/*

* 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/avassert.h"

#include "libavutil/avstring.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "avfilter.h"

#include "audio.h"

#include "internal.h"

typedef struct AudioEchoContext {

} 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;

if (tstr)

new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 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

};

int ret;

layouts = ff_all_channel_counts();

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);

}

#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, av_uninit(index); \

\

av_assert1(channels > 0); /* would corrupt delay_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(ctx->outputs[0], frame->nb_samples);

if (!out_frame) {

av_frame_free(&frame);

return AVERROR(ENOMEM);

}

av_frame_copy_props(out_frame, frame);

}

s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,

frame->nb_samples, inlink->channels);

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;

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->extended_data, frame->extended_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 ff_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,

};