242 lines
8.4 KiB
C
242 lines
8.4 KiB
C
/*
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* Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
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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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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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#include "generate_wave_table.h"
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#define INTERPOLATION_LINEAR 0
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#define INTERPOLATION_QUADRATIC 1
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typedef struct FlangerContext {
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const AVClass *class;
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double delay_min;
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double delay_depth;
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double feedback_gain;
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double delay_gain;
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double speed;
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int wave_shape;
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double channel_phase;
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int interpolation;
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double in_gain;
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int max_samples;
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uint8_t **delay_buffer;
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int delay_buf_pos;
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double *delay_last;
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float *lfo;
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int lfo_length;
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int lfo_pos;
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} FlangerContext;
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#define OFFSET(x) offsetof(FlangerContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption flanger_options[] = {
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{ "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
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{ "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
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{ "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
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{ "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
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{ "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
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{ "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
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{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
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{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
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{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
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{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
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{ "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
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{ "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
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{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
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{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(flanger);
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static int init(AVFilterContext *ctx)
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{
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FlangerContext *s = ctx->priv;
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s->feedback_gain /= 100;
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s->delay_gain /= 100;
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s->channel_phase /= 100;
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s->delay_min /= 1000;
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s->delay_depth /= 1000;
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s->in_gain = 1 / (1 + s->delay_gain);
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s->delay_gain /= 1 + s->delay_gain;
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s->delay_gain *= 1 - fabs(s->feedback_gain);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterChannelLayouts *layouts;
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AVFilterFormats *formats;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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FlangerContext *s = ctx->priv;
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s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
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s->lfo_length = inlink->sample_rate / s->speed;
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s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
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s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
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if (!s->lfo || !s->delay_last)
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return AVERROR(ENOMEM);
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ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
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floor(s->delay_min * inlink->sample_rate + 0.5),
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s->max_samples - 2., 3 * M_PI_2);
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return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
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inlink->channels, s->max_samples,
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inlink->format, 0);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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FlangerContext *s = ctx->priv;
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AVFrame *out_frame;
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int chan, i;
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if (av_frame_is_writable(frame)) {
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out_frame = frame;
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} else {
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out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
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if (!out_frame)
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return AVERROR(ENOMEM);
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av_frame_copy_props(out_frame, frame);
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}
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for (i = 0; i < frame->nb_samples; i++) {
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s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
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for (chan = 0; chan < inlink->channels; chan++) {
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double *src = (double *)frame->extended_data[chan];
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double *dst = (double *)out_frame->extended_data[chan];
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double delayed_0, delayed_1;
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double delayed;
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double in, out;
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int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
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double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
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int int_delay = (int)delay;
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double frac_delay = modf(delay, &delay);
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double *delay_buffer = (double *)s->delay_buffer[chan];
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in = src[i];
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delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
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s->feedback_gain;
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delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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if (s->interpolation == INTERPOLATION_LINEAR) {
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delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
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} else {
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double a, b;
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double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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delayed_2 -= delayed_0;
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delayed_1 -= delayed_0;
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a = delayed_2 * .5 - delayed_1;
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b = delayed_1 * 2 - delayed_2 *.5;
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delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
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}
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s->delay_last[chan] = delayed;
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out = in * s->in_gain + delayed * s->delay_gain;
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dst[i] = out;
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}
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s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
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}
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if (frame != out_frame)
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av_frame_free(&frame);
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return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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FlangerContext *s = ctx->priv;
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av_freep(&s->lfo);
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av_freep(&s->delay_last);
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if (s->delay_buffer)
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av_freep(&s->delay_buffer[0]);
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av_freep(&s->delay_buffer);
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}
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static const AVFilterPad flanger_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad flanger_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_flanger = {
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.name = "flanger",
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.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(FlangerContext),
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.priv_class = &flanger_class,
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.init = init,
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.uninit = uninit,
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.inputs = flanger_inputs,
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.outputs = flanger_outputs,
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};
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