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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 "filters.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(const AVFilterContext *ctx,
AVFilterFormatsConfig **cfg_in,
AVFilterFormatsConfig **cfg_out)
{
static const enum AVSampleFormat formats[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE,
};
static const AVChannelLayout layouts[] = {
AV_CHANNEL_LAYOUT_STEREO,
{ .nb_channels = 0 },
};
int ret;
ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts);
if (ret < 0)
return ret;
return 0;
}
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_FUNC2(query_formats),
};
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