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/*
* Copyright (c) 2011 Mina Nagy Zaki
* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
* This source code is freely redistributable and may be used for any purpose.
* This copyright notice must be maintained. Edward Beingessner And Sundry
* Contributors are not responsible for the consequences of using this
* software.
*
* 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
* Stereo Widening Effect. Adds audio cues to move stereo image in
* front of the listener. Adapted from the libsox earwax effect.
*/
#include "libavutil/channel_layout.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
#define NUMTAPS 32
static const int8_t filt[NUMTAPS * 2] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
-1, -5, /* signal was from 30 degrees */
3, 3, /* from the ear, the other as */
-2, 5, /* if 330 degrees. */
-5, 0,
9, 1,
6, 3, /* Input */
-4, -1, /* Left Right */
-5, -3, /* __________ __________ */
-2, -5, /* | | | | */
-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
6, -7, /* / |__________| |__________| \ */
30, -29, /* / \ / \ */
12, -3, /* / X \ */
-11, 4, /* / / \ \ */
-3, 7, /* ____V_____ __________V V__________ _____V____ */
-20, 23, /* | | | | | | | | */
2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
1, -6, /* |__________| |__________| |__________| |__________| */
-14, -5, /* \ ___ / \ ___ / */
15, -18, /* \ / \ / _____ \ / \ / */
6, 7, /* `->| + |<--' / \ `-->| + |<-' */
15, -10, /* \___/ _/ \_ \___/ */
-14, 22, /* \ / \ / \ / */
-7, -2, /* `--->| | | |<---' */
-4, 9, /* \_/ \_/ */
6, -12, /* */
6, -6, /* Headphones */
0, -11,
0, -5,
4, 0};
typedef struct EarwaxContext {
int16_t filter[2][NUMTAPS];
int16_t taps[4][NUMTAPS * 2];
AVFrame *frame[2];
} EarwaxContext;
static int query_formats(AVFilterContext *ctx)
{
static const int sample_rates[] = { 44100, -1 };
int ret;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
(ret = ff_set_common_samplerates_from_list(ctx, sample_rates)) < 0)
return ret;
return 0;
}
//FIXME: replace with DSPContext.scalarproduct_int16
static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
const int16_t *filt, int16_t *out)
{
int32_t sample;
int16_t j;
while (in < endin) {
sample = 0;
for (j = 0; j < NUMTAPS; j++)
sample += in[j] * filt[j];
*out = av_clip_int16(sample >> 7);
out++;
in++;
}
return out;
}
static int config_input(AVFilterLink *inlink)
{
EarwaxContext *s = inlink->dst->priv;
for (int i = 0; i < NUMTAPS; i++) {
s->filter[0][i] = filt[i * 2];
s->filter[1][i] = filt[i * 2 + 1];
}
return 0;
}
static void convolve(AVFilterContext *ctx, AVFrame *in,
int input_ch, int output_ch,
int filter_ch, int tap_ch)
{
EarwaxContext *s = ctx->priv;
int16_t *taps, *endin, *dst, *src;
int len;
taps = s->taps[tap_ch];
dst = (int16_t *)s->frame[input_ch]->data[output_ch];
src = (int16_t *)in->data[input_ch];
len = FFMIN(NUMTAPS, in->nb_samples);
// copy part of new input and process with saved input
memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
// process current input
if (in->nb_samples >= NUMTAPS) {
endin = src + in->nb_samples - NUMTAPS;
scalarproduct(src, endin, s->filter[filter_ch], dst);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
} else {
memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
}
}
static void mix(AVFilterContext *ctx, AVFrame *out,
int output_ch, int f0, int f1, int i0, int i1)
{
EarwaxContext *s = ctx->priv;
const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
int16_t *dst = (int16_t *)out->data[output_ch];
for (int n = 0; n < out->nb_samples; n++)
dst[n] = av_clip_int16(srcl[n] + srcr[n]);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
EarwaxContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
for (int ch = 0; ch < 2; ch++) {
if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
av_frame_free(&s->frame[ch]);
s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
if (!s->frame[ch]) {
av_frame_free(&in);
av_frame_free(&out);
return AVERROR(ENOMEM);
}
}
}
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
convolve(ctx, in, 0, 0, 0, 0);
convolve(ctx, in, 0, 1, 1, 1);
convolve(ctx, in, 1, 0, 0, 2);
convolve(ctx, in, 1, 1, 1, 3);
mix(ctx, out, 0, 0, 1, 1, 0);
mix(ctx, out, 1, 0, 1, 0, 1);
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
EarwaxContext *s = ctx->priv;
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
}
static const AVFilterPad earwax_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
const AVFilter ff_af_earwax = {
.name = "earwax",
.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
.priv_size = sizeof(EarwaxContext),
.uninit = uninit,
FILTER_INPUTS(earwax_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC(query_formats),
};
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