[FFmpeg-devel] [PATCH] avfilter: add dialogue enhance audio filter

Pierre-Anthony Lemieux pal at sandflow.com
Sun Feb 6 22:49:25 EET 2022


On Sun, Feb 6, 2022 at 11:52 AM Paul B Mahol <onemda at gmail.com> wrote:
>
> 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.

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:
> +
> + 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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