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return AVERROR(ENOMEM);
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ret = av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
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silence->nb_samples);
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av_frame_free(&silence);
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if (ret < 0)
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return ret;
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s->need_padding = 0;
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}
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while (av_audio_fifo_size(s->fifo[0]) > 0) {
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ret = fir_frame(s, outlink);
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if (ret < 0)
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return ret;
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}
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ret = AVERROR_EOF;
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}
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return ret;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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int ret, i;
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layouts = ff_all_channel_counts();
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if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
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return ret;
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for (i = 0; i < 2; i++) {
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layouts = ff_all_channel_counts();
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if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
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return ret;
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}
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formats = ff_make_format_list(sample_fmts);
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if ((ret = ff_set_common_formats(ctx, formats)) < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioFIRContext *s = ctx->priv;
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if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
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ctx->inputs[1]->channels != 1) {
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av_log(ctx, AV_LOG_ERROR,
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"Second input must have same number of channels as first input or "
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"exactly 1 channel.\n");
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return AVERROR(EINVAL);
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}
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s->one2many = ctx->inputs[1]->channels == 1;
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outlink->sample_rate = ctx->inputs[0]->sample_rate;
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outlink->time_base = ctx->inputs[0]->time_base;
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outlink->channel_layout = ctx->inputs[0]->channel_layout;
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outlink->channels = ctx->inputs[0]->channels;
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s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
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s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
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if (!s->fifo[0] || !s->fifo[1])
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return AVERROR(ENOMEM);
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s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
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s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
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s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
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s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
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s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
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if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
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return AVERROR(ENOMEM);
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s->nb_channels = outlink->channels;
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s->nb_coef_channels = ctx->inputs[1]->channels;
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s->want_skip = 1;
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s->need_padding = 1;
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s->pts = AV_NOPTS_VALUE;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioFIRContext *s = ctx->priv;
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int ch;
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if (s->sum) {
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for (ch = 0; ch < s->nb_channels; ch++) {
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av_freep(&s->sum[ch]);
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}
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}
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av_freep(&s->sum);
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if (s->coeff) {
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for (ch = 0; ch < s->nb_coef_channels; ch++) {
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av_freep(&s->coeff[ch]);
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}
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}
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av_freep(&s->coeff);
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if (s->block) {
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for (ch = 0; ch < s->nb_channels; ch++) {
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av_freep(&s->block[ch]);
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}
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}
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av_freep(&s->block);
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if (s->rdft) {
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for (ch = 0; ch < s->nb_channels; ch++) {
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av_rdft_end(s->rdft[ch]);
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|
}
|
|
}
|
|
av_freep(&s->rdft);
|
|
|
|
if (s->irdft) {
|
|
for (ch = 0; ch < s->nb_channels; ch++) {
|
|
av_rdft_end(s->irdft[ch]);
|
|
}
|
|
}
|
|
av_freep(&s->irdft);
|
|
|
|
av_frame_free(&s->in[0]);
|
|
av_frame_free(&s->in[1]);
|
|
av_frame_free(&s->buffer);
|
|
|
|
av_audio_fifo_free(s->fifo[0]);
|
|
av_audio_fifo_free(s->fifo[1]);
|
|
|
|
av_freep(&s->fdsp);
|
|
}
|
|
|
|
static av_cold int init(AVFilterContext *ctx)
|
|
{
|
|
AudioFIRContext *s = ctx->priv;
|
|
|
|
s->fcmul_add = fcmul_add_c;
|
|
|
|
s->fdsp = avpriv_float_dsp_alloc(0);
|
|
if (!s->fdsp)
|
|
return AVERROR(ENOMEM);
|
|
|
|
if (ARCH_X86)
|
|
ff_afir_init_x86(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static const AVFilterPad afir_inputs[] = {
|
|
{
|
|
.name = "main",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
},{
|
|
.name = "ir",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = read_ir,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad afir_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_output,
|
|
.request_frame = request_frame,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
|
|
#define OFFSET(x) offsetof(AudioFIRContext, x)
|
|
|
|
static const AVOption afir_options[] = {
|
|
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
|
|
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
|
|
{ "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
|
|
{ "again", "enable auto gain", OFFSET(again), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
|
|
{ NULL }
|
|
};
|
|
|
|
AVFILTER_DEFINE_CLASS(afir);
|
|
|
|
AVFilter ff_af_afir = {
|
|
.name = "afir",
|
|
.description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
|
|
.priv_size = sizeof(AudioFIRContext),
|
|
.priv_class = &afir_class,
|
|
.query_formats = query_formats,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.inputs = afir_inputs,
|
|
.outputs = afir_outputs,
|
|
.flags = AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|