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Commit 8baaed78 authored by Paul B Mahol's avatar Paul B Mahol
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avfilter: add sinc source filter 

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent db4771af
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Showing with 503 additions and 2 deletions
Changelog
View file @ 8baaed78
......@@ -34,6 +34,7 @@ version <next>:
- audio denoiser as afftdn filter
- AV1 parser
- SER demuxer
- sinc audio filter source
version 4.0:
......
doc/filters.texi
View file @ 8baaed78
......@@ -5370,6 +5370,49 @@ Set number of samples per each frame.
Set window function to be used when generating FIR coefficients.
@end table
@section sinc
Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients.
The resulting stream can be used with @ref{afir} filter for filtering the audio signal.
The filter accepts the following options:
@table @option
@item sample_rate, r
Set sample rate, default is 44100.
@item nb_samples, n
Set number of samples per each frame. Default is 1024.
@item hp
Set high-pass frequency. Default is 0.
@item lp
Set low-pass frequency. Default is 0.
If high-pass frequency is lower than low-pass frequency and low-pass frequency
is higher than 0 then filter will create band-pass filter coefficients,
otherwise band-reject filter coefficients.
@item phase
Set filter phase response. Default is 50. Allowed range is from 0 to 100.
@item beta
Set Kaiser window beta.
@item att
Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.
@item round
Enable rounding, by default is disabled.
@item hptaps
Set number of taps for high-pass filter.
@item lptaps
Set number of taps for low-pass filter.
@end table
@section sine
Generate an audio signal made of a sine wave with amplitude 1/8.
......
libavfilter/Makefile
View file @ 8baaed78
......@@ -141,6 +141,7 @@ OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
OBJS-$(CONFIG_FLITE_FILTER) += asrc_flite.o
OBJS-$(CONFIG_HILBERT_FILTER) += asrc_hilbert.o
OBJS-$(CONFIG_SINC_FILTER) += asrc_sinc.o
OBJS-$(CONFIG_SINE_FILTER) += asrc_sine.o
OBJS-$(CONFIG_ANULLSINK_FILTER) += asink_anullsink.o
......
libavfilter/allfilters.c
View file @ 8baaed78
......@@ -134,6 +134,7 @@ extern AVFilter ff_asrc_anoisesrc;
extern AVFilter ff_asrc_anullsrc;
extern AVFilter ff_asrc_flite;
extern AVFilter ff_asrc_hilbert;
extern AVFilter ff_asrc_sinc;
extern AVFilter ff_asrc_sine;
extern AVFilter ff_asink_anullsink;
......
libavfilter/asrc_sinc.c 0 → 100644
View file @ 8baaed78
/*
* Copyright (c) 2008-2009 Rob Sykes <robs@users.sourceforge.net>
* Copyright (c) 2017 Paul B Mahol
*
* 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/avassert.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct SincContext {
const AVClass *class;
int sample_rate, nb_samples;
float att, beta, phase, Fc0, Fc1, tbw0, tbw1;
int num_taps[2];
int round;
int n, rdft_len;
float *coeffs;
int64_t pts;
RDFTContext *rdft, *irdft;
} SincContext;
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SincContext *s = ctx->priv;
const float *coeffs = s->coeffs;
AVFrame *frame = NULL;
int nb_samples;
nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
if (nb_samples <= 0)
return AVERROR_EOF;
if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
return AVERROR(ENOMEM);
memcpy(frame->data[0], coeffs + s->pts, nb_samples * sizeof(float));
frame->pts = s->pts;
s->pts += nb_samples;
return ff_filter_frame(outlink, frame);
}
static int query_formats(AVFilterContext *ctx)
{
SincContext *s = ctx->priv;
static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
int sample_rates[] = { s->sample_rate, -1 };
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE };
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
int 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;
layouts = avfilter_make_format64_list(chlayouts);
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_rates);
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static float bessel_I_0(float x)
{
float term = 1, sum = 1, last_sum, x2 = x / 2;
int i = 1;
do {
float y = x2 / i++;
last_sum = sum;
sum += term *= y * y;
} while (sum != last_sum);
return sum;
}
static float *make_lpf(int num_taps, float Fc, float beta, float rho,
float scale, int dc_norm)
{
int i, m = num_taps - 1;
float *h = av_calloc(num_taps, sizeof(*h)), sum = 0;
float mult = scale / bessel_I_0(beta), mult1 = 1.f / (.5f * m + rho);
av_assert0(Fc >= 0 && Fc <= 1);
for (i = 0; i <= m / 2; i++) {
float z = i - .5f * m, x = z * M_PI, y = z * mult1;
h[i] = x ? sinf(Fc * x) / x : Fc;
sum += h[i] *= bessel_I_0(beta * sqrtf(1.f - y * y)) * mult;
if (m - i != i) {
h[m - i] = h[i];
sum += h[i];
}
}
for (i = 0; dc_norm && i < num_taps; i++)
h[i] *= scale / sum;
return h;
}
static float kaiser_beta(float att, float tr_bw)
{
if (att >= 60.f) {
static const float coefs[][4] = {
{-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
{-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
{-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
{-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
{8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
{9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
{-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
{-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
{1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
{-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
};
float realm = logf(tr_bw / .0005f) / logf(2.f);
float const *c0 = coefs[av_clip((int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
float const *c1 = coefs[av_clip(1 + (int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
float b1 = ((c1[0] * att + c1[1]) * att + c1[2]) * att + c1[3];
return b0 + (b1 - b0) * (realm - (int)realm);
}
if (att > 50.f)
return .1102f * (att - 8.7f);
if (att > 20.96f)
return .58417f * powf(att - 20.96f, .4f) + .07886f * (att - 20.96f);
return 0;
}
static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
{
*beta = *beta < 0.f ? kaiser_beta(att, tr_bw * .5f / Fc): *beta;
att = att < 60.f ? (att - 7.95f) / (2.285f * M_PI * 2.f) :
((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022f) * *beta + .06186902f;
*num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
}
static float *lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
{
int n = *num_taps;
if ((Fc /= Fn) <= 0.f || Fc >= 1.f) {
*num_taps = 0;
return NULL;
}
att = att ? att : 120.f;
kaiser_params(att, Fc, (tbw ? tbw / Fn : .05f) * .5f, beta, num_taps);
if (!n) {
n = *num_taps;
*num_taps = av_clip(n, 11, 32767);
if (round)
*num_taps = 1 + 2 * (int)((int)((*num_taps / 2) * Fc + .5f) / Fc + .5f);
}
return make_lpf(*num_taps |= 1, Fc, *beta, 0.f, 1.f, 0);
}
static void invert(float *h, int n)
{
for (int i = 0; i < n; i++)
h[i] = -h[i];
h[(n - 1) / 2] += 1;
}
#define PACK(h, n) h[1] = h[n]
#define UNPACK(h, n) h[n] = h[1], h[n + 1] = h[1] = 0;
#define SQR(a) ((a) * (a))
static float safe_log(float x)
{
av_assert0(x >= 0);
if (x)
return logf(x);
return -26;
}
static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
{
float *pi_wraps, *work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.f;
int i, work_len, begin, end, imp_peak = 0, peak = 0;
float imp_sum = 0, peak_imp_sum = 0;
float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
work = av_calloc(work_len + 2, sizeof(*work)); /* +2: (UN)PACK */
pi_wraps = av_calloc(((work_len + 2) / 2), sizeof(*pi_wraps));
if (!work || !pi_wraps)
return AVERROR(ENOMEM);
memcpy(work, *h, *len * sizeof(*work));
av_rdft_end(s->rdft);
av_rdft_end(s->irdft);
s->rdft = s->irdft = NULL;
s->rdft = av_rdft_init(av_log2(work_len), DFT_R2C);
s->irdft = av_rdft_init(av_log2(work_len), IDFT_C2R);
if (!s->rdft || !s->irdft)
return AVERROR(ENOMEM);
av_rdft_calc(s->rdft, work); /* Cepstral: */
UNPACK(work, work_len);
for (i = 0; i <= work_len; i += 2) {
float angle = atan2f(work[i + 1], work[i]);
float detect = 2 * M_PI;
float delta = angle - prev_angle2;
float adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
prev_angle2 = angle;
cum_2pi += adjust;
angle += cum_2pi;
detect = M_PI;
delta = angle - prev_angle1;
adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
prev_angle1 = angle;
cum_1pi += fabsf(adjust); /* fabs for when 2pi and 1pi have combined */
pi_wraps[i >> 1] = cum_1pi;
work[i] = safe_log(sqrtf(SQR(work[i]) + SQR(work[i + 1])));
work[i + 1] = 0;
}
PACK(work, work_len);
av_rdft_calc(s->irdft, work);
for (i = 0; i < work_len; i++)
work[i] *= 2.f / work_len;
for (i = 1; i < work_len / 2; i++) { /* Window to reject acausal components */
work[i] *= 2;
work[i + work_len / 2] = 0;
}
av_rdft_calc(s->rdft, work);
for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
work[i + 1] = phase1 * i / work_len * pi_wraps[work_len >> 1] + (1 - phase1) * (work[i + 1] + pi_wraps[i >> 1]) - pi_wraps[i >> 1];
work[0] = exp(work[0]);
work[1] = exp(work[1]);
for (i = 2; i < work_len; i += 2) {
float x = expf(work[i]);
work[i ] = x * cosf(work[i + 1]);
work[i + 1] = x * sinf(work[i + 1]);
}
av_rdft_calc(s->irdft, work);
for (i = 0; i < work_len; i++)
work[i] *= 2.f / work_len;
/* Find peak pos. */
for (i = 0; i <= (int) (pi_wraps[work_len >> 1] / M_PI + .5f); i++) {
imp_sum += work[i];
if (fabs(imp_sum) > fabs(peak_imp_sum)) {
peak_imp_sum = imp_sum;
peak = i;
}
if (work[i] > work[imp_peak]) /* For debug check only */
imp_peak = i;
}
while (peak && fabsf(work[peak - 1]) > fabsf(work[peak]) && (work[peak - 1] * work[peak] > 0)) {
peak--;
}
if (!phase1) {
begin = 0;
} else if (phase1 == 1) {
begin = peak - *len / 2;
} else {
begin = (.997f - (2 - phase1) * .22f) * *len + .5f;
end = (.997f + (0 - phase1) * .22f) * *len + .5f;
begin = peak - (begin & ~3);
end = peak + 1 + ((end + 3) & ~3);
*len = end - begin;
*h = av_realloc_f(*h, *len, sizeof(**h));
if (!*h) {
av_free(pi_wraps);
av_free(work);
return AVERROR(ENOMEM);
}
}
for (i = 0; i < *len; i++) {
(*h)[i] = work[(begin + (phase > 50.f ? *len - 1 - i : i) + work_len) & (work_len - 1)];
}
*post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
av_log(s, AV_LOG_DEBUG, "%d nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)\n",
work_len, pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
work[imp_peak], *len, *post_len, 100.f - 100.f * *post_len / (*len - 1));
av_free(pi_wraps);
av_free(work);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SincContext *s = ctx->priv;
float Fn = s->sample_rate * .5f;
float *h[2];
int i, n, post_peak, longer;
outlink->sample_rate = s->sample_rate;
s->pts = 0;
if (s->Fc0 >= Fn || s->Fc1 >= Fn) {
av_log(ctx, AV_LOG_ERROR,
"filter frequency must be less than %d/2.\n", s->sample_rate);
return AVERROR(EINVAL);
}
h[0] = lpf(Fn, s->Fc0, s->tbw0, &s->num_taps[0], s->att, &s->beta, s->round);
h[1] = lpf(Fn, s->Fc1, s->tbw1, &s->num_taps[1], s->att, &s->beta, s->round);
if (h[0])
invert(h[0], s->num_taps[0]);
longer = s->num_taps[1] > s->num_taps[0];
n = s->num_taps[longer];
if (h[0] && h[1]) {
for (i = 0; i < s->num_taps[!longer]; i++)
h[longer][i + (n - s->num_taps[!longer]) / 2] += h[!longer][i];
if (s->Fc0 < s->Fc1)
invert(h[longer], n);
av_free(h[!longer]);
}
if (s->phase != 50.f) {
int ret = fir_to_phase(s, &h[longer], &n, &post_peak, s->phase);
if (ret < 0)
return ret;
} else {
post_peak = n >> 1;
}
s->n = 1 << (av_log2(n) + 1);
s->rdft_len = 1 << av_log2(n);
s->coeffs = av_calloc(s->n, sizeof(*s->coeffs));
if (!s->coeffs)
return AVERROR(ENOMEM);
for (i = 0; i < n; i++)
s->coeffs[i] = h[longer][i];
av_free(h[longer]);
av_rdft_end(s->rdft);
av_rdft_end(s->irdft);
s->rdft = s->irdft = NULL;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
SincContext *s = ctx->priv;
av_freep(&s->coeffs);
av_rdft_end(s->rdft);
av_rdft_end(s->irdft);
s->rdft = s->irdft = NULL;
}
static const AVFilterPad sinc_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame,
},
{ NULL }
};
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OFFSET(x) offsetof(SincContext, x)
static const AVOption sinc_options[] = {
{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
{ "hp", "set high-pass filter frequency", OFFSET(Fc0), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
{ "lp", "set low-pass filter frequency", OFFSET(Fc1), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
{ "phase", "set filter phase response", OFFSET(phase), AV_OPT_TYPE_FLOAT, {.dbl=50}, 0, 100, AF },
{ "beta", "set kaiser window beta", OFFSET(beta), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 256, AF },
{ "att", "set stop-band attenuation", OFFSET(att), AV_OPT_TYPE_FLOAT, {.dbl=120}, 40, 180, AF },
{ "round", "enable rounding", OFFSET(round), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
{ "hptaps", "set number of taps for high-pass filter", OFFSET(num_taps[0]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
{ "lptaps", "set number of taps for low-pass filter", OFFSET(num_taps[1]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
{ NULL }
};
AVFILTER_DEFINE_CLASS(sinc);
AVFilter ff_asrc_sinc = {
.name = "sinc",
.description = NULL_IF_CONFIG_SMALL("Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
.priv_size = sizeof(SincContext),
.priv_class = &sinc_class,
.query_formats = query_formats,
.uninit = uninit,
.inputs = NULL,
.outputs = sinc_outputs,
};
libavfilter/version.h
View file @ 8baaed78
......@@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 33
#define LIBAVFILTER_VERSION_MICRO 101
#define LIBAVFILTER_VERSION_MINOR 34
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
......
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