[FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
Paul B Mahol
onemda at gmail.com
Thu Jul 9 18:55:48 CEST 2015
From: LoRd_MuldeR <mulder2 at gmx.de>
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 158 ++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_dynaudnorm.c | 739 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 899 insertions(+)
create mode 100644 libavfilter/af_dynaudnorm.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 3fce874..74c408a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1520,6 +1520,164 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
+ at section dynaudnorm
+Dynamic Audio Normalizer.
+
+This filter applies a certain amount of gain to the input audion in order
+to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
+contrast to more "simple" normalization algorithms, the Dynamic Audio
+Normalizer *dynamically* re-adjusts the gain factor to the input audio.
+This allows for applying extra gain to the "quiet" sections of the audio
+while avoiding distortions or clipping the "loud" sections. In other words:
+The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
+sections, in the sense that the volume of each section is brought to the
+same target level. Note, however, that the Dynamic Audio Normalizer achieves
+this goal *without* applying "dynamic range compressing". It will retain 100%
+of the dynamic range *within* each section of the audio file.
+
+ at table @option
+ at item f
+Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
+Default is 500 milliseconds.
+The Dynamic Audio Normalizer processes the input audio in small chunks,
+referred to as frames. This is required, because a peak magnitude has no
+meaning for just a single sample value. Instead, we need to determine the
+peak magnitude for a contiguous sequence of sample values. While a "standard"
+normalizer would simply use the peak magnitude of the complete file, the
+Dynamic Audio Normalizer determines the peak magnitude individually for each
+frame. The length of a frame is specified in milliseconds. By default, the
+Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
+been found to give good results with most files.
+Note that the exact frame length, in number of samples, will be determined
+automatically, based on the sampling rate of the individual input audio file.
+
+ at item g
+Set the Gaussian filter window size. In range from 3 to 301, must be odd
+number. Default is 31.
+Probably the most important parameter of the Dynamic Audio Normalizer is the
+ at code{window size} of the Gaussian smoothing filter. The filter's window size
+is specified in frames, centered around the current frame. For the sake of
+simplicity, this must be an odd number. Consequently, the default value of 31
+takes into account the current frame, as well as the 15 preceding frames and
+the 15 subsequent frames. Using a larger window results in a stronger
+smoothing effect and thus in less gain variation, i.e. slower gain
+adaptation. Conversely, using a smaller window results in a weaker smoothing
+effect and thus in more gain variation, i.e. faster gain adaptation.
+In other words, the more you increase this value, the more the Dynamic Audio
+Normalizer will behave like a "traditional" normalization filter. On the
+contrary, the more you decrease this value, the more the Dynamic Audio
+Normalizer will behave like a dynamic range compressor.
+
+ at item p
+Set the target peak value. This specifies the highest permissible magnitude
+level for the normalized audio input. This filter will try to approach the
+target peak magnitude as closely as possible, but at the same time it also
+makes sure that the normalized signal will never exceed the peak magnitude.
+A frame's maximum local gain factor is imposed directly by the target peak
+magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
+It is not recommended to go above this value.
+
+ at item m
+Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
+The Dynamic Audio Normalizer determines the maximum possible (local) gain
+factor for each input frame, i.e. the maximum gain factor that does not
+result in clipping or distortion. The maximum gain factor is determined by
+the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
+additionally bounds the frame's maximum gain factor by a predetermined
+(global) maximum gain factor. This is done in order to avoid excessive gain
+factors in "silent" or almost silent frames. By default, the maximum gain
+factor is 10.0, For most inputs the default value should be sufficient and
+it usually is not recommended to increase this value. Though, for input
+with an extremely low overall volume level, it may be necessary to allow even
+higher gain factors. Note, however, that the Dynamic Audio Normalizer does
+not simply apply a "hard" threshold (i.e. cut off values above the threshold).
+Instead, a "sigmoid" threshold function will be applied. This way, the
+gainfactors will smoothly approach the threshold value, but never exceed that
+value.
+
+ at item r
+Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
+By default, the Dynamic Audio Normalizer performs "peak" normalization.
+This means that the maximum local gain factor for each frame is defined
+(only) by the frame's highest magnitude sample. This way, the samples can
+be amplified as much as possible without exceeding the maximum signal
+level, i.e. without clipping. Optionally, however, the Dynamic Audio
+Normalizer can also take into account the frame's root mean square,
+abbreviated RMS. In electrical engineering, the RMS is commonly used to
+determine the power of a time-varying signal. It is therefore considered
+that the RMS is a better approximation of the "perceived loudness" than
+just looking at the signal's peak magnitude. Consequently, by adjusting all
+frames to a constant RMS value, a uniform "perceived loudness" can be
+established. If a target RMS value has been specified, a frame's local gain
+factor is defined as the factor that would result in exactly that RMS value.
+Note, however, that the maximum local gain factor is still restricted by the
+frame's highest magnitude sample, in order to prevent clipping.
+
+ at item n
+Enable channels coupling. By default is enabled.
+By default, the Dynamic Audio Normalizer will amplify all channels by the same
+amount. This means the same gain factor will be applied to all channels, i.e.
+the maximum possible gain factor is determined by the "loudest" channel.
+However, in some recordings, it may happen that the volume of the different
+channels is uneven, e.g. one channel may be "quieter" than the other one(s).
+In this case, this option can be used to disable the channel coupling. This way,
+the gain factor will be determined independently for each channel, depending
+only on the individual channel's highest magnitude sample. This allows for
+harmonizing the volume of the different channels.
+
+ at item c
+Enable DC bias correction. By default is disabled.
+An audio signal (in the time domain) is a sequence of sample values.
+In the Dynamic Audio Normalizer these sample values are represented in the
+-1.0 to 1.0 range, regardless of the original input format. Normally, the
+audio signal, or "waveform", should be centered around the zero point.
+That means if we calculate the mean value of all samples in a file, or in a
+single frame, then the result should be 0.0 or at least very close to that
+value. If, however, there is a significant deviation of the mean value from
+0.0, in either positive or negative direction, this is referred to as a
+DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
+Audio Normalizer provides optional DC bias correction.
+With DC bias correction enabled, the Dynamic Audio Normalizer will determine
+the mean value, or "DC correction" offset, of each input frame and subtract
+that value from all of the frame's sample values which ensures those samples
+are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
+boundaries, the DC correction offset values will be interpolated smoothly
+between neighbouring frames.
+
+ at item b
+Enable alternative boundary mode. By default is disabled.
+The Dynamic Audio Normalizer takes into account a certain neighbourhood
+around each frame. This includes the preceding frames as well as the
+subsequent frames. However, for the "boundary" frames, located at the very
+beginning and at the very end of the audio file, not all neighbouring
+frames are available. In particular, for the first few frames in the audio
+file, the preceding frames are not known. And, similarly, for the last few
+frames in the audio file, the subsequent frames are not known. Thus, the
+question arises which gain factors should be assumed for the missing frames
+in the "boundary" region. The Dynamic Audio Normalizer implements two modes
+to deal with this situation. The default boundary mode assumes a gain factor
+of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
+"fade out" at the beginning and at the end of the input, respectively.
+
+ at item s
+Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
+By default, the Dynamic Audio Normalizer does not apply "traditional"
+compression. This means that signal peaks will not be pruned and thus the
+full dynamic range will be retained within each local neighbourhood. However,
+in some cases it may be desirable to combine the Dynamic Audio Normalizer's
+normalization algorithm with a more "traditional" compression.
+For this purpose, the Dynamic Audio Normalizer provides an optional compression
+(thresholding) function. If (and only if) the compression feature is enabled,
+all input frames will be processed by a soft knee thresholding function prior
+to the actual normalization process. Put simply, the thresholding function is
+going to prune all samples whose magnitude exceeds a certain threshold value.
+However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
+value. Instead, the threshold value will be adjusted for each individual
+frame.
+In general, smaller parameters result in stronger compression, and vice versa.
+Values below 3.0 are not recommended, because audible distortion may appear.
+ at end table
+
@section earwax
Make audio easier to listen to on headphones.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index a623433..0b1fc7f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -67,6 +67,7 @@ OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_CHORUS_FILTER) += af_chorus.o generate_wave_table.o
OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
+OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
new file mode 100644
index 0000000..2e02d6b
--- /dev/null
+++ b/libavfilter/af_dynaudnorm.c
@@ -0,0 +1,739 @@
+/*
+ * Dynamic Audio Normalizer
+ * Copyright (c) 2015 LoRd_MuldeR <mulder2 at gmx.de>. Some rights reserved.
+ *
+ * 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
+ * Dynamic Audio Normalizer
+ */
+
+#include <float.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+
+#define FF_BUFQUEUE_SIZE 302
+#include "libavfilter/bufferqueue.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct cqueue {
+ double *elements;
+ size_t size;
+ size_t nb_elements;
+ size_t first;
+} cqueue;
+
+typedef struct DynamicAudioNormalizerContext {
+ const AVClass *class;
+
+ struct FFBufQueue queue;
+
+ int frame_len;
+ int frame_len_msec;
+ int filter_size;
+ int dc_correction;
+ int channels_coupled;
+ int alt_boundary_mode;
+
+ double peak_value;
+ double max_amplification;
+ double target_rms;
+ double compress_factor;
+ double *prev_amplification_factor;
+ double *dc_correction_value;
+ double *compress_threshold;
+ double *fade_factors[2];
+ double *weights;
+
+ int channels;
+ int delay;
+
+ cqueue **gain_history_original;
+ cqueue **gain_history_minimum;
+ cqueue **gain_history_smoothed;
+} DynamicAudioNormalizerContext;
+
+#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption dynaudnorm_options[] = {
+ { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
+ { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
+ { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
+ { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
+ { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
+ { "n", "enable channels coupling", OFFSET(channels_coupled), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, FLAGS },
+ { "c", "enable DC correction", OFFSET(dc_correction), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, FLAGS },
+ { "b", "enable alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, FLAGS },
+ { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(dynaudnorm);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ DynamicAudioNormalizerContext *s = ctx->priv;
+
+ if (!(s->filter_size & 1)) {
+ av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+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_layouts();
+ 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 inline int frame_size(int sample_rate, int frame_len_msec)
+{
+ const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
+ return frame_size + (frame_size % 2);
+}
+
+static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
+{
+ const double step_size = 1.0 / frame_len;
+ int pos;
+
+ for (pos = 0; pos < frame_len; pos++) {
+ fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
+ fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
+ }
+}
+
+static cqueue *cqueue_create(size_t size)
+{
+ cqueue *q;
+
+ q = av_malloc(sizeof(cqueue));
+ if (!q)
+ return NULL;
+
+ q->size = size;
+ q->nb_elements = 0;
+ q->first = 0;
+
+ q->elements = av_malloc(sizeof(double) * size);
+ if (!q->elements) {
+ av_free(q);
+ return NULL;
+ }
+
+ return q;
+}
+
+static void cqueue_free(cqueue *q)
+{
+ av_free(q->elements);
+ av_free(q);
+}
+
+static size_t cqueue_size(cqueue *q)
+{
+ return q->nb_elements;
+}
+
+static int cqueue_empty(cqueue *q)
+{
+ return !q->nb_elements;
+}
+
+static int cqueue_full(cqueue *q)
+{
+ return q->nb_elements == q->size;
+}
+
+static int cqueue_enqueue(cqueue *q, double element)
+{
+ size_t i;
+
+ av_assert0(!cqueue_full(q));
+
+ i = (q->first + q->nb_elements) % q->size;
+ q->elements[i] = element;
+ q->nb_elements++;
+
+ return 0;
+}
+
+static double cqueue_peak(cqueue *q, int index)
+{
+ av_assert0(index < q->nb_elements);
+ return q->elements[(q->first + index) % q->size];
+}
+
+static int cqueue_dequeue(cqueue *q, double *element)
+{
+ av_assert0(!cqueue_empty(q));
+
+ *element = q->elements[q->first];
+ q->first = (q->first + 1) % q->size;
+ q->nb_elements--;
+
+ return 0;
+}
+
+static int cqueue_pop(cqueue *q)
+{
+ av_assert0(!cqueue_empty(q));
+
+ q->first = (q->first + 1) % q->size;
+ q->nb_elements--;
+
+ return 0;
+}
+
+static const double s_pi = 3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679;
+
+static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
+{
+ double total_weight = 0.0;
+ const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
+ double adjust;
+ int i;
+
+ // Pre-computer constants
+ const int offset = s->filter_size / 2;
+ const double c1 = 1.0 / (sigma * sqrt(2.0 * s_pi));
+ const double c2 = 2.0 * pow(sigma, 2.0);
+
+ // Compute weights
+ for (i = 0; i < s->filter_size; i++) {
+ const int x = i - offset;
+
+ s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
+ total_weight += s->weights[i];
+ }
+
+ // Adjust weights
+ adjust = 1.0 / total_weight;
+ for (i = 0; i < s->filter_size; i++) {
+ s->weights[i] *= adjust;
+ }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ DynamicAudioNormalizerContext *s = ctx->priv;
+ int c;
+
+ s->frame_len =
+ inlink->min_samples =
+ inlink->max_samples =
+ inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
+ av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
+
+ s->fade_factors[0] = av_malloc(s->frame_len * sizeof(*s->fade_factors[0]));
+ s->fade_factors[1] = av_malloc(s->frame_len * sizeof(*s->fade_factors[1]));
+
+ s->prev_amplification_factor = av_malloc(inlink->channels * sizeof(*s->prev_amplification_factor));
+ s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
+ s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
+ s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
+ s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
+ s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
+ s->weights = av_malloc(s->filter_size * sizeof(*s->weights));
+ if (!s->prev_amplification_factor || !s->dc_correction_value ||
+ !s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
+ !s->gain_history_original || !s->gain_history_minimum ||
+ !s->gain_history_smoothed || !s->weights)
+ return AVERROR(ENOMEM);
+
+ for (c = 0; c < inlink->channels; c++) {
+ s->prev_amplification_factor[c] = 1.0;
+
+ s->gain_history_original[c] = cqueue_create(s->filter_size);
+ s->gain_history_minimum[c] = cqueue_create(s->filter_size);
+ s->gain_history_smoothed[c] = cqueue_create(s->filter_size);
+
+ if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
+ !s->gain_history_smoothed[c])
+ return AVERROR(ENOMEM);
+ }
+
+ precalculate_fade_factors(s->fade_factors, s->frame_len);
+ init_gaussian_filter(s);
+
+ s->channels = inlink->channels;
+ s->delay = s->filter_size;
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
+ return 0;
+}
+
+static inline double fade(double prev, double next, int pos,
+ double *fade_factors[2])
+{
+ return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
+}
+
+static inline double pow2(const double value)
+{
+ return value * value;
+}
+
+static inline double bound(const double threshold, const double val)
+{
+ const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
+ return erf(CONST * (val / threshold)) * threshold;
+}
+
+static double find_peak_magnitude(AVFrame *frame, int channel)
+{
+ double max = DBL_EPSILON;
+ int c, i;
+
+ if (channel == -1) {
+ for (c = 0; c < frame->channels; c++) {
+ double *data_ptr = (double *)frame->extended_data[c];
+
+ for (i = 0; i < frame->nb_samples; i++)
+ max = FFMAX(max, fabs(data_ptr[i]));
+ }
+ } else {
+ double *data_ptr = (double *)frame->extended_data[channel];
+
+ for (i = 0; i < frame->nb_samples; i++)
+ max = FFMAX(max, fabs(data_ptr[i]));
+ }
+
+ return max;
+}
+
+static double compute_frame_rms(AVFrame *frame, int channel)
+{
+ double rms_value = 0.0;
+ int c, i;
+
+ if (channel == -1) {
+ for (c = 0; c < frame->channels; c++) {
+ const double *data_ptr = (double *)frame->extended_data[c];
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ rms_value += pow2(data_ptr[i]);
+ }
+ }
+
+ rms_value /= frame->nb_samples * frame->channels;
+ } else {
+ const double *data_ptr = (double *)frame->extended_data[channel];
+ for (i = 0; i < frame->nb_samples; i++) {
+ rms_value += pow2(data_ptr[i]);
+ }
+
+ rms_value /= frame->nb_samples;
+ }
+
+ return FFMAX(sqrt(rms_value), DBL_EPSILON);
+}
+
+static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
+ int channel)
+{
+ const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
+ const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
+ return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
+}
+
+static double minimum_filter(cqueue *q)
+{
+ double min = DBL_MAX;
+ int i;
+
+ for (i = 0; i < cqueue_size(q); i++) {
+ min = FFMIN(min, cqueue_peak(q, i));
+ }
+
+ return min;
+}
+
+static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
+{
+ double result = 0.0;
+ int i;
+
+ for (i = 0; i < cqueue_size(q); i++) {
+ result += cqueue_peak(q, i) * s->weights[i];
+ }
+
+ return result;
+}
+
+static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
+ double current_gain_factor)
+{
+ if (cqueue_empty(s->gain_history_original[channel]) ||
+ cqueue_empty(s->gain_history_minimum[channel])) {
+ const int pre_fill_size = s->filter_size / 2;
+
+ s->prev_amplification_factor[channel] = s->alt_boundary_mode ? current_gain_factor : 1.0;
+
+ while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
+ cqueue_enqueue(s->gain_history_original[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
+ }
+
+ while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
+ cqueue_enqueue(s->gain_history_minimum[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
+ }
+ }
+
+ cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
+
+ while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
+ av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
+ const double minimum = minimum_filter(s->gain_history_original[channel]);
+
+ cqueue_enqueue(s->gain_history_minimum[channel], minimum);
+
+ cqueue_pop(s->gain_history_original[channel]);
+ }
+
+ while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
+ av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
+ const double smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);
+
+ cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
+
+ cqueue_pop(s->gain_history_minimum[channel]);
+ }
+}
+
+static inline double update_value(double new, double old, double aggressiveness)
+{
+ av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
+ return aggressiveness * new + (1.0 - aggressiveness) * old;
+}
+
+static void perform_dcc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
+{
+ const double diff = 1.0 / frame->nb_samples;
+ int is_first_frame = cqueue_empty(s->gain_history_original[0]);
+ int c, i;
+
+ for (c = 0; c < s->channels; c++) {
+ double *dst_ptr = (double *)frame->extended_data[c];
+ double current_average_value = 0.0;
+
+ for (i = 0; i < frame->nb_samples; i++)
+ current_average_value += dst_ptr[i] * diff;
+
+ const double prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
+ s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
+ }
+ }
+}
+
+static double setup_compress_thresh(double threshold)
+{
+ if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
+ double current_threshold = threshold;
+ double step_size = 1.0;
+
+ while (step_size > DBL_EPSILON) {
+ while ((current_threshold + step_size > current_threshold) &&
+ (bound(current_threshold + step_size, 1.0) <= threshold)) {
+ current_threshold += step_size;
+ }
+
+ step_size /= 2.0;
+ }
+
+ return current_threshold;
+ } else {
+ return threshold;
+ }
+}
+
+static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
+ AVFrame *frame, int channel)
+{
+ double variance = 0.0;
+ int i, c;
+
+ if (channel == -1) {
+ for (c = 0; c < s->channels; c++) {
+ const double *data_ptr = (double *)frame->extended_data[c];
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ variance += pow2(data_ptr[i]); // Assume that MEAN is *zero*
+ }
+ }
+ variance /= (s->channels * frame->nb_samples) - 1;
+ } else {
+ const double *data_ptr = (double *)frame->extended_data[channel];
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ variance += pow2(data_ptr[i]); // Assume that MEAN is *zero*
+ }
+ variance /= frame->nb_samples - 1;
+ }
+
+ return FFMAX(sqrt(variance), DBL_EPSILON);
+}
+
+static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
+{
+ int is_first_frame = cqueue_empty(s->gain_history_original[0]);
+ int c, i;
+
+ if (s->channels_coupled) {
+ const double standard_deviation = compute_frame_std_dev(s, frame, -1);
+ const double current_threshold = FFMIN(1.0, s->compress_factor * standard_deviation);
+
+ const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
+ s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
+
+ const double prev_actual_thresh = setup_compress_thresh(prev_value);
+ const double curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
+
+ for (c = 0; c < s->channels; c++) {
+ double *const dst_ptr = (double *)frame->extended_data[c];
+ for (i = 0; i < frame->nb_samples; i++) {
+ const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
+ dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
+ }
+ }
+ } else {
+ for (c = 0; c < s->channels; c++) {
+ const double standard_deviation = compute_frame_std_dev(s, frame, c);
+ const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation));
+
+ const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
+ s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
+
+ const double prev_actual_thresh = setup_compress_thresh(prev_value);
+ const double curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
+
+ double *const dst_ptr = (double *)frame->extended_data[c];
+ for (i = 0; i < frame->nb_samples; i++) {
+ const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
+ dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
+ }
+ }
+ }
+}
+
+static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
+{
+ if (s->dc_correction) {
+ perform_dcc_correction(s, frame);
+ }
+
+ if (s->compress_factor > DBL_EPSILON) {
+ perform_compression(s, frame);
+ }
+
+ if (s->channels_coupled) {
+ const double current_gain_factor = get_max_local_gain(s, frame, -1);
+ int c;
+
+ for (c = 0; c < s->channels; c++)
+ update_gain_history(s, c, current_gain_factor);
+ } else {
+ int c;
+
+ for (c = 0; c < s->channels; c++)
+ update_gain_history(s, c, get_max_local_gain(s, frame, c));
+ }
+}
+
+static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
+{
+ int c, i;
+
+ for (c = 0; c < s->channels; c++) {
+ double *dst_ptr = (double *)frame->extended_data[c];
+ double current_amplification_factor;
+
+ cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor);
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ const double amplification_factor = fade(s->prev_amplification_factor[c],
+ current_amplification_factor, i,
+ s->fade_factors);
+
+ dst_ptr[i] *= amplification_factor;
+
+ if (fabs(dst_ptr[i]) > s->peak_value)
+ dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
+ }
+
+ s->prev_amplification_factor[c] = current_amplification_factor;
+ }
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ DynamicAudioNormalizerContext *s = ctx->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int ret = 0;
+
+ if (!cqueue_empty(s->gain_history_smoothed[0])) {
+ AVFrame *out = ff_bufqueue_get(&s->queue);
+
+ amplify_frame(s, out);
+ ret = ff_filter_frame(outlink, out);
+ }
+
+ analyze_frame(s, in);
+ ff_bufqueue_add(ctx, &s->queue, in);
+
+ return ret;
+}
+
+static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
+ AVFilterLink *outlink)
+{
+ AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
+ int c, i;
+
+ if (!out)
+ return AVERROR(ENOMEM);
+
+ for (c = 0; c < s->channels; c++) {
+ double *dst_ptr = (double *)out->extended_data[c];
+
+ for (i = 0; i < out->nb_samples; i++) {
+ dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value);
+ if (s->dc_correction) {
+ dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
+ dst_ptr[i] += s->dc_correction_value[c];
+ }
+ }
+ }
+
+ s->delay--;
+ return filter_frame(inlink, out);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ DynamicAudioNormalizerContext *s = ctx->priv;
+ int ret = 0;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay)
+ ret = flush_buffer(s, ctx->inputs[0], outlink);
+
+ return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ DynamicAudioNormalizerContext *s = ctx->priv;
+ int c;
+
+ av_freep(&s->prev_amplification_factor);
+ av_freep(&s->dc_correction_value);
+ av_freep(&s->compress_threshold);
+ av_freep(&s->fade_factors[0]);
+ av_freep(&s->fade_factors[1]);
+
+ for (c = 0; c < s->channels; c++) {
+ cqueue_free(s->gain_history_original[c]);
+ cqueue_free(s->gain_history_minimum[c]);
+ cqueue_free(s->gain_history_smoothed[c]);
+ }
+
+ av_freep(&s->gain_history_original);
+ av_freep(&s->gain_history_minimum);
+ av_freep(&s->gain_history_smoothed);
+
+ av_freep(&s->weights);
+
+ ff_bufqueue_discard_all(&s->queue);
+}
+
+static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ .needs_writable = 1,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_dynaudnorm = {
+ .name = "dynaudnorm",
+ .description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(DynamicAudioNormalizerContext),
+ .init = init,
+ .uninit = uninit,
+ .inputs = avfilter_af_dynaudnorm_inputs,
+ .outputs = avfilter_af_dynaudnorm_outputs,
+ .priv_class = &dynaudnorm_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ab0dc1d..0421f04 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -83,6 +83,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(CHORUS, chorus, af);
REGISTER_FILTER(COMPAND, compand, af);
REGISTER_FILTER(DCSHIFT, dcshift, af);
+ REGISTER_FILTER(DYNAUDNORM, dynaudnorm, af);
REGISTER_FILTER(EARWAX, earwax, af);
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);
--
1.7.11.2
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