diff --git a/doc/filters.texi b/doc/filters.texi index 49fab59057..518aef8f22 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1544,6 +1544,164 @@ 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 +@section dynaudnorm +Dynamic Audio Normalizer. + +This filter applies a certain amount of gain to the input audio 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. + +@table @option +@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. + +@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 +@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. + +@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. + +@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 +gain factors will smoothly approach the threshold value, but never exceed that +value. + +@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. + +@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. + +@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. + +@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. + +@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. +@end table + @section earwax Make audio easier to listen to on headphones. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 058b9e9520..a259851548 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 0000000000..fb83c201ce --- /dev/null +++ b/libavfilter/af_dynaudnorm.c @@ -0,0 +1,734 @@ +/* + * Dynamic Audio Normalizer + * Copyright (c) 2015 LoRd_MuldeR . 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 + +#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", "enable channel 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(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(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 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 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-compute 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_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; + + 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_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; + + 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_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) + 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 b0d841064b..01c9e387d5 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);