mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-18 03:19:31 +02:00
a04ad248a0
This is possible now that the next-API is gone. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> Signed-off-by: James Almer <jamrial@gmail.com>
229 lines
8.0 KiB
C
229 lines
8.0 KiB
C
/*
|
|
* 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/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, "source" },
|
|
{ "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" },
|
|
{ "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" },
|
|
{ "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" },
|
|
{ "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "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 , AV_CH_LAYOUT_STEREO)) < 0 ||
|
|
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
|
|
return ret;
|
|
|
|
formats = ff_all_samplerates();
|
|
return ff_set_common_samplerates(ctx, formats);
|
|
}
|
|
|
|
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,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
const AVFilter ff_af_haas = {
|
|
.name = "haas",
|
|
.description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(HaasContext),
|
|
.priv_class = &haas_class,
|
|
.uninit = uninit,
|
|
.inputs = inputs,
|
|
.outputs = outputs,
|
|
};
|