avfilter: add chorus filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -16,6 +16,7 @@ version <next>:
 - unpack DivX-style packed B-frames in MPEG-4 bitstream filter
 - WebM Live Chunk Muxer
 - nvenc level and tier options
+- chorus filter
 
 
 version 2.6:

doc/filters.texi

@@ -1320,6 +1320,61 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
 side_right.wav
 @end example
 
+@section chorus
+Add a chorus effect to the audio.
+
+Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
+
+Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
+constant, with chorus, it is varied using using sinusoidal or triangular modulation.
+The modulation depth defines the range the modulated delay is played before or after
+the delay. Hence the delayed sound will sound slower or faster, that is the delayed
+sound tuned around the original one, like in a chorus where some vocals are slightly
+off key.
+
+It accepts the following parameters:
+@table @option
+@item in_gain
+Set input gain. Default is 0.4.
+
+@item out_gain
+Set output gain. Default is 0.4.
+
+@item delays
+Set delays. A typical delay is around 40ms to 60ms.
+
+@item decays
+Set decays.
+
+@item speeds
+Set speeds.
+
+@item depths
+Set depths.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+A single delay:
+@example
+chorus=0.7:0.9:55:0.4:0.25:2
+@end example
+
+@item
+Two delays:
+@example
+chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
+@end example
+
+@item
+Fuller sounding chorus with three delays:
+@example
+chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
+@end example
+@end itemize
+
 @section compand
 Compress or expand the audio's dynamic range.
 

libavfilter/Makefile

@@ -64,6 +64,7 @@ OBJS-$(CONFIG_BIQUAD_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_BS2B_FILTER)                   += af_bs2b.o
 OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o
 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
+OBJS-$(CONFIG_CHORUS_FILTER)                 += af_chorus.o generate_wave_table.o
 OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o
 OBJS-$(CONFIG_DCSHIFT_FILTER)                += af_dcshift.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o

libavfilter/af_chorus.c

+/*
+ * 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/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.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;
+        new_nb_items += sscanf(tstr, "%f", &items[i]) == 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 query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+    };
+    int ret;
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+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->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->channels, sizeof(*s->counter));
+    if (!s->counter)
+        return AVERROR(ENOMEM);
+
+    s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
+    if (!s->phase)
+        return AVERROR(ENOMEM);
+
+    for (n = 0; n < outlink->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->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(inlink, frame->nb_samples);
+        if (!out_frame)
+            return AVERROR(ENOMEM);
+        av_frame_copy_props(out_frame, frame);
+    }
+
+    for (c = 0; c < inlink->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->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,
+    },
+    { NULL }
+};
+
+static const AVFilterPad chorus_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .request_frame = request_frame,
+        .config_props  = config_output,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_chorus = {
+    .name          = "chorus",
+    .description   = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(ChorusContext),
+    .priv_class    = &chorus_class,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = chorus_inputs,
+    .outputs       = chorus_outputs,
+};

libavfilter/allfilters.c

@@ -80,6 +80,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(BS2B,           bs2b,           af);
     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
+    REGISTER_FILTER(CHORUS,         chorus,         af);
     REGISTER_FILTER(COMPAND,        compand,        af);
     REGISTER_FILTER(DCSHIFT,        dcshift,        af);
     REGISTER_FILTER(EARWAX,         earwax,         af);

libavfilter/version.h

@@ -30,8 +30,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  5
-#define LIBAVFILTER_VERSION_MINOR  13
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR  14
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \