391 lines
13 KiB
C
391 lines
13 KiB
C
/*
|
|
* 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 "filters.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;
|
|
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;
|
|
}
|
|
|
|
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 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->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);
|
|
}
|
|
|
|
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_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad aecho_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.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,
|
|
.activate = activate,
|
|
.uninit = uninit,
|
|
.inputs = aecho_inputs,
|
|
.outputs = aecho_outputs,
|
|
};
|