Message ID | 20200503191601.15344-1-onemda@gmail.com |
---|---|
State | New |
Headers | show |
Series | [FFmpeg-devel,RFC] avfilter: add speechnorm filter | expand |
Context | Check | Description |
---|---|---|
andriy/default | pending | |
andriy/make | success | Make finished |
andriy/make_fate | success | Make fate finished |
Paul B Mahol: > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > libavfilter/Makefile | 1 + > libavfilter/af_speechnorm.c | 381 ++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 3 files changed, 383 insertions(+) > create mode 100644 libavfilter/af_speechnorm.c > > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index f982afe15f..421a01753e 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -134,6 +134,7 @@ OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o > OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o > OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o > OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o > +OBJS-$(CONFIG_SPEECHNORM_FILTER) += af_speechnorm.o > OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o > OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o > OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o > diff --git a/libavfilter/af_speechnorm.c b/libavfilter/af_speechnorm.c > new file mode 100644 > index 0000000000..52fc8e6e42 > --- /dev/null > +++ b/libavfilter/af_speechnorm.c > @@ -0,0 +1,381 @@ > +/* > + * Speech Normalizer > + * Copyright (c) 2020 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 > + */ > + > +/** > + * @file > + * Speech Normalizer > + */ > + > +#include <float.h> > + > +#include "libavutil/avassert.h" > +#include "libavutil/opt.h" > + > +#define FF_BUFQUEUE_SIZE (1024) > +#include "bufferqueue.h" > + > +#include "audio.h" > +#include "avfilter.h" > +#include "filters.h" > +#include "internal.h" > + > +#define MAX_ITEMS 882000 > + > +typedef struct PeriodItem { > + int size; > + int type; > + double max_peak; > +} PeriodItem; > + > +typedef struct SpeechNormalizerContext { > + const AVClass *class; > + > + double peak_value; > + double max_amplification; > + double threshold_value; > + double feedback; > + double decay; > + int channels; > + > + int max_period; > + int eof; > + int64_t pts; > + int state[12]; > + > + PeriodItem pi[12][MAX_ITEMS]; > + double gain_state[12]; > + int pi_start[12]; > + int pi_end[12]; > + > + struct FFBufQueue queue; > +} SpeechNormalizerContext; > + > +#define OFFSET(x) offsetof(SpeechNormalizerContext, x) > +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM > + > +static const AVOption speechnorm_options[] = { > + { "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, > + { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, > + { "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 2.0}, 1.0, 10.0, FLAGS }, > + { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 2.0}, 1.0, 10.0, FLAGS }, > + { "threshold", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0}, 0.0, 1.0, FLAGS }, > + { "t", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0}, 0.0, 1.0, FLAGS }, > + { "feedback", "set the feedback value", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl = 0.001}, 0.0, 1.0, FLAGS }, > + { "f", "set the feedback value", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl = 0.001}, 0.0, 1.0, FLAGS }, > + { "decay", "set the decay value", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl = 0.999}, 0.0, 1.0, FLAGS }, > + { "d", "set the decay value", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl = 0.999}, 0.0, 1.0, FLAGS }, > + { NULL } > +}; > + > +AVFILTER_DEFINE_CLASS(speechnorm); > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats; > + AVFilterChannelLayouts *layouts; > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_DBLP, > + AV_SAMPLE_FMT_NONE > + }; > + int ret; > + > + layouts = ff_all_channel_counts(); > + if (!layouts) > + return AVERROR(ENOMEM); > + ret = ff_set_common_channel_layouts(ctx, layouts); > + if (ret < 0) > + return 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; > + > + formats = ff_all_samplerates(); > + if (!formats) > + return AVERROR(ENOMEM); > + return ff_set_common_samplerates(ctx, formats); > +} > + > +static int config_input(AVFilterLink *inlink) > +{ > + AVFilterContext *ctx = inlink->dst; > + SpeechNormalizerContext *s = ctx->priv; > + > + s->max_period = inlink->sample_rate / (2 * 20); > + s->channels = inlink->channels; > + for (int ch = 0; ch < s->channels; ch++) > + s->state[ch] = -1; > + > + return 0; > +} > + > +static int get_pi_samples(PeriodItem *pi, int start, int end, int mode) > +{ > + int sum; > + > + if (mode && pi[start].type == 0) > + return 0; > + > + sum = pi[start].size; > + av_assert0(sum > 0); > + while (start != end) { > + start++; > + if (start >= MAX_ITEMS) > + start = 0; > + if (mode && pi[start].type == 0) > + break; > + av_assert0(pi[start].size > 0); > + sum += pi[start].size; > + if (pi[start].type == 0) > + break; > + } > + > + return sum; > +} > + > +static int consume_pi(PeriodItem *pi, int start, int end, int nb_samples) > +{ > + int sum; > + > + sum = pi[start].size; > + av_assert0(pi[start].size > 0); > + while (sum < nb_samples) { > + av_assert0(pi[start].type == 1); > + av_assert0(start != end); > + start++; > + if (start >= MAX_ITEMS) > + start = 0; > + av_assert0(pi[start].size > 0); > + sum += pi[start].size; > + } > + > + av_assert0(pi[start].size >= sum - nb_samples); > + pi[start].size = sum - nb_samples; > + av_assert0(pi[start].size >= 0); > + if (pi[start].size == 0 && start != end) { > + start++; > + if (start >= MAX_ITEMS) > + start = 0; > + } > + > + return start; > +} > + > +static int get_queued_samples(SpeechNormalizerContext *s) > +{ > + int sum = 0; > + > + for (int i = 0; i < s->queue.available; i++) { > + AVFrame *frame = ff_bufqueue_peek(&s->queue, i); > + sum += frame->nb_samples; > + } > + > + return sum; > +} > + > +static int filter_frame(AVFilterContext *ctx) > +{ > + SpeechNormalizerContext *s = ctx->priv; > + AVFilterLink *outlink = ctx->outputs[0]; > + AVFilterLink *inlink = ctx->inputs[0]; > + int min_pi_nb_samples; > + AVFrame *in = NULL; > + int ret; > + > + for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) { > + ret = ff_inlink_consume_frame(inlink, &in); > + if (ret < 0) > + return ret; > + if (ret == 0) > + break; > + > + ff_bufqueue_add(ctx, &s->queue, in); > + > + for (int ch = 0; ch < inlink->channels; ch++) { > + const double *src = (const double *)in->extended_data[ch]; > + int n = 0; > + > + if (s->state[ch] < 0) > + s->state[ch] = src[0] >= 0.; > + > + while (n < in->nb_samples) { > + if (s->state[ch] != (src[n] >= 0.) || s->pi[ch][s->pi_end[ch]].size > s->max_period) { > + s->state[ch] = src[n] >= 0.; > + av_assert0(s->pi[ch][s->pi_end[ch]].size > 0); > + s->pi[ch][s->pi_end[ch]].type = 1; > + s->pi_end[ch]++; > + if (s->pi_end[ch] >= MAX_ITEMS) > + s->pi_end[ch] = 0; > + s->pi[ch][s->pi_end[ch]].max_peak = DBL_MIN; > + s->pi[ch][s->pi_end[ch]].type = 0; > + s->pi[ch][s->pi_end[ch]].size = 0; > + av_assert0(s->pi_end[ch] != s->pi_start[ch]); > + } > + > + if (src[n] >= 0.) { > + while (src[n] >= 0.) { > + s->pi[ch][s->pi_end[ch]].max_peak = FFMAX(s->pi[ch][s->pi_end[ch]].max_peak, FFABS(src[n])); > + s->pi[ch][s->pi_end[ch]].size++; > + n++; > + if (n >= in->nb_samples) > + break; > + } > + } else { > + while (src[n] < 0.) { > + s->pi[ch][s->pi_end[ch]].max_peak = FFMAX(s->pi[ch][s->pi_end[ch]].max_peak, FFABS(src[n])); > + s->pi[ch][s->pi_end[ch]].size++; > + n++; > + if (n >= in->nb_samples) > + break; > + } > + } > + } > + } > + } > + > + if (s->queue.available > 0) { > + in = ff_bufqueue_peek(&s->queue, 0); > + if (!in) > + return 1; > + } else { > + return 1; > + } > + > + min_pi_nb_samples = get_pi_samples(s->pi[0], s->pi_start[0], s->pi_end[0], 1); > + for (int ch = 1; ch < inlink->channels; ch++) { > + min_pi_nb_samples = FFMIN(min_pi_nb_samples, get_pi_samples(s->pi[ch], s->pi_start[ch], s->pi_end[ch], 1)); > + } > + > + if (min_pi_nb_samples >= in->nb_samples) { > + int nb_samples = get_queued_samples(s); > + > + in = ff_bufqueue_get(&s->queue); > + > + av_frame_make_writable(in); > + > + nb_samples -= in->nb_samples; > + > + for (int ch = 0; ch < inlink->channels; ch++) { > + double *src = (double *)in->extended_data[ch]; > + int start = s->pi_start[ch]; > + int offset = 0; > + double gain; > + > + for (int n = 0; n < in->nb_samples; n++) { > + if (n >= offset) { > + if (s->pi[ch][start].max_peak > s->threshold_value) > + gain = FFMIN(s->max_amplification, s->peak_value / s->pi[ch][start].max_peak); > + else > + gain = 1.; > + av_assert0(s->pi[ch][start].size > 0); > + offset += s->pi[ch][start++].size; > + if (start >= MAX_ITEMS) > + start = 0; > + } > + s->gain_state[ch] = FFMIN(gain, gain * s->feedback + s->gain_state[ch] * s->decay); > + src[n] *= s->gain_state[ch]; > + } > + } > + > + for (int ch = 0; ch < inlink->channels; ch++) { > + s->pi_start[ch] = consume_pi(s->pi[ch], s->pi_start[ch], s->pi_end[ch], in->nb_samples); > + } > + > + for (int ch = 0; ch < inlink->channels; ch++) { > + int pi_nb_samples = get_pi_samples(s->pi[ch], s->pi_start[ch], s->pi_end[ch], 0); > + > + if (nb_samples != pi_nb_samples) { > + av_assert0(0); > + } > + } > + > + return ff_filter_frame(outlink, in); > + } > + > + return 1; > +} > + > +static int activate(AVFilterContext *ctx) > +{ > + AVFilterLink *inlink = ctx->inputs[0]; > + AVFilterLink *outlink = ctx->outputs[0]; > + SpeechNormalizerContext *s = ctx->priv; > + int ret = 0, status; > + int64_t pts; > + > + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); > + > + ret = filter_frame(ctx); > + if (ret <= 0) > + return ret; > + > + if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { > + if (status == AVERROR_EOF) > + s->eof = 1; > + } > + > + if (s->eof && ff_inlink_queued_samples(inlink) == 0) { > + ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); > + return 0; > + } > + > + if (!s->eof) > + FF_FILTER_FORWARD_WANTED(outlink, inlink); > + > + return FFERROR_NOT_READY; > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > +} Are AVFilters required to have an uninit function even if empty? And is it really certain that the FFBufQueue is empty when uninit is called? - Andreas
diff --git a/libavfilter/Makefile b/libavfilter/Makefile index f982afe15f..421a01753e 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -134,6 +134,7 @@ OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o +OBJS-$(CONFIG_SPEECHNORM_FILTER) += af_speechnorm.o OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o diff --git a/libavfilter/af_speechnorm.c b/libavfilter/af_speechnorm.c new file mode 100644 index 0000000000..52fc8e6e42 --- /dev/null +++ b/libavfilter/af_speechnorm.c @@ -0,0 +1,381 @@ +/* + * Speech Normalizer + * Copyright (c) 2020 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 + */ + +/** + * @file + * Speech Normalizer + */ + +#include <float.h> + +#include "libavutil/avassert.h" +#include "libavutil/opt.h" + +#define FF_BUFQUEUE_SIZE (1024) +#include "bufferqueue.h" + +#include "audio.h" +#include "avfilter.h" +#include "filters.h" +#include "internal.h" + +#define MAX_ITEMS 882000 + +typedef struct PeriodItem { + int size; + int type; + double max_peak; +} PeriodItem; + +typedef struct SpeechNormalizerContext { + const AVClass *class; + + double peak_value; + double max_amplification; + double threshold_value; + double feedback; + double decay; + int channels; + + int max_period; + int eof; + int64_t pts; + int state[12]; + + PeriodItem pi[12][MAX_ITEMS]; + double gain_state[12]; + int pi_start[12]; + int pi_end[12]; + + struct FFBufQueue queue; +} SpeechNormalizerContext; + +#define OFFSET(x) offsetof(SpeechNormalizerContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM + +static const AVOption speechnorm_options[] = { + { "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, + { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, + { "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 2.0}, 1.0, 10.0, FLAGS }, + { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 2.0}, 1.0, 10.0, FLAGS }, + { "threshold", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0}, 0.0, 1.0, FLAGS }, + { "t", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0}, 0.0, 1.0, FLAGS }, + { "feedback", "set the feedback value", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl = 0.001}, 0.0, 1.0, FLAGS }, + { "f", "set the feedback value", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl = 0.001}, 0.0, 1.0, FLAGS }, + { "decay", "set the decay value", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl = 0.999}, 0.0, 1.0, FLAGS }, + { "d", "set the decay value", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl = 0.999}, 0.0, 1.0, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(speechnorm); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return 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; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + SpeechNormalizerContext *s = ctx->priv; + + s->max_period = inlink->sample_rate / (2 * 20); + s->channels = inlink->channels; + for (int ch = 0; ch < s->channels; ch++) + s->state[ch] = -1; + + return 0; +} + +static int get_pi_samples(PeriodItem *pi, int start, int end, int mode) +{ + int sum; + + if (mode && pi[start].type == 0) + return 0; + + sum = pi[start].size; + av_assert0(sum > 0); + while (start != end) { + start++; + if (start >= MAX_ITEMS) + start = 0; + if (mode && pi[start].type == 0) + break; + av_assert0(pi[start].size > 0); + sum += pi[start].size; + if (pi[start].type == 0) + break; + } + + return sum; +} + +static int consume_pi(PeriodItem *pi, int start, int end, int nb_samples) +{ + int sum; + + sum = pi[start].size; + av_assert0(pi[start].size > 0); + while (sum < nb_samples) { + av_assert0(pi[start].type == 1); + av_assert0(start != end); + start++; + if (start >= MAX_ITEMS) + start = 0; + av_assert0(pi[start].size > 0); + sum += pi[start].size; + } + + av_assert0(pi[start].size >= sum - nb_samples); + pi[start].size = sum - nb_samples; + av_assert0(pi[start].size >= 0); + if (pi[start].size == 0 && start != end) { + start++; + if (start >= MAX_ITEMS) + start = 0; + } + + return start; +} + +static int get_queued_samples(SpeechNormalizerContext *s) +{ + int sum = 0; + + for (int i = 0; i < s->queue.available; i++) { + AVFrame *frame = ff_bufqueue_peek(&s->queue, i); + sum += frame->nb_samples; + } + + return sum; +} + +static int filter_frame(AVFilterContext *ctx) +{ + SpeechNormalizerContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + AVFilterLink *inlink = ctx->inputs[0]; + int min_pi_nb_samples; + AVFrame *in = NULL; + int ret; + + for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) { + ret = ff_inlink_consume_frame(inlink, &in); + if (ret < 0) + return ret; + if (ret == 0) + break; + + ff_bufqueue_add(ctx, &s->queue, in); + + for (int ch = 0; ch < inlink->channels; ch++) { + const double *src = (const double *)in->extended_data[ch]; + int n = 0; + + if (s->state[ch] < 0) + s->state[ch] = src[0] >= 0.; + + while (n < in->nb_samples) { + if (s->state[ch] != (src[n] >= 0.) || s->pi[ch][s->pi_end[ch]].size > s->max_period) { + s->state[ch] = src[n] >= 0.; + av_assert0(s->pi[ch][s->pi_end[ch]].size > 0); + s->pi[ch][s->pi_end[ch]].type = 1; + s->pi_end[ch]++; + if (s->pi_end[ch] >= MAX_ITEMS) + s->pi_end[ch] = 0; + s->pi[ch][s->pi_end[ch]].max_peak = DBL_MIN; + s->pi[ch][s->pi_end[ch]].type = 0; + s->pi[ch][s->pi_end[ch]].size = 0; + av_assert0(s->pi_end[ch] != s->pi_start[ch]); + } + + if (src[n] >= 0.) { + while (src[n] >= 0.) { + s->pi[ch][s->pi_end[ch]].max_peak = FFMAX(s->pi[ch][s->pi_end[ch]].max_peak, FFABS(src[n])); + s->pi[ch][s->pi_end[ch]].size++; + n++; + if (n >= in->nb_samples) + break; + } + } else { + while (src[n] < 0.) { + s->pi[ch][s->pi_end[ch]].max_peak = FFMAX(s->pi[ch][s->pi_end[ch]].max_peak, FFABS(src[n])); + s->pi[ch][s->pi_end[ch]].size++; + n++; + if (n >= in->nb_samples) + break; + } + } + } + } + } + + if (s->queue.available > 0) { + in = ff_bufqueue_peek(&s->queue, 0); + if (!in) + return 1; + } else { + return 1; + } + + min_pi_nb_samples = get_pi_samples(s->pi[0], s->pi_start[0], s->pi_end[0], 1); + for (int ch = 1; ch < inlink->channels; ch++) { + min_pi_nb_samples = FFMIN(min_pi_nb_samples, get_pi_samples(s->pi[ch], s->pi_start[ch], s->pi_end[ch], 1)); + } + + if (min_pi_nb_samples >= in->nb_samples) { + int nb_samples = get_queued_samples(s); + + in = ff_bufqueue_get(&s->queue); + + av_frame_make_writable(in); + + nb_samples -= in->nb_samples; + + for (int ch = 0; ch < inlink->channels; ch++) { + double *src = (double *)in->extended_data[ch]; + int start = s->pi_start[ch]; + int offset = 0; + double gain; + + for (int n = 0; n < in->nb_samples; n++) { + if (n >= offset) { + if (s->pi[ch][start].max_peak > s->threshold_value) + gain = FFMIN(s->max_amplification, s->peak_value / s->pi[ch][start].max_peak); + else + gain = 1.; + av_assert0(s->pi[ch][start].size > 0); + offset += s->pi[ch][start++].size; + if (start >= MAX_ITEMS) + start = 0; + } + s->gain_state[ch] = FFMIN(gain, gain * s->feedback + s->gain_state[ch] * s->decay); + src[n] *= s->gain_state[ch]; + } + } + + for (int ch = 0; ch < inlink->channels; ch++) { + s->pi_start[ch] = consume_pi(s->pi[ch], s->pi_start[ch], s->pi_end[ch], in->nb_samples); + } + + for (int ch = 0; ch < inlink->channels; ch++) { + int pi_nb_samples = get_pi_samples(s->pi[ch], s->pi_start[ch], s->pi_end[ch], 0); + + if (nb_samples != pi_nb_samples) { + av_assert0(0); + } + } + + return ff_filter_frame(outlink, in); + } + + return 1; +} + +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + SpeechNormalizerContext *s = ctx->priv; + int ret = 0, status; + int64_t pts; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = filter_frame(ctx); + if (ret <= 0) + return ret; + + if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { + if (status == AVERROR_EOF) + s->eof = 1; + } + + if (s->eof && ff_inlink_queued_samples(inlink) == 0) { + ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); + return 0; + } + + if (!s->eof) + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ +} + +static const AVFilterPad avfilter_af_speechnorm_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad avfilter_af_speechnorm_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_speechnorm = { + .name = "speechnorm", + .description = NULL_IF_CONFIG_SMALL("Speech Normalizer."), + .query_formats = query_formats, + .priv_size = sizeof(SpeechNormalizerContext), + .priv_class = &speechnorm_class, + .activate = activate, + .uninit = uninit, + .inputs = avfilter_af_speechnorm_inputs, + .outputs = avfilter_af_speechnorm_outputs, + .process_command = ff_filter_process_command, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 1b94501da0..fbe9633a99 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -128,6 +128,7 @@ extern AVFilter ff_af_sidechaingate; extern AVFilter ff_af_silencedetect; extern AVFilter ff_af_silenceremove; extern AVFilter ff_af_sofalizer; +extern AVFilter ff_af_speechnorm; extern AVFilter ff_af_stereotools; extern AVFilter ff_af_stereowiden; extern AVFilter ff_af_superequalizer;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- libavfilter/Makefile | 1 + libavfilter/af_speechnorm.c | 381 ++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 3 files changed, 383 insertions(+) create mode 100644 libavfilter/af_speechnorm.c