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avfilter: add firequalizer filter

Browse Source 
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This commit is contained in:
Muhammad Faiz 2016-02-17 01:02:22 +07:00
parent 1387f3a051
commit bfc61b0fcc
8 changed files with 708 additions and 1 deletions
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1
Changelog
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@ -6,6 +6,7 @@ version <next>:
- fieldhint filter
- loop video filter and aloop audio filter
- Bob Weaver deinterlacing filter
- firequalizer filter
version 3.0:

1
MAINTAINERS
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@ -353,6 +353,7 @@ Filters:
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
af_firequalizer.c Muhammad Faiz
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
af_sidechaincompress.c Paul B Mahol

2
configure vendored
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@ -2861,6 +2861,8 @@ eq_filter_deps="gpl"
fftfilt_filter_deps="avcodec"
fftfilt_filter_select="rdft"
find_rect_filter_deps="avcodec avformat gpl"
firequalizer_filter_deps="avcodec"
firequalizer_filter_select="rdft"
flite_filter_deps="libflite"
frei0r_filter_deps="frei0r dlopen"
frei0r_src_filter_deps="frei0r dlopen"

109
doc/filters.texi
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@ -2366,6 +2366,115 @@ Sets the difference coefficient (default: 2.5). 0.0 means mono sound
Enable clipping. By default is enabled.
@end table
@section firequalizer
Apply FIR Equalization using arbitrary frequency response.
The filter accepts the following option:
@table @option
@item gain
Set gain curve equation (in dB). The expression can contain variables:
@table @option
@item f
the evaluated frequency
@item sr
sample rate
@item ch
channel number, set to 0 when multichannels evaluation is disabled
@item chid
channel id, see libavutil/channel_layout.h, set to the first channel id when
multichannels evaluation is disabled
@item chs
number of channels
@item chlayout
channel_layout, see libavutil/channel_layout.h
@end table
and functions:
@table @option
@item gain_interpolate(f)
interpolate gain on frequency f based on gain_entry
@end table
This option is also available as command. Default is @code{gain_interpolate(f)}.
@item gain_entry
Set gain entry for gain_interpolate function. The expression can
contain functions:
@table @option
@item entry(f, g)
store gain entry at frequency f with value g
@end table
This option is also available as command.
@item delay
Set filter delay in seconds. Higher value means more accurate.
Default is @code{0.01}.
@item accuracy
Set filter accuracy in Hz. Lower value means more accurate.
Default is @code{5}.
@item wfunc
Set window function. Acceptable values are:
@table @option
@item rectangular
rectangular window, useful when gain curve is already smooth
@item hann
hann window (default)
@item hamming
hamming window
@item blackman
blackman window
@item nuttall3
3-terms continuous 1st derivative nuttall window
@item mnuttall3
minimum 3-terms discontinuous nuttall window
@item nuttall
4-terms continuous 1st derivative nuttall window
@item bnuttall
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
@item bharris
blackman-harris window
@end table
@item fixed
If enabled, use fixed number of audio samples. This improves speed when
filtering with large delay. Default is disabled.
@item multi
Enable multichannels evaluation on gain. Default is disabled.
@end table
@subsection Examples
@itemize
@item
lowpass at 1000 Hz:
@example
firequalizer=gain='if(lt(f,1000), 0, -INF)'
@end example
@item
lowpass at 1000 Hz with gain_entry:
@example
firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
@end example
@item
custom equalization:
@example
firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
@end example
@item
higher delay:
@example
firequalizer=delay=0.1:fixed=on
@end example
@item
lowpass on left channel, highpass on right channel:
@example
firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
@end example
@end itemize
@section flanger
Apply a flanging effect to the audio.

1
libavfilter/Makefile
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@ -80,6 +80,7 @@ OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o
OBJS-$(CONFIG_EXTRASTEREO_FILTER) += af_extrastereo.o
OBJS-$(CONFIG_FIREQUALIZER_FILTER) += af_firequalizer.o
OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table.o
OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o

592
libavfilter/af_firequalizer.c Normal file
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@ -0,0 +1,592 @@
/*
* Copyright (c) 2016 Muhammad Faiz <mfcc64@gmail.com>
*
* 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/opt.h"
#include "libavutil/eval.h"
#include "libavutil/avassert.h"
#include "libavcodec/avfft.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#define RDFT_BITS_MIN 4
#define RDFT_BITS_MAX 16
enum WindowFunc {
WFUNC_MIN,
WFUNC_RECTANGULAR = WFUNC_MIN,
WFUNC_HANN,
WFUNC_HAMMING,
WFUNC_BLACKMAN,
WFUNC_NUTTALL3,
WFUNC_MNUTTALL3,
WFUNC_NUTTALL,
WFUNC_BNUTTALL,
WFUNC_BHARRIS,
WFUNC_MAX = WFUNC_BHARRIS
};
#define NB_GAIN_ENTRY_MAX 4096
typedef struct {
double freq;
double gain;
} GainEntry;
typedef struct {
int buf_idx;
int overlap_idx;
} OverlapIndex;
typedef struct {
const AVClass *class;
RDFTContext *analysis_irdft;
RDFTContext *rdft;
RDFTContext *irdft;
int analysis_rdft_len;
int rdft_len;
float *analysis_buf;
float *kernel_tmp_buf;
float *kernel_buf;
float *conv_buf;
OverlapIndex *conv_idx;
int fir_len;
int nsamples_max;
int64_t next_pts;
int frame_nsamples_max;
int remaining;
char *gain_cmd;
char *gain_entry_cmd;
const char *gain;
const char *gain_entry;
double delay;
double accuracy;
int wfunc;
int fixed;
int multi;
int nb_gain_entry;
int gain_entry_err;
GainEntry gain_entry_tbl[NB_GAIN_ENTRY_MAX];
} FIREqualizerContext;
#define OFFSET(x) offsetof(FIREqualizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption firequalizer_options[] = {
{ "gain", "set gain curve", OFFSET(gain), AV_OPT_TYPE_STRING, { .str = "gain_interpolate(f)" }, 0, 0, FLAGS },
{ "gain_entry", "set gain entry", OFFSET(gain_entry), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, FLAGS },
{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.0, 1e10, FLAGS },
{ "accuracy", "set accuracy", OFFSET(accuracy), AV_OPT_TYPE_DOUBLE, { .dbl = 5.0 }, 0.0, 1e10, FLAGS },
{ "wfunc", "set window function", OFFSET(wfunc), AV_OPT_TYPE_INT, { .i64 = WFUNC_HANN }, WFUNC_MIN, WFUNC_MAX, FLAGS, "wfunc" },
{ "rectangular", "rectangular window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_RECTANGULAR }, 0, 0, FLAGS, "wfunc" },
{ "hann", "hann window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HANN }, 0, 0, FLAGS, "wfunc" },
{ "hamming", "hamming window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HAMMING }, 0, 0, FLAGS, "wfunc" },
{ "blackman", "blackman window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BLACKMAN }, 0, 0, FLAGS, "wfunc" },
{ "nuttall3", "3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL3 }, 0, 0, FLAGS, "wfunc" },
{ "mnuttall3", "minimum 3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_MNUTTALL3 }, 0, 0, FLAGS, "wfunc" },
{ "nuttall", "nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL }, 0, 0, FLAGS, "wfunc" },
{ "bnuttall", "blackman-nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BNUTTALL }, 0, 0, FLAGS, "wfunc" },
{ "bharris", "blackman-harris window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BHARRIS }, 0, 0, FLAGS, "wfunc" },
{ "fixed", "set fixed frame samples", OFFSET(fixed), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
{ "multi", "set multi channels mode", OFFSET(multi), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(firequalizer);
static void common_uninit(FIREqualizerContext *s)
{
av_rdft_end(s->analysis_irdft);
av_rdft_end(s->rdft);
av_rdft_end(s->irdft);
s->analysis_irdft = s->rdft = s->irdft = NULL;
av_freep(&s->analysis_buf);
av_freep(&s->kernel_tmp_buf);
av_freep(&s->kernel_buf);
av_freep(&s->conv_buf);
av_freep(&s->conv_idx);
}
static av_cold void uninit(AVFilterContext *ctx)
{
FIREqualizerContext *s = ctx->priv;
common_uninit(s);
av_freep(&s->gain_cmd);
av_freep(&s->gain_entry_cmd);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
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);
}
static void fast_convolute(FIREqualizerContext *s, const float *kernel_buf, float *conv_buf,
OverlapIndex *idx, float *data, int nsamples)
{
if (nsamples <= s->nsamples_max) {
float *buf = conv_buf + idx->buf_idx * s->rdft_len;
float *obuf = conv_buf + !idx->buf_idx * s->rdft_len + idx->overlap_idx;
int k;
memcpy(buf, data, nsamples * sizeof(*data));
memset(buf + nsamples, 0, (s->rdft_len - nsamples) * sizeof(*data));
av_rdft_calc(s->rdft, buf);
buf[0] *= kernel_buf[0];
buf[1] *= kernel_buf[1];
for (k = 2; k < s->rdft_len; k += 2) {
float re, im;
re = buf[k] * kernel_buf[k] - buf[k+1] * kernel_buf[k+1];
im = buf[k] * kernel_buf[k+1] + buf[k+1] * kernel_buf[k];
buf[k] = re;
buf[k+1] = im;
}
av_rdft_calc(s->irdft, buf);
for (k = 0; k < s->rdft_len - idx->overlap_idx; k++)
buf[k] += obuf[k];
memcpy(data, buf, nsamples * sizeof(*data));
idx->buf_idx = !idx->buf_idx;
idx->overlap_idx = nsamples;
} else {
while (nsamples > s->nsamples_max * 2) {
fast_convolute(s, kernel_buf, conv_buf, idx, data, s->nsamples_max);
data += s->nsamples_max;
nsamples -= s->nsamples_max;
}
fast_convolute(s, kernel_buf, conv_buf, idx, data, nsamples/2);
fast_convolute(s, kernel_buf, conv_buf, idx, data + nsamples/2, nsamples - nsamples/2);
}
}
static double entry_func(void *p, double freq, double gain)
{
AVFilterContext *ctx = p;
FIREqualizerContext *s = ctx->priv;
if (s->nb_gain_entry >= NB_GAIN_ENTRY_MAX) {
av_log(ctx, AV_LOG_ERROR, "entry table overflow.\n");
s->gain_entry_err = AVERROR(EINVAL);
return 0;
}
if (isnan(freq)) {
av_log(ctx, AV_LOG_ERROR, "nan frequency (%g, %g).\n", freq, gain);
s->gain_entry_err = AVERROR(EINVAL);
return 0;
}
if (s->nb_gain_entry > 0 && freq <= s->gain_entry_tbl[s->nb_gain_entry - 1].freq) {
av_log(ctx, AV_LOG_ERROR, "unsorted frequency (%g, %g).\n", freq, gain);
s->gain_entry_err = AVERROR(EINVAL);
return 0;
}
s->gain_entry_tbl[s->nb_gain_entry].freq = freq;
s->gain_entry_tbl[s->nb_gain_entry].gain = gain;
s->nb_gain_entry++;
return 0;
}
static int gain_entry_compare(const void *key, const void *memb)
{
const double *freq = key;
const GainEntry *entry = memb;
if (*freq < entry[0].freq)
return -1;
if (*freq > entry[1].freq)
return 1;
return 0;
}
static double gain_interpolate_func(void *p, double freq)
{
AVFilterContext *ctx = p;
FIREqualizerContext *s = ctx->priv;
GainEntry *res;
double d0, d1, d;
if (isnan(freq))
return freq;
if (!s->nb_gain_entry)
return 0;
if (freq <= s->gain_entry_tbl[0].freq)
return s->gain_entry_tbl[0].gain;
if (freq >= s->gain_entry_tbl[s->nb_gain_entry-1].freq)
return s->gain_entry_tbl[s->nb_gain_entry-1].gain;
res = bsearch(&freq, &s->gain_entry_tbl, s->nb_gain_entry - 1, sizeof(*res), gain_entry_compare);
av_assert0(res);
d = res[1].freq - res[0].freq;
d0 = freq - res[0].freq;
d1 = res[1].freq - freq;
if (d0 && d1)
return (d0 * res[1].gain + d1 * res[0].gain) / d;
if (d0)
return res[1].gain;
return res[0].gain;
}
static const char *const var_names[] = {
"f",
"sr",
"ch",
"chid",
"chs",
"chlayout",
NULL
};
enum VarOffset {
VAR_F,
VAR_SR,
VAR_CH,
VAR_CHID,
VAR_CHS,
VAR_CHLAYOUT,
VAR_NB
};
static int generate_kernel(AVFilterContext *ctx, const char *gain, const char *gain_entry)
{
FIREqualizerContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
const char *gain_entry_func_names[] = { "entry", NULL };
const char *gain_func_names[] = { "gain_interpolate", NULL };
double (*gain_entry_funcs[])(void *, double, double) = { entry_func, NULL };
double (*gain_funcs[])(void *, double) = { gain_interpolate_func, NULL };
double vars[VAR_NB];
AVExpr *gain_expr;
int ret, k, center, ch;
s->nb_gain_entry = 0;
s->gain_entry_err = 0;
if (gain_entry) {
double result = 0.0;
ret = av_expr_parse_and_eval(&result, gain_entry, NULL, NULL, NULL, NULL,
gain_entry_func_names, gain_entry_funcs, ctx, 0, ctx);
if (ret < 0)
return ret;
if (s->gain_entry_err < 0)
return s->gain_entry_err;
}
av_log(ctx, AV_LOG_DEBUG, "nb_gain_entry = %d.\n", s->nb_gain_entry);
ret = av_expr_parse(&gain_expr, gain, var_names,
gain_func_names, gain_funcs, NULL, NULL, 0, ctx);
if (ret < 0)
return ret;
vars[VAR_CHS] = inlink->channels;
vars[VAR_CHLAYOUT] = inlink->channel_layout;
vars[VAR_SR] = inlink->sample_rate;
for (ch = 0; ch < inlink->channels; ch++) {
vars[VAR_CH] = ch;
vars[VAR_CHID] = av_channel_layout_extract_channel(inlink->channel_layout, ch);
vars[VAR_F] = 0.0;
s->analysis_buf[0] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
vars[VAR_F] = 0.5 * inlink->sample_rate;
s->analysis_buf[1] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
for (k = 1; k < s->analysis_rdft_len/2; k++) {
vars[VAR_F] = k * ((double)inlink->sample_rate /(double)s->analysis_rdft_len);
s->analysis_buf[2*k] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
s->analysis_buf[2*k+1] = 0.0;
}
av_rdft_calc(s->analysis_irdft, s->analysis_buf);
center = s->fir_len / 2;
for (k = 0; k <= center; k++) {
double u = k * (M_PI/center);
double win;
switch (s->wfunc) {
case WFUNC_RECTANGULAR:
win = 1.0;
break;
case WFUNC_HANN:
win = 0.5 + 0.5 * cos(u);
break;
case WFUNC_HAMMING:
win = 0.53836 + 0.46164 * cos(u);
break;
case WFUNC_BLACKMAN:
win = 0.48 + 0.5 * cos(u) + 0.02 * cos(2*u);
break;
case WFUNC_NUTTALL3:
win = 0.40897 + 0.5 * cos(u) + 0.09103 * cos(2*u);
break;
case WFUNC_MNUTTALL3:
win = 0.4243801 + 0.4973406 * cos(u) + 0.0782793 * cos(2*u);
break;
case WFUNC_NUTTALL:
win = 0.355768 + 0.487396 * cos(u) + 0.144232 * cos(2*u) + 0.012604 * cos(3*u);
break;
case WFUNC_BNUTTALL:
win = 0.3635819 + 0.4891775 * cos(u) + 0.1365995 * cos(2*u) + 0.0106411 * cos(3*u);
break;
case WFUNC_BHARRIS:
win = 0.35875 + 0.48829 * cos(u) + 0.14128 * cos(2*u) + 0.01168 * cos(3*u);
break;
default:
av_assert0(0);
}
s->analysis_buf[k] *= (2.0/s->analysis_rdft_len) * (2.0/s->rdft_len) * win;
}
for (k = 0; k < center - k; k++) {
float tmp = s->analysis_buf[k];
s->analysis_buf[k] = s->analysis_buf[center - k];
s->analysis_buf[center - k] = tmp;
}
for (k = 1; k <= center; k++)
s->analysis_buf[center + k] = s->analysis_buf[center - k];
memset(s->analysis_buf + s->fir_len, 0, (s->rdft_len - s->fir_len) * sizeof(*s->analysis_buf));
av_rdft_calc(s->rdft, s->analysis_buf);
for (k = 0; k < s->rdft_len; k++) {
if (isnan(s->analysis_buf[k]) || isinf(s->analysis_buf[k])) {
av_log(ctx, AV_LOG_ERROR, "filter kernel contains nan or infinity.\n");
av_expr_free(gain_expr);
return AVERROR(EINVAL);
}
}
memcpy(s->kernel_tmp_buf + ch * s->rdft_len, s->analysis_buf, s->rdft_len * sizeof(*s->analysis_buf));
if (!s->multi)
break;
}
memcpy(s->kernel_buf, s->kernel_tmp_buf, (s->multi ? inlink->channels : 1) * s->rdft_len * sizeof(*s->kernel_buf));
av_expr_free(gain_expr);
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
FIREqualizerContext *s = ctx->priv;
int rdft_bits;
common_uninit(s);
s->next_pts = 0;
s->frame_nsamples_max = 0;
s->fir_len = FFMAX(2 * (int)(inlink->sample_rate * s->delay) + 1, 3);
s->remaining = s->fir_len - 1;
for (rdft_bits = RDFT_BITS_MIN; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
s->rdft_len = 1 << rdft_bits;
s->nsamples_max = s->rdft_len - s->fir_len + 1;
if (s->nsamples_max * 2 >= s->fir_len)
break;
}
if (rdft_bits > RDFT_BITS_MAX) {
av_log(ctx, AV_LOG_ERROR, "too large delay, please decrease it.\n");
return AVERROR(EINVAL);
}
if (!(s->rdft = av_rdft_init(rdft_bits, DFT_R2C)) || !(s->irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
return AVERROR(ENOMEM);
for ( ; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
s->analysis_rdft_len = 1 << rdft_bits;
if (inlink->sample_rate <= s->accuracy * s->analysis_rdft_len)
break;
}
if (rdft_bits > RDFT_BITS_MAX) {
av_log(ctx, AV_LOG_ERROR, "too small accuracy, please increase it.\n");
return AVERROR(EINVAL);
}
if (!(s->analysis_irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
return AVERROR(ENOMEM);
s->analysis_buf = av_malloc_array(s->analysis_rdft_len, sizeof(*s->analysis_buf));
s->kernel_tmp_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_tmp_buf));
s->kernel_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_buf));
s->conv_buf = av_calloc(2 * s->rdft_len * inlink->channels, sizeof(*s->conv_buf));
s->conv_idx = av_calloc(inlink->channels, sizeof(*s->conv_idx));
if (!s->analysis_buf || !s->kernel_tmp_buf || !s->kernel_buf || !s->conv_buf || !s->conv_idx)
return AVERROR(ENOMEM);
av_log(ctx, AV_LOG_DEBUG, "sample_rate = %d, channels = %d, analysis_rdft_len = %d, rdft_len = %d, fir_len = %d, nsamples_max = %d.\n",
inlink->sample_rate, inlink->channels, s->analysis_rdft_len, s->rdft_len, s->fir_len, s->nsamples_max);
if (s->fixed)
inlink->min_samples = inlink->max_samples = inlink->partial_buf_size = s->nsamples_max;
return generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain,
s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
FIREqualizerContext *s = ctx->priv;
int ch;
for (ch = 0; ch < inlink->channels; ch++) {
fast_convolute(s, s->kernel_buf + (s->multi ? ch * s->rdft_len : 0),
s->conv_buf + 2 * ch * s->rdft_len, s->conv_idx + ch,
(float *) frame->extended_data[ch], frame->nb_samples);
}
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, av_make_q(1, inlink->sample_rate), inlink->time_base);
s->frame_nsamples_max = FFMAX(s->frame_nsamples_max, frame->nb_samples);
return ff_filter_frame(ctx->outputs[0], frame);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
FIREqualizerContext *s= ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && s->remaining > 0 && s->frame_nsamples_max > 0) {
AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(s->remaining, s->frame_nsamples_max));
if (!frame)
return AVERROR(ENOMEM);
av_samples_set_silence(frame->extended_data, 0, frame->nb_samples, outlink->channels, frame->format);
frame->pts = s->next_pts;
s->remaining -= frame->nb_samples;
ret = filter_frame(ctx->inputs[0], frame);
}
return ret;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
FIREqualizerContext *s = ctx->priv;
int ret = AVERROR(ENOSYS);
if (!strcmp(cmd, "gain")) {
char *gain_cmd;
gain_cmd = av_strdup(args);
if (!gain_cmd)
return AVERROR(ENOMEM);
ret = generate_kernel(ctx, gain_cmd, s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
if (ret >= 0) {
av_freep(&s->gain_cmd);
s->gain_cmd = gain_cmd;
} else {
av_freep(&gain_cmd);
}
} else if (!strcmp(cmd, "gain_entry")) {
char *gain_entry_cmd;
gain_entry_cmd = av_strdup(args);
if (!gain_entry_cmd)
return AVERROR(ENOMEM);
ret = generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain, gain_entry_cmd);
if (ret >= 0) {
av_freep(&s->gain_entry_cmd);
s->gain_entry_cmd = gain_entry_cmd;
} else {
av_freep(&gain_entry_cmd);
}
}
return ret;
}
static const AVFilterPad firequalizer_inputs[] = {
{
.name = "default",
.config_props = config_input,
.filter_frame = filter_frame,
.type = AVMEDIA_TYPE_AUDIO,
.needs_writable = 1,
},
{ NULL }
};
static const AVFilterPad firequalizer_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_firequalizer = {
.name = "firequalizer",
.description = NULL_IF_CONFIG_SMALL("Finite Impulse Response Equalizer"),
.uninit = uninit,
.query_formats = query_formats,
.process_command = process_command,
.priv_size = sizeof(FIREqualizerContext),
.inputs = firequalizer_inputs,
.outputs = firequalizer_outputs,
.priv_class = &firequalizer_class,
};

1
libavfilter/allfilters.c
View File

@ -101,6 +101,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);
REGISTER_FILTER(EXTRASTEREO, extrastereo, af);
REGISTER_FILTER(FIREQUALIZER, firequalizer, af);
REGISTER_FILTER(FLANGER, flanger, af);
REGISTER_FILTER(HIGHPASS, highpass, af);
REGISTER_FILTER(JOIN, join, af);

2
libavfilter/version.h
View File

@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 34
#define LIBAVFILTER_VERSION_MINOR 35
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \