From 8baaed7889f34369e26c52bdd379894b0198e3b0 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Tue, 19 Dec 2017 11:15:11 +0100 Subject: [PATCH] avfilter: add sinc source filter Signed-off-by: Paul B Mahol --- Changelog | 1 + doc/filters.texi | 43 ++++ libavfilter/Makefile | 1 + libavfilter/allfilters.c | 1 + libavfilter/asrc_sinc.c | 455 +++++++++++++++++++++++++++++++++++++++ libavfilter/version.h | 4 +- 6 files changed, 503 insertions(+), 2 deletions(-) create mode 100644 libavfilter/asrc_sinc.c diff --git a/Changelog b/Changelog index 4a22ab4cb3..1b0bc95b7a 100644 --- a/Changelog +++ b/Changelog @@ -34,6 +34,7 @@ version : - audio denoiser as afftdn filter - AV1 parser - SER demuxer +- sinc audio filter source version 4.0: diff --git a/doc/filters.texi b/doc/filters.texi index cadf78c93c..54b85c4bb9 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -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. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 62cc2f561f..b03b2457eb 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -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 diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 5e72803b13..725bac94a0 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -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; diff --git a/libavfilter/asrc_sinc.c b/libavfilter/asrc_sinc.c new file mode 100644 index 0000000000..0135eb9023 --- /dev/null +++ b/libavfilter/asrc_sinc.c @@ -0,0 +1,455 @@ +/* + * Copyright (c) 2008-2009 Rob Sykes + * 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, +}; diff --git a/libavfilter/version.h b/libavfilter/version.h index 30e961b999..bb57c5fbed 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -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, \