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FFmpeg/libavfilter/af_haas.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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/*
* Copyright (c) 2001-2010 Vladimir Sadovnikov
*
* 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/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
#define MAX_HAAS_DELAY 40
typedef struct HaasContext {
const AVClass *class;
int par_m_source;
double par_delay0;
double par_delay1;
int par_phase0;
int par_phase1;
int par_middle_phase;
double par_side_gain;
double par_gain0;
double par_gain1;
double par_balance0;
double par_balance1;
double level_in;
double level_out;
double *buffer;
size_t buffer_size;
uint32_t write_ptr;
uint32_t delay[2];
double balance_l[2];
double balance_r[2];
double phase0;
double phase1;
} HaasContext;
#define OFFSET(x) offsetof(HaasContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption haas_options[] = {
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, .unit = "source" },
{ "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "source" },
{ "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "source" },
{ "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "source" },
{ "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, .unit = "source" },
{ "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
{ "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
{ "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
{ "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
{ "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(haas);
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
int ret;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 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)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
HaasContext *s = ctx->priv;
size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
size_t new_buf_size = 1;
while (new_buf_size < min_buf_size)
new_buf_size <<= 1;
av_freep(&s->buffer);
s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
if (!s->buffer)
return AVERROR(ENOMEM);
s->buffer_size = new_buf_size;
s->write_ptr = 0;
s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
s->phase0 = s->par_phase0 ? 1.0 : -1.0;
s->phase1 = s->par_phase1 ? 1.0 : -1.0;
s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
HaasContext *s = ctx->priv;
const double *src = (const double *)in->data[0];
const double level_in = s->level_in;
const double level_out = s->level_out;
const uint32_t mask = s->buffer_size - 1;
double *buffer = s->buffer;
AVFrame *out;
double *dst;
int n;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
double mid, side[2], side_l, side_r;
uint32_t s0_ptr, s1_ptr;
switch (s->par_m_source) {
case 0: mid = src[0]; break;
case 1: mid = src[1]; break;
case 2: mid = (src[0] + src[1]) * 0.5; break;
case 3: mid = (src[0] - src[1]) * 0.5; break;
}
mid *= level_in;
buffer[s->write_ptr] = mid;
s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
if (s->par_middle_phase)
mid = -mid;
side[0] = buffer[s0_ptr] * s->par_side_gain;
side[1] = buffer[s1_ptr] * s->par_side_gain;
side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
dst[0] = (mid + side_l) * level_out;
dst[1] = (mid + side_r) * level_out;
s->write_ptr = (s->write_ptr + 1) & mask;
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
HaasContext *s = ctx->priv;
av_freep(&s->buffer);
s->buffer_size = 0;
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
const AVFilter ff_af_haas = {
.name = "haas",
.description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
.priv_size = sizeof(HaasContext),
.priv_class = &haas_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC(query_formats),
};
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