2 * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
3 * This source code is freely redistributable and may be used for
4 * any purpose. This copyright notice must be maintained.
5 * Juergen Mueller And Sundry Contributors are not responsible for
6 * the consequences of using this software.
8 * Copyright (c) 2015 Paul B Mahol
10 * This file is part of FFmpeg.
12 * FFmpeg is free software; you can redistribute it and/or
13 * modify it under the terms of the GNU Lesser General Public
14 * License as published by the Free Software Foundation; either
15 * version 2.1 of the License, or (at your option) any later version.
17 * FFmpeg is distributed in the hope that it will be useful,
18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20 * Lesser General Public License for more details.
22 * You should have received a copy of the GNU Lesser General Public
23 * License along with FFmpeg; if not, write to the Free Software
24 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
32 #include "libavutil/avstring.h"
33 #include "libavutil/opt.h"
37 #include "generate_wave_table.h"
39 typedef struct ChorusContext {
41 float in_gain, out_gain;
53 int32_t **lookup_table;
63 #define OFFSET(x) offsetof(ChorusContext, x)
64 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
66 static const AVOption chorus_options[] = {
67 { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
68 { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
69 { "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
70 { "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
71 { "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
72 { "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
76 AVFILTER_DEFINE_CLASS(chorus);
78 static void count_items(char *item_str, int *nb_items)
83 for (p = item_str; *p; p++) {
90 static void fill_items(char *item_str, int *nb_items, float *items)
92 char *p, *saveptr = NULL;
93 int i, new_nb_items = 0;
96 for (i = 0; i < *nb_items; i++) {
97 char *tstr = av_strtok(p, "|", &saveptr);
99 new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;
102 *nb_items = new_nb_items;
105 static av_cold int init(AVFilterContext *ctx)
107 ChorusContext *s = ctx->priv;
108 int nb_delays, nb_decays, nb_speeds, nb_depths;
110 if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
111 av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
112 return AVERROR(EINVAL);
115 count_items(s->delays_str, &nb_delays);
116 count_items(s->decays_str, &nb_decays);
117 count_items(s->speeds_str, &nb_speeds);
118 count_items(s->depths_str, &nb_depths);
120 s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
121 s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
122 s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
123 s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
125 if (!s->delays || !s->decays || !s->speeds || !s->depths)
126 return AVERROR(ENOMEM);
128 fill_items(s->delays_str, &nb_delays, s->delays);
129 fill_items(s->decays_str, &nb_decays, s->decays);
130 fill_items(s->speeds_str, &nb_speeds, s->speeds);
131 fill_items(s->depths_str, &nb_depths, s->depths);
133 if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
134 av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
135 return AVERROR(EINVAL);
138 s->num_chorus = nb_delays;
140 if (s->num_chorus < 1) {
141 av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
142 return AVERROR(EINVAL);
145 s->length = av_calloc(s->num_chorus, sizeof(*s->length));
146 s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
148 if (!s->length || !s->lookup_table)
149 return AVERROR(ENOMEM);
151 s->next_pts = AV_NOPTS_VALUE;
156 static int query_formats(AVFilterContext *ctx)
158 AVFilterFormats *formats;
159 AVFilterChannelLayouts *layouts;
160 static const enum AVSampleFormat sample_fmts[] = {
161 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
165 layouts = ff_all_channel_counts();
167 return AVERROR(ENOMEM);
168 ret = ff_set_common_channel_layouts(ctx, layouts);
172 formats = ff_make_format_list(sample_fmts);
174 return AVERROR(ENOMEM);
175 ret = ff_set_common_formats(ctx, formats);
179 formats = ff_all_samplerates();
181 return AVERROR(ENOMEM);
182 return ff_set_common_samplerates(ctx, formats);
185 static int config_output(AVFilterLink *outlink)
187 AVFilterContext *ctx = outlink->src;
188 ChorusContext *s = ctx->priv;
189 float sum_in_volume = 1.0;
192 s->channels = outlink->channels;
194 for (n = 0; n < s->num_chorus; n++) {
195 int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
196 int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
198 s->length[n] = outlink->sample_rate / s->speeds[n];
200 s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
201 if (!s->lookup_table[n])
202 return AVERROR(ENOMEM);
204 ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
205 s->length[n], 0., depth_samples, 0);
206 s->max_samples = FFMAX(s->max_samples, samples);
209 for (n = 0; n < s->num_chorus; n++)
210 sum_in_volume += s->decays[n];
212 if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
213 av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
215 s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
217 return AVERROR(ENOMEM);
219 s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
221 return AVERROR(ENOMEM);
223 for (n = 0; n < outlink->channels; n++) {
224 s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
226 return AVERROR(ENOMEM);
229 s->fade_out = s->max_samples;
231 return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
237 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
239 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
241 AVFilterContext *ctx = inlink->dst;
242 ChorusContext *s = ctx->priv;
246 if (av_frame_is_writable(frame)) {
249 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
251 av_frame_free(&frame);
252 return AVERROR(ENOMEM);
254 av_frame_copy_props(out_frame, frame);
257 for (c = 0; c < inlink->channels; c++) {
258 const float *src = (const float *)frame->extended_data[c];
259 float *dst = (float *)out_frame->extended_data[c];
260 float *chorusbuf = (float *)s->chorusbuf[c];
261 int *phase = s->phase[c];
263 for (i = 0; i < frame->nb_samples; i++) {
264 float out, in = src[i];
266 out = in * s->in_gain;
268 for (n = 0; n < s->num_chorus; n++) {
269 out += chorusbuf[MOD(s->max_samples + s->counter[c] -
270 s->lookup_table[n][phase[n]],
271 s->max_samples)] * s->decays[n];
272 phase[n] = MOD(phase[n] + 1, s->length[n]);
279 chorusbuf[s->counter[c]] = in;
280 s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
284 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
286 if (frame != out_frame)
287 av_frame_free(&frame);
289 return ff_filter_frame(ctx->outputs[0], out_frame);
292 static int request_frame(AVFilterLink *outlink)
294 AVFilterContext *ctx = outlink->src;
295 ChorusContext *s = ctx->priv;
298 ret = ff_request_frame(ctx->inputs[0]);
300 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
301 int nb_samples = FFMIN(s->fade_out, 2048);
304 frame = ff_get_audio_buffer(outlink, nb_samples);
306 return AVERROR(ENOMEM);
307 s->fade_out -= nb_samples;
309 av_samples_set_silence(frame->extended_data, 0,
314 frame->pts = s->next_pts;
315 if (s->next_pts != AV_NOPTS_VALUE)
316 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
318 ret = filter_frame(ctx->inputs[0], frame);
324 static av_cold void uninit(AVFilterContext *ctx)
326 ChorusContext *s = ctx->priv;
329 av_freep(&s->delays);
330 av_freep(&s->decays);
331 av_freep(&s->speeds);
332 av_freep(&s->depths);
335 av_freep(&s->chorusbuf[0]);
336 av_freep(&s->chorusbuf);
339 for (n = 0; n < s->channels; n++)
340 av_freep(&s->phase[n]);
343 av_freep(&s->counter);
344 av_freep(&s->length);
347 for (n = 0; n < s->num_chorus; n++)
348 av_freep(&s->lookup_table[n]);
349 av_freep(&s->lookup_table);
352 static const AVFilterPad chorus_inputs[] = {
355 .type = AVMEDIA_TYPE_AUDIO,
356 .filter_frame = filter_frame,
361 static const AVFilterPad chorus_outputs[] = {
364 .type = AVMEDIA_TYPE_AUDIO,
365 .request_frame = request_frame,
366 .config_props = config_output,
371 AVFilter ff_af_chorus = {
373 .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
374 .query_formats = query_formats,
375 .priv_size = sizeof(ChorusContext),
376 .priv_class = &chorus_class,
379 .inputs = chorus_inputs,
380 .outputs = chorus_outputs,