/* * Copyright (c) 1998 Juergen Mueller And Sundry Contributors * This source code is freely redistributable and may be used for * any purpose.  This copyright notice must be maintained. * Juergen Mueller And Sundry Contributors are not responsible for * the consequences of using this software. * * Copyright (c) 2015 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 */ /** * @file * chorus audio filter */ #include "libavutil/avstring.h"#include "libavutil/mem.h"#include "libavutil/opt.h"#include "audio.h"#include "avfilter.h"#include "filters.h"#include "generate_wave_table.h" typedef struct ChorusContext {    const AVClass *class;    float in_gain, out_gain;    char *delays_str;    char *decays_str;    char *speeds_str;    char *depths_str;    float *delays;    float *decays;    float *speeds;    float *depths;    uint8_t **chorusbuf;    int **phase;    int *length;    int32_t **lookup_table;    int *counter;    int num_chorus;    int max_samples;    int channels;    int modulation;    int fade_out;    int64_t next_pts;} ChorusContext; #define OFFSET(x) offsetof(ChorusContext, x)#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption chorus_options[] = {    { "in_gain",  "set input gain",  OFFSET(in_gain),    AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },    { "out_gain", "set output gain", OFFSET(out_gain),   AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },    { "delays",   "set delays",      OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },    { "decays",   "set decays",      OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },    { "speeds",   "set speeds",      OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },    { "depths",   "set depths",      OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },    { NULL }}; AVFILTER_DEFINE_CLASS(chorus); static void count_items(char *item_str, int *nb_items){    char *p;     *nb_items = 1;    for (p = item_str; *p; p++) {        if (*p == '|')            (*nb_items)++;    } } static void fill_items(char *item_str, int *nb_items, float *items){    char *p, *saveptr = NULL;    int i, new_nb_items = 0;     p = item_str;    for (i = 0; i < *nb_items; i++) {        char *tstr = av_strtok(p, "|", &saveptr);        p = NULL;        if (tstr)            new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;    }     *nb_items = new_nb_items;} static av_cold int init(AVFilterContext *ctx){    ChorusContext *s = ctx->priv;    int nb_delays, nb_decays, nb_speeds, nb_depths;     if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {        av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");        return AVERROR(EINVAL);    }     count_items(s->delays_str, &nb_delays);    count_items(s->decays_str, &nb_decays);    count_items(s->speeds_str, &nb_speeds);    count_items(s->depths_str, &nb_depths);     s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));    s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));    s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));    s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));     if (!s->delays || !s->decays || !s->speeds || !s->depths)        return AVERROR(ENOMEM);     fill_items(s->delays_str, &nb_delays, s->delays);    fill_items(s->decays_str, &nb_decays, s->decays);    fill_items(s->speeds_str, &nb_speeds, s->speeds);    fill_items(s->depths_str, &nb_depths, s->depths);     if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {        av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");        return AVERROR(EINVAL);    }     s->num_chorus = nb_delays;     if (s->num_chorus < 1) {        av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");        return AVERROR(EINVAL);    }     s->length = av_calloc(s->num_chorus, sizeof(*s->length));    s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));     if (!s->length || !s->lookup_table)        return AVERROR(ENOMEM);     s->next_pts = AV_NOPTS_VALUE;     return 0;} static int config_output(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    ChorusContext *s = ctx->priv;    float sum_in_volume = 1.0;    int n;     s->channels = outlink->ch_layout.nb_channels;     for (n = 0; n < s->num_chorus; n++) {        int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);        int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);         s->length[n] = outlink->sample_rate / s->speeds[n];         s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);        if (!s->lookup_table[n])            return AVERROR(ENOMEM);         ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],                               s->length[n], 0., depth_samples, 0);        s->max_samples = FFMAX(s->max_samples, samples);    }     for (n = 0; n < s->num_chorus; n++)        sum_in_volume += s->decays[n];     if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)        av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");     s->counter = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->counter));    if (!s->counter)        return AVERROR(ENOMEM);     s->phase = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->phase));    if (!s->phase)        return AVERROR(ENOMEM);     for (n = 0; n < outlink->ch_layout.nb_channels; n++) {        s->phase[n] = av_calloc(s->num_chorus, sizeof(int));        if (!s->phase[n])            return AVERROR(ENOMEM);    }     s->fade_out = s->max_samples;     return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,                                              outlink->ch_layout.nb_channels,                                              s->max_samples,                                              outlink->format, 0);} #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) static int filter_frame(AVFilterLink *inlink, AVFrame *frame){    AVFilterContext *ctx = inlink->dst;    ChorusContext *s = ctx->priv;    AVFrame *out_frame;    int c, i, n;     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 (c = 0; c < inlink->ch_layout.nb_channels; c++) {        const float *src = (const float *)frame->extended_data[c];        float *dst = (float *)out_frame->extended_data[c];        float *chorusbuf = (float *)s->chorusbuf[c];        int *phase = s->phase[c];         for (i = 0; i < frame->nb_samples; i++) {            float out, in = src[i];             out = in * s->in_gain;             for (n = 0; n < s->num_chorus; n++) {                out += chorusbuf[MOD(s->max_samples + s->counter[c] -                                     s->lookup_table[n][phase[n]],                                     s->max_samples)] * s->decays[n];                phase[n] = MOD(phase[n] + 1, s->length[n]);            }             out *= s->out_gain;             dst[i] = out;             chorusbuf[s->counter[c]] = in;            s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);        }    }     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);     if (frame != out_frame)        av_frame_free(&frame);     return ff_filter_frame(ctx->outputs[0], out_frame);} static int request_frame(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    ChorusContext *s = ctx->priv;    int ret;     ret = ff_request_frame(ctx->inputs[0]);     if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {        int nb_samples = FFMIN(s->fade_out, 2048);        AVFrame *frame;         frame = ff_get_audio_buffer(outlink, nb_samples);        if (!frame)            return AVERROR(ENOMEM);        s->fade_out -= nb_samples;         av_samples_set_silence(frame->extended_data, 0,                               frame->nb_samples,                               outlink->ch_layout.nb_channels,                               frame->format);         frame->pts = s->next_pts;        if (s->next_pts != AV_NOPTS_VALUE)            s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);         ret = filter_frame(ctx->inputs[0], frame);    }     return ret;} static av_cold void uninit(AVFilterContext *ctx){    ChorusContext *s = ctx->priv;    int n;     av_freep(&s->delays);    av_freep(&s->decays);    av_freep(&s->speeds);    av_freep(&s->depths);     if (s->chorusbuf)        av_freep(&s->chorusbuf[0]);    av_freep(&s->chorusbuf);     if (s->phase)        for (n = 0; n < s->channels; n++)            av_freep(&s->phase[n]);    av_freep(&s->phase);     av_freep(&s->counter);    av_freep(&s->length);     if (s->lookup_table)        for (n = 0; n < s->num_chorus; n++)            av_freep(&s->lookup_table[n]);    av_freep(&s->lookup_table);} static const AVFilterPad chorus_inputs[] = {    {        .name         = "default",        .type         = AVMEDIA_TYPE_AUDIO,        .filter_frame = filter_frame,    },}; static const AVFilterPad chorus_outputs[] = {    {        .name          = "default",        .type          = AVMEDIA_TYPE_AUDIO,        .request_frame = request_frame,        .config_props  = config_output,    },}; const FFFilter ff_af_chorus = {    .p.name        = "chorus",    .p.description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),    .p.priv_class  = &chorus_class,    .priv_size     = sizeof(ChorusContext),    .init          = init,    .uninit        = uninit,    FILTER_INPUTS(chorus_inputs),    FILTER_OUTPUTS(chorus_outputs),    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),};