/* * 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/mem.h"#include "libavutil/opt.h"#include "libavutil/samplefmt.h"#include "avfilter.h"#include "audio.h"#include "filters.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;    int eof;    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;        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;} #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->ch_layout.nb_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->ch_layout.nb_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 nb_samples = FFMIN(s->fade_out, 2048);    AVFrame *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->ch_layout.nb_channels,                           frame->format);     s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,                    frame->nb_samples, outlink->ch_layout.nb_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);} static int activate(AVFilterContext *ctx){    AVFilterLink *inlink = ctx->inputs[0];    AVFilterLink *outlink = ctx->outputs[0];    AudioEchoContext *s = ctx->priv;    AVFrame *in;    int ret, status;    int64_t pts;     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);     ret = ff_inlink_consume_frame(inlink, &in);    if (ret < 0)        return ret;    if (ret > 0)        return filter_frame(inlink, in);     if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {        if (status == AVERROR_EOF)            s->eof = 1;    }     if (s->eof && s->fade_out <= 0) {        ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);        return 0;    }     if (!s->eof)        FF_FILTER_FORWARD_WANTED(outlink, inlink);     return request_frame(outlink);} static const AVFilterPad aecho_outputs[] = {    {        .name          = "default",        .config_props  = config_output,        .type          = AVMEDIA_TYPE_AUDIO,    },}; const FFFilter ff_af_aecho = {    .p.name        = "aecho",    .p.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),    .p.priv_class  = &aecho_class,    .priv_size     = sizeof(AudioEchoContext),    .init          = init,    .activate      = activate,    .uninit        = uninit,    FILTER_INPUTS(ff_audio_default_filterpad),    FILTER_OUTPUTS(aecho_outputs),    FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,                      AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),};