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