/* * 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/channel_layout.h" #include "libavutil/eval.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #define MIN_FILTER_SIZE 3 #define MAX_FILTER_SIZE 301 #define FF_BUFQUEUE_SIZE (MAX_FILTER_SIZE + 1) #include "libavfilter/bufferqueue.h" #include "audio.h" #include "avfilter.h" #include "filters.h" #include "internal.h" static const char * const var_names[] = { "ch", ///< the value of the current channel "sn", ///< number of samples "nb_channels", "t", ///< timestamp expressed in seconds "sr", ///< sample rate "p", ///< peak value NULL }; enum var_name { VAR_CH, VAR_SN, VAR_NB_CHANNELS, VAR_T, VAR_SR, VAR_P, VAR_VARS_NB }; typedef struct local_gain { double max_gain; double threshold; } local_gain; typedef struct cqueue { double *elements; int size; int max_size; int nb_elements; } 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 overlap; char *expr_str; double peak_value; double max_amplification; double target_rms; double compress_factor; double threshold; double *prev_amplification_factor; double *dc_correction_value; double *compress_threshold; double *weights; int channels; int sample_advance; int eof; char *channels_to_filter; AVChannelLayout ch_layout; int64_t pts; cqueue **gain_history_original; cqueue **gain_history_minimum; cqueue **gain_history_smoothed; cqueue **threshold_history; cqueue *is_enabled; AVFrame *window; AVExpr *expr; double var_values[VAR_VARS_NB]; } DynamicAudioNormalizerContext; typedef struct ThreadData { AVFrame *in, *out; int enabled; } ThreadData; #define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption dynaudnorm_options[] = { { "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS }, { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS }, { "gausssize", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS }, { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS }, { "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 = 10.0}, 1.0, 100.0, FLAGS }, { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS }, { "targetrms", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "coupling", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS }, { "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS }, { "correctdc", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, { "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, { "altboundary", "set alternative boundary mode", OFFSET(alt_boundary_mode), 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 }, { "compress", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, { "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS }, { "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS }, { "overlap", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS }, { "o", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS }, { "curve", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, { "v", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = 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_WARNING, "filter size %d is invalid. Changing to an odd value.\n", s->filter_size); s->filter_size |= 1; } return 0; } 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 cqueue *cqueue_create(int size, int max_size) { cqueue *q; if (max_size < size) return NULL; q = av_malloc(sizeof(cqueue)); if (!q) return NULL; q->max_size = max_size; q->size = size; q->nb_elements = 0; q->elements = av_malloc_array(max_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 <= 0; } static int cqueue_enqueue(cqueue *q, double element) { av_assert2(q->nb_elements < q->max_size); q->elements[q->nb_elements] = element; q->nb_elements++; return 0; } static double cqueue_peek(cqueue *q, int index) { av_assert2(index < q->nb_elements); return q->elements[index]; } static int cqueue_dequeue(cqueue *q, double *element) { av_assert2(!cqueue_empty(q)); *element = q->elements[0]; memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double)); q->nb_elements--; return 0; } static int cqueue_pop(cqueue *q) { av_assert2(!cqueue_empty(q)); memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double)); q->nb_elements--; return 0; } static void cqueue_resize(cqueue *q, int new_size) { av_assert2(q->max_size >= new_size); av_assert2(MIN_FILTER_SIZE <= new_size); if (new_size > q->nb_elements) { const int side = (new_size - q->nb_elements) / 2; memmove(q->elements + side, q->elements, sizeof(double) * q->nb_elements); for (int i = 0; i < side; i++) q->elements[i] = q->elements[side]; q->nb_elements = new_size - 1 - side; } else { int count = (q->size - new_size + 1) / 2; while (count-- > 0) cqueue_pop(q); } q->size = new_size; } 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; // 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 (int 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 (int i = 0; i < s->filter_size; i++) { s->weights[i] *= adjust; } } static av_cold void uninit(AVFilterContext *ctx) { DynamicAudioNormalizerContext *s = ctx->priv; av_freep(&s->prev_amplification_factor); av_freep(&s->dc_correction_value); av_freep(&s->compress_threshold); for (int 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]); if (s->threshold_history) cqueue_free(s->threshold_history[c]); } av_freep(&s->gain_history_original); av_freep(&s->gain_history_minimum); av_freep(&s->gain_history_smoothed); av_freep(&s->threshold_history); cqueue_free(s->is_enabled); s->is_enabled = NULL; av_freep(&s->weights); av_channel_layout_uninit(&s->ch_layout); ff_bufqueue_discard_all(&s->queue); av_frame_free(&s->window); av_expr_free(s->expr); s->expr = NULL; } static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; int ret = 0; uninit(ctx); s->channels = inlink->ch_layout.nb_channels; s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len); s->prev_amplification_factor = av_malloc_array(inlink->ch_layout.nb_channels, sizeof(*s->prev_amplification_factor)); s->dc_correction_value = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dc_correction_value)); s->compress_threshold = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->compress_threshold)); s->gain_history_original = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_original)); s->gain_history_minimum = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_minimum)); s->gain_history_smoothed = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_smoothed)); s->threshold_history = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->threshold_history)); s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights)); s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE); if (!s->prev_amplification_factor || !s->dc_correction_value || !s->compress_threshold || !s->gain_history_original || !s->gain_history_minimum || !s->gain_history_smoothed || !s->threshold_history || !s->is_enabled || !s->weights) return AVERROR(ENOMEM); for (int c = 0; c < inlink->ch_layout.nb_channels; c++) { s->prev_amplification_factor[c] = 1.0; s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); s->gain_history_minimum[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); s->gain_history_smoothed[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); s->threshold_history[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); if (!s->gain_history_original[c] || !s->gain_history_minimum[c] || !s->gain_history_smoothed[c] || !s->threshold_history[c]) return AVERROR(ENOMEM); } init_gaussian_filter(s); s->window = ff_get_audio_buffer(ctx->outputs[0], s->frame_len * 2); if (!s->window) return AVERROR(ENOMEM); s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap))); s->var_values[VAR_SR] = inlink->sample_rate; s->var_values[VAR_NB_CHANNELS] = s->channels; if (s->expr_str) ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL, NULL, NULL, 0, ctx); return ret; } static inline double fade(double prev, double next, int pos, int length) { const double step_size = 1.0 / length; const double f0 = 1.0 - (step_size * (pos + 1.0)); const double f1 = 1.0 - f0; return f0 * prev + f1 * 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; if (channel == -1) { for (int c = 0; c < frame->ch_layout.nb_channels; c++) { double *data_ptr = (double *)frame->extended_data[c]; for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } } else { double *data_ptr = (double *)frame->extended_data[channel]; for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } return max; } static double compute_frame_rms(AVFrame *frame, int channel) { double rms_value = 0.0; if (channel == -1) { for (int c = 0; c < frame->ch_layout.nb_channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } } rms_value /= frame->nb_samples * frame->ch_layout.nb_channels; } else { const double *data_ptr = (double *)frame->extended_data[channel]; for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } rms_value /= frame->nb_samples; } return fmax(sqrt(rms_value), DBL_EPSILON); } static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame, int channel) { const double peak_magnitude = find_peak_magnitude(frame, channel); const double maximum_gain = s->peak_value / peak_magnitude; const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX; double target_gain = DBL_MAX; local_gain gain; if (s->expr_str) { double var_values[VAR_VARS_NB]; memcpy(var_values, s->var_values, sizeof(var_values)); var_values[VAR_CH] = channel; var_values[VAR_P] = peak_magnitude; target_gain = av_expr_eval(s->expr, var_values, s) / peak_magnitude; } gain.threshold = peak_magnitude > s->threshold; gain.max_gain = bound(s->max_amplification, fmin(target_gain, fmin(maximum_gain, rms_gain))); return gain; } static double minimum_filter(cqueue *q) { double min = DBL_MAX; for (int i = 0; i < cqueue_size(q); i++) { min = fmin(min, cqueue_peek(q, i)); } return min; } static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq) { const double *weights = s->weights; double result = 0.0, tsum = 0.0; for (int i = 0; i < cqueue_size(q); i++) { double tq_item = cqueue_peek(tq, i); double q_item = cqueue_peek(q, i); tsum += tq_item * weights[i]; result += tq_item * weights[i] * q_item; } if (tsum == 0.0) result = 1.0; return result; } static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, local_gain gain) { if (cqueue_empty(s->gain_history_original[channel])) { const int pre_fill_size = s->filter_size / 2; const double initial_value = s->alt_boundary_mode ? gain.max_gain : fmin(1.0, gain.max_gain); 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->threshold_history[channel], gain.threshold); } } cqueue_enqueue(s->gain_history_original[channel], gain.max_gain); while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) { double minimum; 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 = fmin(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_enqueue(s->threshold_history[channel], gain.threshold); cqueue_pop(s->gain_history_original[channel]); } while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) { double smoothed, limit; smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]); limit = cqueue_peek(s->gain_history_original[channel], 0); smoothed = fmin(smoothed, limit); cqueue_enqueue(s->gain_history_smoothed[channel], smoothed); cqueue_pop(s->gain_history_minimum[channel]); cqueue_pop(s->threshold_history[channel]); } } static int update_gain_histories(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { DynamicAudioNormalizerContext *s = ctx->priv; AVFrame *analyze_frame = arg; const int channels = s->channels; const int start = (channels * jobnr) / nb_jobs; const int end = (channels * (jobnr+1)) / nb_jobs; for (int c = start; c < end; c++) update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c)); return 0; } 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 inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch) { enum AVChannel channel = av_channel_layout_channel_from_index(&frame->ch_layout, ch); return av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0; } 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]); for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); double *dst_ptr = (double *)frame->extended_data[c]; double current_average_value = 0.0; double prev_value; for (int 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 (int i = 0; i < frame->nb_samples && !bypass; i++) { dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples); } } } 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; if (channel == -1) { for (int c = 0; c < s->channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; for (int 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 (int i = 0; i < frame->nb_samples; i++) { variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* } variance /= frame->nb_samples - 1; } return fmax(sqrt(variance), DBL_EPSILON); } static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame) { int is_first_frame = cqueue_empty(s->gain_history_original[0]); if (s->channels_coupled) { const double standard_deviation = compute_frame_std_dev(s, frame, -1); const double current_threshold = fmin(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 (int c = 0; c < s->channels; c++) { double *const dst_ptr = (double *)frame->extended_data[c]; const int bypass = bypass_channel(s, frame, c); if (bypass) continue; for (int i = 0; i < frame->nb_samples; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } } } else { for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double standard_deviation = compute_frame_std_dev(s, frame, c); const double current_threshold = setup_compress_thresh(fmin(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 (int i = 0; i < frame->nb_samples && !bypass; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } } } } static int analyze_frame(AVFilterContext *ctx, AVFilterLink *outlink, AVFrame **frame) { DynamicAudioNormalizerContext *s = ctx->priv; AVFrame *analyze_frame; if (s->dc_correction || s->compress_factor > DBL_EPSILON) { int ret; if (!av_frame_is_writable(*frame)) { AVFrame *out = ff_get_audio_buffer(outlink, (*frame)->nb_samples); if (!out) { av_frame_free(frame); return AVERROR(ENOMEM); } ret = av_frame_copy_props(out, *frame); if (ret < 0) { av_frame_free(frame); av_frame_free(&out); return ret; } ret = av_frame_copy(out, *frame); if (ret < 0) { av_frame_free(frame); av_frame_free(&out); return ret; } av_frame_free(frame); *frame = out; } } if (s->dc_correction) perform_dc_correction(s, *frame); if (s->compress_factor > DBL_EPSILON) perform_compression(s, *frame); if (s->frame_len != s->sample_advance) { const int offset = s->frame_len - s->sample_advance; for (int c = 0; c < s->channels; c++) { double *src = (double *)s->window->extended_data[c]; memmove(src, &src[s->sample_advance], offset * sizeof(double)); memcpy(&src[offset], (*frame)->extended_data[c], (*frame)->nb_samples * sizeof(double)); memset(&src[offset + (*frame)->nb_samples], 0, (s->sample_advance - (*frame)->nb_samples) * sizeof(double)); } analyze_frame = s->window; } else { av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0, FFMIN(s->frame_len, (*frame)->nb_samples), (*frame)->ch_layout.nb_channels, (*frame)->format); analyze_frame = *frame; } s->var_values[VAR_SN] = outlink->sample_count_in; s->var_values[VAR_T] = s->var_values[VAR_SN] * (double)1/outlink->sample_rate; if (s->channels_coupled) { const local_gain gain = get_max_local_gain(s, analyze_frame, -1); for (int c = 0; c < s->channels; c++) update_gain_history(s, c, gain); } else { ff_filter_execute(ctx, update_gain_histories, analyze_frame, NULL, FFMIN(s->channels, ff_filter_get_nb_threads(ctx))); } return 0; } static void amplify_channel(DynamicAudioNormalizerContext *s, AVFrame *in, AVFrame *frame, int enabled, int c) { const int bypass = bypass_channel(s, frame, c); const double *src_ptr = (const double *)in->extended_data[c]; double *dst_ptr = (double *)frame->extended_data[c]; double current_amplification_factor; cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor); for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) { const double amplification_factor = fade(s->prev_amplification_factor[c], current_amplification_factor, i, frame->nb_samples); dst_ptr[i] = src_ptr[i] * amplification_factor; } s->prev_amplification_factor[c] = current_amplification_factor; } static int amplify_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { DynamicAudioNormalizerContext *s = ctx->priv; ThreadData *td = arg; AVFrame *out = td->out; AVFrame *in = td->in; const int enabled = td->enabled; const int channels = s->channels; const int start = (channels * jobnr) / nb_jobs; const int end = (channels * (jobnr+1)) / nb_jobs; for (int ch = start; ch < end; ch++) amplify_channel(s, in, out, enabled, ch); return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; ThreadData td; int ret; while (((s->queue.available >= s->filter_size) || (s->eof && s->queue.available)) && !cqueue_empty(s->gain_history_smoothed[0])) { AVFrame *in = ff_bufqueue_get(&s->queue); AVFrame *out; double is_enabled; cqueue_dequeue(s->is_enabled, &is_enabled); if (av_frame_is_writable(in)) { out = in; } else { out = ff_get_audio_buffer(outlink, in->nb_samples); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); } td.in = in; td.out = out; td.enabled = is_enabled > 0.; ff_filter_execute(ctx, amplify_channels, &td, NULL, FFMIN(s->channels, ff_filter_get_nb_threads(ctx))); s->pts = out->pts + av_rescale_q(out->nb_samples, av_make_q(1, outlink->sample_rate), outlink->time_base); if (out != in) av_frame_free(&in); ret = ff_filter_frame(outlink, out); if (ret < 0) return ret; } ret = analyze_frame(ctx, outlink, &in); if (ret < 0) return ret; if (!s->eof) { ff_bufqueue_add(ctx, &s->queue, in); cqueue_enqueue(s->is_enabled, !ctx->is_disabled); } else { av_frame_free(&in); } return 1; } static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, AVFilterLink *outlink) { AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance); if (!out) return AVERROR(ENOMEM); for (int c = 0; c < s->channels; c++) { double *dst_ptr = (double *)out->extended_data[c]; for (int i = 0; i < out->nb_samples; i++) { dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(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]; } } } return filter_frame(inlink, out); } static int flush(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AVFilterLink *inlink = ctx->inputs[0]; DynamicAudioNormalizerContext *s = ctx->priv; while (s->eof && cqueue_empty(s->gain_history_smoothed[0])) { for (int c = 0; c < s->channels; c++) update_gain_history(s, c, (local_gain){ cqueue_peek(s->gain_history_original[c], 0), 1.0}); } return flush_buffer(s, inlink, outlink); } static int activate(AVFilterContext *ctx) { AVFilterLink *inlink = ctx->inputs[0]; AVFilterLink *outlink = ctx->outputs[0]; DynamicAudioNormalizerContext *s = ctx->priv; AVFrame *in = NULL; int ret = 0, status; int64_t pts; ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout); if (ret < 0) return ret; if (strcmp(s->channels_to_filter, "all")) av_channel_layout_from_string(&s->ch_layout, s->channels_to_filter); FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); if (!s->eof) { ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in); if (ret < 0) return ret; if (ret > 0) { ret = filter_frame(inlink, in); if (ret <= 0) return ret; } if (ff_inlink_check_available_samples(inlink, s->sample_advance) > 0) { ff_filter_set_ready(ctx, 10); return 0; } } if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { if (status == AVERROR_EOF) s->eof = 1; } if (s->eof && s->queue.available) return flush(outlink); if (s->eof && !s->queue.available) { 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 int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags) { DynamicAudioNormalizerContext *s = ctx->priv; AVFilterLink *inlink = ctx->inputs[0]; int prev_filter_size = s->filter_size; int ret; ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); if (ret < 0) return ret; s->filter_size |= 1; if (prev_filter_size != s->filter_size) { init_gaussian_filter(s); for (int c = 0; c < s->channels; c++) { cqueue_resize(s->gain_history_original[c], s->filter_size); cqueue_resize(s->gain_history_minimum[c], s->filter_size); cqueue_resize(s->threshold_history[c], s->filter_size); } } s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap))); if (s->expr_str) { ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL, NULL, NULL, 0, ctx); if (ret < 0) return ret; } return 0; } static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, }, }; const AVFilter ff_af_dynaudnorm = { .name = "dynaudnorm", .description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."), .priv_size = sizeof(DynamicAudioNormalizerContext), .init = init, .uninit = uninit, .activate = activate, FILTER_INPUTS(avfilter_af_dynaudnorm_inputs), FILTER_OUTPUTS(ff_audio_default_filterpad), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP), .priv_class = &dynaudnorm_class, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | AVFILTER_FLAG_SLICE_THREADS, .process_command = process_command, };