@@ -3,6 +3,8 @@ releases are sorted from youngest to oldest.

 version <next>

+- aecho filter

+

 version 2.0:

@@ -347,6 +347,66 @@ aconvert=u8:auto

 @end example

 @end itemize

+@section aecho

+

+Apply echoing to the input audio.

+

+Echoes are reflected sound and can occur naturally amongst mountains

+(and sometimes large buildings) when talking or shouting; digital echo

+effects emulate this behaviour and are often used to help fill out the

+sound of a single instrument or vocal. The time difference between the

+original signal and the reflection is the @code{delay}, and the

+loudness of the reflected signal is the @code{decay}.

+Multiple echoes can have different delays and decays.

+

+A description of the accepted parameters follows.

+

+@table @option

+@item in_gain

+Set input gain of reflected signal. Default is @code{0.6}.

+

+@item out_gain

+Set output gain of reflected signal. Default is @code{0.3}.

+

+@item delays

+Set list of time intervals in milliseconds between original signal and reflections

+separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.

+Default is @code{1000}.

+

+@item decays

+Set list of loudnesses of reflected signals separated by '|'.

+Allowed range for each @code{decay} is @code{(0 - 1.0]}.

+Default is @code{0.5}.

+@end table

+

+@subsection Examples

+

+@itemize

+@item

+Make it sound as if there are twice as many instruments as are actually playing:

+@example

+aecho=0.8:0.88:60:0.4

+@end example

+

+@item

+If delay is very short, then it sound like a (metallic) robot playing music:

+@example

+aecho=0.8:0.88:6:0.4

+@end example

+

+@item

+A longer delay will sound like an open air concert in the mountains:

+@example

+aecho=0.8:0.9:1000:0.3

+@end example

+

+@item

+Same as above but with one more mountain:

+@example

+aecho=0.8:0.9:1000|1800:0.3|0.25

+@end example

+@end itemize

+

 @section afade

 Apply fade-in/out effect to input audio.

@@ -52,6 +52,7 @@ OBJS-$(CONFIG_AVFORMAT)                      += lavfutils.o

 OBJS-$(CONFIG_SWSCALE)                       += lswsutils.o

 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o

+OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o

 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o

 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o

 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o

new file mode 100644

@@ -0,0 +1,357 @@

+/*

+ * Copyright (c) 2013 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

+ *

+ */

+

+#include "libavutil/avstring.h"

+#include "libavutil/opt.h"

+#include "libavutil/samplefmt.h"

+#include "libavutil/avassert.h"

+#include "avfilter.h"

+#include "audio.h"

+#include "internal.h"

+

+typedef struct AudioEchoContext {

+    const AVClass *class;

+    float in_gain, out_gain;

+    char *delays, *decays;

+    float *delay, *decay;

+    int nb_echoes;

+    int delay_index;

+    uint8_t **delayptrs;

+    int max_samples, fade_out;

+    int *samples;

+    int64_t next_pts;

+

+    void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,

+                         uint8_t * const *src, uint8_t **dst,

+                         int nb_samples, int channels);

+} AudioEchoContext;

+

+#define OFFSET(x) offsetof(AudioEchoContext, x)

+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

+

+static const AVOption aecho_options[] = {

+    { "in_gain",  "set signal input gain",  OFFSET(in_gain),  AV_OPT_TYPE_FLOAT,  {.dbl=0.6}, 0, 1, A },

+    { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT,  {.dbl=0.3}, 0, 1, A },

+    { "delays",   "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },

+    { "decays",   "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },

+    { NULL },

+};

+

+AVFILTER_DEFINE_CLASS(aecho);

+

+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 void uninit(AVFilterContext *ctx)

+{

+    AudioEchoContext *s = ctx->priv;

+

+    av_freep(&s->delay);

+    av_freep(&s->decay);

+    av_freep(&s->samples);

+

+    if (s->delayptrs)

+        av_freep(s->delayptrs[0]);

+    av_freep(&s->delayptrs);

+}

+

+static av_cold int init(AVFilterContext *ctx)

+{

+    AudioEchoContext *s = ctx->priv;

+    int nb_delays, nb_decays, i;

+

+    if (!s->delays || !s->decays) {

+        av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");

+        return AVERROR(EINVAL);

+    }

+

+    count_items(s->delays, &nb_delays);

+    count_items(s->decays, &nb_decays);

+

+    s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));

+    s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));

+    if (!s->delay || !s->decay)

+        return AVERROR(ENOMEM);

+

+    fill_items(s->delays, &nb_delays, s->delay);

+    fill_items(s->decays, &nb_decays, s->decay);

+

+    if (nb_delays != nb_decays) {

+        av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);

+        return AVERROR(EINVAL);

+    }

+

+    s->nb_echoes = nb_delays;

+    if (!s->nb_echoes) {

+        av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");

+        return AVERROR(EINVAL);

+    }

+

+    s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));

+    if (!s->samples)

+        return AVERROR(ENOMEM);

+

+    for (i = 0; i < nb_delays; i++) {

+        if (s->delay[i] <= 0 || s->delay[i] > 90000) {

+            av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);

+            return AVERROR(EINVAL);

+        }

+        if (s->decay[i] <= 0 || s->decay[i] > 1) {

+            av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);

+            return AVERROR(EINVAL);

+        }

+    }

+

+    s->next_pts = AV_NOPTS_VALUE;

+

+    av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);

+    return 0;

+}

+

+static int query_formats(AVFilterContext *ctx)

+{

+    AVFilterChannelLayouts *layouts;

+    AVFilterFormats *formats;

+    static const enum AVSampleFormat sample_fmts[] = {

+        AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,

+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,

+        AV_SAMPLE_FMT_NONE

+    };

+

+    layouts = ff_all_channel_layouts();

+    if (!layouts)

+        return AVERROR(ENOMEM);

+    ff_set_common_channel_layouts(ctx, layouts);

+

+    formats = ff_make_format_list(sample_fmts);

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_formats(ctx, formats);

+

+    formats = ff_all_samplerates();

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_samplerates(ctx, formats);

+

+    return 0;

+}

+

+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

+

+#define ECHO(name, type, min, max)                                          \

+static void echo_samples_## name ##p(AudioEchoContext *ctx,                 \

+                                     uint8_t **delayptrs,                   \

+                                     uint8_t * const *src, uint8_t **dst,   \

+                                     int nb_samples, int channels)          \

+{                                                                           \

+    const double out_gain = ctx->out_gain;                                  \

+    const double in_gain = ctx->in_gain;                                    \

+    const int nb_echoes = ctx->nb_echoes;                                   \

+    const int max_samples = ctx->max_samples;                               \

+    int i, j, chan, index;                                                  \

+                                                                            \

+    for (chan = 0; chan < channels; chan++) {                               \

+        const type *s = (type *)src[chan];                                  \

+        type *d = (type *)dst[chan];                                        \

+        type *dbuf = (type *)delayptrs[chan];                               \

+                                                                            \

+        index = ctx->delay_index;                                           \

+        for (i = 0; i < nb_samples; i++, s++, d++) {                        \

+            double out, in;                                                 \

+                                                                            \

+            in = *s;                                                        \

+            out = in * in_gain;                                             \

+            for (j = 0; j < nb_echoes; j++) {                               \

+                int ix = index + max_samples - ctx->samples[j];             \

+                ix = MOD(ix, max_samples);                                  \

+                out += dbuf[ix] * ctx->decay[j];                            \

+            }                                                               \

+            out *= out_gain;                                                \

+                                                                            \

+            *d = av_clipd(out, min, max);                                   \

+            dbuf[index] = in;                                               \

+                                                                            \

+            index = MOD(index + 1, max_samples);                            \

+        }                                                                   \

+    }                                                                       \

+    ctx->delay_index = index;                                               \

+}

+

+ECHO(dbl, double,  -1.0,      1.0      )

+ECHO(flt, float,   -1.0,      1.0      )

+ECHO(s16, int16_t, INT16_MIN, INT16_MAX)

+ECHO(s32, int32_t, INT32_MIN, INT32_MAX)

+

+static int config_output(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    AudioEchoContext *s = ctx->priv;

+    float volume = 1.0;

+    int i;

+

+    for (i = 0; i < s->nb_echoes; i++) {

+        s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;

+        s->max_samples = FFMAX(s->max_samples, s->samples[i]);

+        volume += s->decay[i];

+    }

+

+    if (s->max_samples <= 0) {

+        av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");

+        return AVERROR(EINVAL);

+    }

+    s->fade_out = s->max_samples;

+

+    if (volume * s->in_gain * s->out_gain > 1.0)

+        av_log(ctx, AV_LOG_WARNING,

+               "out_gain %f can cause saturation of output\n", s->out_gain);

+

+    switch (outlink->format) {

+    case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;

+    case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;

+    case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;

+    case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;

+    }

+

+

+    if (s->delayptrs)

+        av_freep(s->delayptrs[0]);

+    av_freep(&s->delayptrs);

+

+    return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,

+                                              outlink->channels,

+                                              s->max_samples,

+                                              outlink->format, 0);

+}

+

+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

+{

+    AVFilterContext *ctx = inlink->dst;

+    AudioEchoContext *s = ctx->priv;

+    AVFrame *out_frame;

+

+    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);

+    }

+

+    s->echo_samples(s, s->delayptrs, frame->data, out_frame->data,

+                    frame->nb_samples, inlink->channels);

+

+    if (frame != out_frame)

+        av_frame_free(&frame);

+

+    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);

+    return ff_filter_frame(ctx->outputs[0], out_frame);

+}

+

+static int request_frame(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    AudioEchoContext *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);

+

+        s->echo_samples(s, s->delayptrs, frame->data, frame->data,

+                        frame->nb_samples, outlink->channels);

+

+        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);

+

+        return ff_filter_frame(outlink, frame);

+    }

+

+    return ret;

+}

+

+static const AVFilterPad aecho_inputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .filter_frame = filter_frame,

+    },

+    { NULL },

+};

+

+static const AVFilterPad aecho_outputs[] = {

+    {

+        .name          = "default",

+        .request_frame = request_frame,

+        .config_props  = config_output,

+        .type          = AVMEDIA_TYPE_AUDIO,

+    },

+    { NULL },

+};

+

+AVFilter avfilter_af_aecho = {

+    .name          = "aecho",

+    .description   = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),

+    .query_formats = query_formats,

+    .priv_size     = sizeof(AudioEchoContext),

+    .priv_class    = &aecho_class,

+    .init          = init,

+    .uninit        = uninit,

+    .inputs        = aecho_inputs,

+    .outputs       = aecho_outputs,

+};

@@ -48,6 +48,7 @@ void avfilter_register_all(void)

 #if FF_API_ACONVERT_FILTER

     REGISTER_FILTER(ACONVERT,       aconvert,       af);

 #endif

+    REGISTER_FILTER(AECHO,          aecho,          af);

     REGISTER_FILTER(AFADE,          afade,          af);

     REGISTER_FILTER(AFORMAT,        aformat,        af);

     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);

@@ -30,8 +30,8 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  3

-#define LIBAVFILTER_VERSION_MINOR  79

-#define LIBAVFILTER_VERSION_MICRO 101

+#define LIBAVFILTER_VERSION_MINOR  80

+#define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

                                                LIBAVFILTER_VERSION_MINOR, \