[FFmpeg-devel] [PATCH] avfilter: add dialogue enhance audio filter
Paul B Mahol
onemda at gmail.com
Sun Feb 6 21:53:10 EET 2022
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
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
+ at section dialoguenhance
+Enhance dialogue in stereo audio.
+
+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:
+
+ at table @option
+ at 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.
+
+ at 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.
+
+ at item voice
+Set the voice detection factor.
+Allowed range is from 2 to 32. Default value is 2.
+ at end table
+
+ at 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 <float.h>
+
+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
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