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[209.85.166.44]) by smtp.gmail.com with ESMTPSA id x7sm338054ilp.88.2022.02.06.12.49.37 for (version=TLS1_3 cipher=TLS_AES_128_GCM_SHA256 bits=128/128); Sun, 06 Feb 2022 12:49:37 -0800 (PST) Received: by mail-io1-f44.google.com with SMTP id m185so6147944iof.10 for ; Sun, 06 Feb 2022 12:49:37 -0800 (PST) X-Received: by 2002:a5d:88c1:: with SMTP id i1mr4146700iol.154.1644180576961; Sun, 06 Feb 2022 12:49:36 -0800 (PST) MIME-Version: 1.0 References: <20220206195310.1434953-1-onemda@gmail.com> In-Reply-To: <20220206195310.1434953-1-onemda@gmail.com> From: Pierre-Anthony Lemieux Date: Sun, 6 Feb 2022 12:49:25 -0800 X-Gmail-Original-Message-ID: Message-ID: To: FFmpeg development discussions and patches Subject: Re: [FFmpeg-devel] [PATCH] avfilter: add dialogue enhance audio filter X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Archived-At: List-Archive: List-Post: On Sun, Feb 6, 2022 at 11:52 AM Paul B Mahol wrote: > > Signed-off-by: Paul B Mahol > --- > doc/filters.texi | 28 +++ > libavfilter/Makefile | 1 + > libavfilter/af_dialoguenhance.c | 407 ++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 4 files changed, 437 insertions(+) > create mode 100644 libavfilter/af_dialoguenhance.c > > diff --git a/doc/filters.texi b/doc/filters.texi > index 04c34cb1fb..10c11c1f55 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -4178,6 +4178,34 @@ Default value is @var{o}. > > @end table > > +@section dialoguenhance > +Enhance dialogue in stereo audio. I suggest adding a link to an explainer/article and/or including an overview description of the algorithm. > + > +This filter accepts stereo input and produce surround (3.0) channels output. > +The newly produced front center channel have enhanced speech dialogue originally > +available in both stereo channels. > +This filter outputs front left and front right channels same as available in stereo input. > + > +The filter accepts the following options: > + > +@table @option > +@item original > +Set the original center factor to keep in front center channel output. > +Allowed range is from 0 to 1. Default value is 1. > + > +@item enhance > +Set the dialogue enhance factor to put in front center channel output. > +Allowed range is from 0 to 3. Default value is 1. > + > +@item voice > +Set the voice detection factor. > +Allowed range is from 2 to 32. Default value is 2. > +@end table > + > +@subsection Commands > + > +This filter supports the all above options as @ref{commands}. > + > @section drmeter > Measure audio dynamic range. > > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 282967144b..56d33e6480 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -124,6 +124,7 @@ OBJS-$(CONFIG_CROSSFEED_FILTER) += af_crossfeed.o > OBJS-$(CONFIG_CRYSTALIZER_FILTER) += af_crystalizer.o > OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o > OBJS-$(CONFIG_DEESSER_FILTER) += af_deesser.o > +OBJS-$(CONFIG_DIALOGUENHANCE_FILTER) += af_dialoguenhance.o > OBJS-$(CONFIG_DRMETER_FILTER) += af_drmeter.o > OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o > OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o > diff --git a/libavfilter/af_dialoguenhance.c b/libavfilter/af_dialoguenhance.c > new file mode 100644 > index 0000000000..87cf131320 > --- /dev/null > +++ b/libavfilter/af_dialoguenhance.c > @@ -0,0 +1,407 @@ > +/* > + * Copyright (c) 2022 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/channel_layout.h" > +#include "libavutil/opt.h" > +#include "libavutil/tx.h" > +#include "audio.h" > +#include "avfilter.h" > +#include "filters.h" > +#include "internal.h" > +#include "window_func.h" > + > +#include > + > +typedef struct AudioDialogueEnhancementContext { > + const AVClass *class; > + > + double original, enhance, voice; > + > + int fft_size; > + int overlap; > + > + float *window; > + float prev_vad; > + > + AVFrame *in; > + AVFrame *in_frame; > + AVFrame *out_dist_frame; > + AVFrame *windowed_frame; > + AVFrame *windowed_out; > + AVFrame *windowed_prev; > + AVFrame *center_frame; > + > + AVTXContext *tx_ctx[2], *itx_ctx; > + av_tx_fn tx_fn, itx_fn; > +} AudioDialogueEnhanceContext; > + > +#define OFFSET(x) offsetof(AudioDialogueEnhanceContext, x) > +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM > + > +static const AVOption dialoguenhance_options[] = { > + { "original", "set original center factor", OFFSET(original), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS }, > + { "enhance", "set dialog enhance factor", OFFSET(enhance), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 3, FLAGS }, > + { "voice", "set voice detection factor", OFFSET(voice), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 2,32, FLAGS }, > + {NULL} > +}; > + > +AVFILTER_DEFINE_CLASS(dialoguenhance); > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats = NULL; > + AVFilterChannelLayouts *in_layout = NULL, *out_layout = NULL; > + int ret; > + > + if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_FLTP )) < 0 || > + (ret = ff_set_common_formats (ctx , formats )) < 0 || > + (ret = ff_add_channel_layout (&in_layout , AV_CH_LAYOUT_STEREO)) < 0 || > + (ret = ff_channel_layouts_ref(in_layout, &ctx->inputs[0]->outcfg.channel_layouts)) < 0 || > + (ret = ff_add_channel_layout (&out_layout , AV_CH_LAYOUT_SURROUND)) < 0 || > + (ret = ff_channel_layouts_ref(out_layout, &ctx->outputs[0]->incfg.channel_layouts)) < 0) > + return ret; > + > + return ff_set_common_all_samplerates(ctx); > +} > + > +static int config_input(AVFilterLink *inlink) > +{ > + AVFilterContext *ctx = inlink->dst; > + AudioDialogueEnhanceContext *s = ctx->priv; > + float scale = 1.f, iscale, overlap; > + int ret; > + > + s->fft_size = inlink->sample_rate > 100000 ? 8192 : inlink->sample_rate > 50000 ? 4096 : 2048; > + s->overlap = s->fft_size / 4; > + > + s->window = av_calloc(s->fft_size, sizeof(*s->window)); > + if (!s->window) > + return AVERROR(ENOMEM); > + > + s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 4); > + s->center_frame = ff_get_audio_buffer(inlink, s->fft_size * 4); > + s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 4); > + s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 4); > + s->windowed_out = ff_get_audio_buffer(inlink, s->fft_size * 4); > + s->windowed_prev = ff_get_audio_buffer(inlink, s->fft_size * 4); > + if (!s->in_frame || !s->windowed_out || !s->windowed_prev || > + !s->out_dist_frame || !s->windowed_frame || !s->center_frame) > + return AVERROR(ENOMEM); > + > + generate_window_func(s->window, s->fft_size, WFUNC_SINE, &overlap); > + > + iscale = 1.f / s->fft_size; > + > + ret = av_tx_init(&s->tx_ctx[0], &s->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_size, &scale, 0); > + if (ret < 0) > + return ret; > + > + ret = av_tx_init(&s->tx_ctx[1], &s->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_size, &scale, 0); > + if (ret < 0) > + return ret; > + > + ret = av_tx_init(&s->itx_ctx, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, s->fft_size, &iscale, 0); > + if (ret < 0) > + return ret; > + > + return 0; > +} > + > +static void apply_window(AudioDialogueEnhanceContext *s, > + const float *in_frame, float *out_frame, const int add_to_out_frame) > +{ > + const float *window = s->window; > + > + if (add_to_out_frame) { > + for (int i = 0; i < s->fft_size; i++) > + out_frame[i] += in_frame[i] * window[i]; > + } else { > + for (int i = 0; i < s->fft_size; i++) > + out_frame[i] = in_frame[i] * window[i]; > + } > +} > + > +static float sqrf(float x) > +{ > + return x * x; > +} > + > +static void get_centere(AVComplexFloat *left, AVComplexFloat *right, > + AVComplexFloat *center, int N) > +{ > + for (int i = 0; i < N; i++) { > + const float l_re = left[i].re; > + const float l_im = left[i].im; > + const float r_re = right[i].re; > + const float r_im = right[i].im; > + const float a = 0.5f * (1.f - sqrtf((sqrf(l_re - r_re) + sqrf(l_im - r_im))/ > + (sqrf(l_re + r_re) + sqrf(l_im + r_im) + FLT_EPSILON))); > + > + center[i].re = a * (l_re + r_re); > + center[i].im = a * (l_im + r_im); > + } > +} > + > +static float flux(float *curf, float *prevf, int N) > +{ > + AVComplexFloat *cur = (AVComplexFloat *)curf; > + AVComplexFloat *prev = (AVComplexFloat *)prevf; > + float sum = 0.f; > + > + for (int i = 0; i < N; i++) { > + float c_re = cur[i].re; > + float c_im = cur[i].im; > + float p_re = prev[i].re; > + float p_im = prev[i].im; > + > + sum += sqrf(hypotf(c_re, c_im) - hypotf(p_re, p_im)); > + } > + > + return sum; > +} > + > +static float fluxlr(float *lf, float *lpf, > + float *rf, float *rpf, > + int N) > +{ > + AVComplexFloat *l = (AVComplexFloat *)lf; > + AVComplexFloat *lp = (AVComplexFloat *)lpf; > + AVComplexFloat *r = (AVComplexFloat *)rf; > + AVComplexFloat *rp = (AVComplexFloat *)rpf; > + float sum = 0.f; > + > + for (int i = 0; i < N; i++) { > + float c_re = l[i].re - r[i].re; > + float c_im = l[i].im - r[i].im; > + float p_re = lp[i].re - rp[i].re; > + float p_im = lp[i].im - rp[i].im; > + > + sum += sqrf(hypotf(c_re, c_im) - hypotf(p_re, p_im)); > + } > + > + return sum; > +} > + > +static float calc_vad(float fc, float flr, float a) > +{ > + const float vad = a * (fc / (fc + flr) - 0.5f); > + > + return av_clipf(vad, 0.f, 1.f); > +} > + > +static void get_final(float *c, float *l, > + float *r, float vad, int N, > + float original, float enhance) > +{ > + AVComplexFloat *center = (AVComplexFloat *)c; > + AVComplexFloat *left = (AVComplexFloat *)l; > + AVComplexFloat *right = (AVComplexFloat *)r; > + > + for (int i = 0; i < N; i++) { > + float cP = sqrf(center[i].re) + sqrf(center[i].im); > + float lrP = sqrf(left[i].re - right[i].re) + sqrf(left[i].im - right[i].im); > + float G = cP / (cP + lrP + FLT_EPSILON); > + float re, im; > + > + re = center[i].re * (original + vad * G * enhance); > + im = center[i].im * (original + vad * G * enhance); > + > + center[i].re = re; > + center[i].im = im; > + } > +} > + > +static int de_stereo(AVFilterContext *ctx, AVFrame *out) > +{ > + AudioDialogueEnhanceContext *s = ctx->priv; > + float *center = (float *)s->center_frame->extended_data[0]; > + float *center_prev = (float *)s->center_frame->extended_data[1]; > + float *left_in = (float *)s->in_frame->extended_data[0]; > + float *right_in = (float *)s->in_frame->extended_data[1]; > + float *left_out = (float *)s->out_dist_frame->extended_data[0]; > + float *right_out = (float *)s->out_dist_frame->extended_data[1]; > + float *left_samples = (float *)s->in->extended_data[0]; > + float *right_samples = (float *)s->in->extended_data[1]; > + float *windowed_left = (float *)s->windowed_frame->extended_data[0]; > + float *windowed_right = (float *)s->windowed_frame->extended_data[1]; > + float *windowed_oleft = (float *)s->windowed_out->extended_data[0]; > + float *windowed_oright = (float *)s->windowed_out->extended_data[1]; > + float *windowed_pleft = (float *)s->windowed_prev->extended_data[0]; > + float *windowed_pright = (float *)s->windowed_prev->extended_data[1]; > + float *left_osamples = (float *)out->extended_data[0]; > + float *right_osamples = (float *)out->extended_data[1]; > + float *center_osamples = (float *)out->extended_data[2]; > + const int offset = s->fft_size - s->overlap; > + float vad; > + > + // shift in/out buffers > + memmove(left_in, &left_in[s->overlap], offset * sizeof(float)); > + memmove(right_in, &right_in[s->overlap], offset * sizeof(float)); > + memmove(left_out, &left_out[s->overlap], offset * sizeof(float)); > + memmove(right_out, &right_out[s->overlap], offset * sizeof(float)); > + > + memcpy(&left_in[offset], left_samples, s->overlap * sizeof(float)); > + memcpy(&right_in[offset], right_samples, s->overlap * sizeof(float)); > + memset(&left_out[offset], 0, s->overlap * sizeof(float)); > + memset(&right_out[offset], 0, s->overlap * sizeof(float)); > + > + apply_window(s, left_in, windowed_left, 0); > + apply_window(s, right_in, windowed_right, 0); > + > + s->tx_fn(s->tx_ctx[0], windowed_oleft, windowed_left, sizeof(float)); > + s->tx_fn(s->tx_ctx[1], windowed_oright, windowed_right, sizeof(float)); > + > + get_centere((AVComplexFloat *)windowed_oleft, > + (AVComplexFloat *)windowed_oright, > + (AVComplexFloat *)center, > + s->fft_size / 2 + 1); > + > + vad = calc_vad(flux(center, center_prev, s->fft_size / 2 + 1), > + fluxlr(windowed_oleft, windowed_pleft, > + windowed_oright, windowed_pright, s->fft_size / 2 + 1), s->voice); > + vad = vad * 0.1 + 0.9 * s->prev_vad; > + s->prev_vad = vad; > + > + memcpy(center_prev, center, s->fft_size * sizeof(float)); > + memcpy(windowed_pleft, windowed_oleft, s->fft_size * sizeof(float)); > + memcpy(windowed_pright, windowed_oright, s->fft_size * sizeof(float)); > + > + get_final(center, windowed_oleft, windowed_oright, vad, s->fft_size / 2 + 1, > + s->original, s->enhance); > + > + s->itx_fn(s->itx_ctx, windowed_oleft, center, sizeof(float)); > + > + apply_window(s, windowed_oleft, left_out, 1); > + > + for (int i = 0; i < s->overlap; i++) { > + // 4 times overlap with squared hanning window results in 1.5 time increase in amplitude > + if (!ctx->is_disabled) > + center_osamples[i] = left_out[i] / 1.5f; > + else > + center_osamples[i] = 0.f; > + left_osamples[i] = left_in[i]; > + right_osamples[i] = right_in[i]; > + } > + > + return 0; > +} > + > +static int filter_frame(AVFilterLink *inlink, AVFrame *in) > +{ > + AVFilterContext *ctx = inlink->dst; > + AVFilterLink *outlink = ctx->outputs[0]; > + AudioDialogueEnhanceContext *s = ctx->priv; > + AVFrame *out; > + int ret; > + > + out = ff_get_audio_buffer(outlink, s->overlap); > + if (!out) { > + ret = AVERROR(ENOMEM); > + goto fail; > + } > + > + s->in = in; > + de_stereo(ctx, out); > + > + out->pts = in->pts; > + out->nb_samples = in->nb_samples; > + ret = ff_filter_frame(outlink, out); > +fail: > + av_frame_free(&in); > + s->in = NULL; > + return ret < 0 ? ret : 0; > +} > + > +static int activate(AVFilterContext *ctx) > +{ > + AVFilterLink *inlink = ctx->inputs[0]; > + AVFilterLink *outlink = ctx->outputs[0]; > + AudioDialogueEnhanceContext *s = ctx->priv; > + AVFrame *in = NULL; > + int ret = 0, status; > + int64_t pts; > + > + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); > + > + ret = ff_inlink_consume_samples(inlink, s->overlap, s->overlap, &in); > + if (ret < 0) > + return ret; > + > + if (ret > 0) { > + return filter_frame(inlink, in); > + } else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { > + ff_outlink_set_status(outlink, status, pts); > + return 0; > + } else { > + if (ff_inlink_queued_samples(inlink) >= s->overlap) { > + ff_filter_set_ready(ctx, 10); > + } else if (ff_outlink_frame_wanted(outlink)) { > + ff_inlink_request_frame(inlink); > + } > + return 0; > + } > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + AudioDialogueEnhanceContext *s = ctx->priv; > + > + av_freep(&s->window); > + > + av_frame_free(&s->in_frame); > + av_frame_free(&s->center_frame); > + av_frame_free(&s->out_dist_frame); > + av_frame_free(&s->windowed_frame); > + av_frame_free(&s->windowed_out); > + av_frame_free(&s->windowed_prev); > + > + av_tx_uninit(&s->tx_ctx[0]); > + av_tx_uninit(&s->tx_ctx[1]); > + av_tx_uninit(&s->itx_ctx); > +} > + > +static const AVFilterPad inputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + .config_props = config_input, > + }, > +}; > + > +static const AVFilterPad outputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + }, > +}; > + > +const AVFilter ff_af_dialoguenhance = { > + .name = "dialoguenhance", > + .description = NULL_IF_CONFIG_SMALL("Audio Dialogue Enhancement."), > + .priv_size = sizeof(AudioDialogueEnhanceContext), > + .priv_class = &dialoguenhance_class, > + .uninit = uninit, > + FILTER_INPUTS(inputs), > + FILTER_OUTPUTS(outputs), > + FILTER_QUERY_FUNC(query_formats), > + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, > + .activate = activate, > + .process_command = ff_filter_process_command, > +}; > diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c > index 714468afce..f5caee3a62 100644 > --- a/libavfilter/allfilters.c > +++ b/libavfilter/allfilters.c > @@ -115,6 +115,7 @@ extern const AVFilter ff_af_crossfeed; > extern const AVFilter ff_af_crystalizer; > extern const AVFilter ff_af_dcshift; > extern const AVFilter ff_af_deesser; > +extern const AVFilter ff_af_dialoguenhance; > extern const AVFilter ff_af_drmeter; > extern const AVFilter ff_af_dynaudnorm; > extern const AVFilter ff_af_earwax; > -- > 2.33.0 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".