@@ -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:

@@ -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.

@@ -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

new file mode 100644

@@ -0,0 +1,379 @@

+/*

+ * 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,

+};

@@ -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);

@@ -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, \