/*
* Dynamic Audio Normalizer
* Copyright (c) 2015 LoRd_MuldeR <mulder2@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;
int size;
int nb_elements;
int 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", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
{ "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.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_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 inline int frame_size(int sample_rate, int frame_len_msec)
{
const int frame_size = lrint((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(int 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_array(size, sizeof(double));
if (!q->elements) {
av_free(q);
return NULL;
}
return q;
}
static void cqueue_free(cqueue *q)
{
if (q)
av_free(q->elements);
av_free(q);
}
static int cqueue_size(cqueue *q)
{
return q->nb_elements;
}
static int cqueue_empty(cqueue *q)
{
return !q->nb_elements;
}
static int cqueue_enqueue(cqueue *q, double element)
{
int i;
av_assert2(q->nb_elements != q->size);
i = (q->first + q->nb_elements) % q->size;
q->elements[i] = element;
q->nb_elements++;
return 0;
}
static double cqueue_peek(cqueue *q, int index)
{
av_assert2(index < q->nb_elements);
return q->elements[(q->first + index) % q->size];
}
static int cqueue_dequeue(cqueue *q, double *element)
{
av_assert2(!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_assert2(!cqueue_empty(q));
q->first = (q->first + 1) % q->size;
q->nb_elements--;
return 0;
}
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-compute constants
const int offset = s->filter_size / 2;
const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
const double c2 = 2.0 * sigma * sigma;
// Compute weights
for (i = 0; i < s->filter_size; i++) {
const int x = i - offset;
s->weights[i] = c1 * exp(-x * x / 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 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++) {
if (s->gain_history_original)
cqueue_free(s->gain_history_original[c]);
if (s->gain_history_minimum)
cqueue_free(s->gain_history_minimum[c]);
if (s->gain_history_smoothed)
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 int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
DynamicAudioNormalizerContext *s = ctx->priv;
int c;
uninit(ctx);
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_array(s->frame_len, sizeof(*s->fade_factors[0]));
s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
s->prev_amplification_factor = av_malloc_array(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_array(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 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 pow_2(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 += pow_2(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 += pow_2(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_peek(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_peek(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;
const double initial_value = s->alt_boundary_mode ? current_gain_factor : 1.0;
s->prev_amplification_factor[channel] = initial_value;
while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
cqueue_enqueue(s->gain_history_original[channel], initial_value);
}
}
cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
double minimum;
av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
if (cqueue_empty(s->gain_history_minimum[channel])) {
const int pre_fill_size = s->filter_size / 2;
double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
int input = pre_fill_size;
while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
input++;
initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], input));
cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
}
}
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) {
double smoothed;
av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
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_dc_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;
double prev_value;
for (i = 0; i < frame->nb_samples; i++)
current_average_value += dst_ptr[i] * diff;
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 ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) >
llrint(current_threshold * (UINT64_C(1) << 63))) &&
(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 += pow_2(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 += pow_2(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];
double prev_actual_thresh, curr_actual_thresh;
s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
prev_actual_thresh = setup_compress_thresh(prev_value);
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];
double prev_actual_thresh, curr_actual_thresh;
double *dst_ptr;
s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
prev_actual_thresh = setup_compress_thresh(prev_value);
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
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_dc_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) {
if (!cqueue_empty(s->gain_history_smoothed[0])) {
ret = flush_buffer(s, ctx->inputs[0], outlink);
} else if (s->queue.available) {
AVFrame *out = ff_bufqueue_get(&s->queue);
ret = ff_filter_frame(outlink, out);
}
}
return ret;
}
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,
.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,
};