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FFmpeg/libavfilter/asrc_sinc.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

450 lines
14 KiB
C

/*
* 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/channel_layout.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.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 activate(AVFilterContext *ctx)
{
AVFilterLink *outlink = ctx->outputs[0];
SincContext *s = ctx->priv;
const float *coeffs = s->coeffs;
AVFrame *frame = NULL;
int nb_samples;
if (!ff_outlink_frame_wanted(outlink))
return FFERROR_NOT_READY;
nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
if (nb_samples <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
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 };
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
if (ret < 0)
return ret;
return ff_set_common_samplerates_from_list(ctx, sample_rates);
}
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);
/* The first part is for work (+2 for (UN)PACK), the latter for pi_wraps. */
work = av_calloc((work_len + 2) + (work_len / 2 + 1), sizeof(float));
if (!work)
return AVERROR(ENOMEM);
pi_wraps = &work[work_len + 2];
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) {
av_free(work);
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(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(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,
},
};
#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);
const 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,
.uninit = uninit,
.activate = activate,
.inputs = NULL,
FILTER_OUTPUTS(sinc_outputs),
FILTER_QUERY_FUNC(query_formats),
};