| /* |
| * 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 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" |
| |
| 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 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 eof; |
| int64_t pts; |
| |
| cqueue **gain_history_original; |
| cqueue **gain_history_minimum; |
| cqueue **gain_history_smoothed; |
| cqueue **threshold_history; |
| |
| cqueue *is_enabled; |
| } DynamicAudioNormalizerContext; |
| |
| #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 }, |
| { 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 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 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; |
| 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); |
| |
| 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]); |
| 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); |
| |
| 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->channels = inlink->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->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->threshold_history = av_calloc(inlink->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 (c = 0; c < inlink->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); |
| |
| return 0; |
| } |
| |
| 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; |
| 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 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; |
| local_gain gain; |
| |
| gain.threshold = peak_magnitude > s->threshold; |
| gain.max_gain = bound(s->max_amplification, FFMIN(maximum_gain, rms_gain)); |
| |
| return 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, cqueue *tq) |
| { |
| double result = 0.0, tsum = 0.0; |
| int i; |
| |
| for (i = 0; i < cqueue_size(q); i++) { |
| tsum += cqueue_peek(tq, i) * s->weights[i]; |
| result += cqueue_peek(q, i) * s->weights[i] * cqueue_peek(tq, i); |
| } |
| |
| 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 : s->peak_value; |
| |
| 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 = 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_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 = FFMIN(smoothed, limit); |
| |
| cqueue_enqueue(s->gain_history_smoothed[channel], smoothed); |
| |
| cqueue_pop(s->gain_history_minimum[channel]); |
| cqueue_pop(s->threshold_history[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, 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; |
| 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, frame->nb_samples); |
| 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, frame->nb_samples); |
| 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 local_gain gain = get_max_local_gain(s, frame, -1); |
| int c; |
| |
| for (c = 0; c < s->channels; c++) |
| update_gain_history(s, c, gain); |
| } 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 enabled) |
| { |
| 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 && enabled; i++) { |
| const double amplification_factor = fade(s->prev_amplification_factor[c], |
| current_amplification_factor, i, |
| frame->nb_samples); |
| |
| dst_ptr[i] *= amplification_factor; |
| } |
| |
| 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 = ctx->outputs[0]; |
| int ret = 1; |
| |
| while (((s->queue.available >= s->filter_size) || |
| (s->eof && s->queue.available)) && |
| !cqueue_empty(s->gain_history_smoothed[0])) { |
| AVFrame *out = ff_bufqueue_get(&s->queue); |
| double is_enabled; |
| |
| cqueue_dequeue(s->is_enabled, &is_enabled); |
| |
| amplify_frame(s, out, is_enabled > 0.); |
| s->pts = out->pts + out->nb_samples; |
| ret = ff_filter_frame(outlink, out); |
| } |
| |
| av_frame_make_writable(in); |
| analyze_frame(s, in); |
| if (!s->eof) { |
| ff_bufqueue_add(ctx, &s->queue, in); |
| cqueue_enqueue(s->is_enabled, !ctx->is_disabled); |
| } else { |
| av_frame_free(&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]; |
| } |
| } |
| } |
| |
| return filter_frame(inlink, out); |
| } |
| |
| static int flush(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| DynamicAudioNormalizerContext *s = ctx->priv; |
| int ret = 0; |
| |
| 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); |
| |
| s->pts = out->pts + out->nb_samples; |
| ret = ff_filter_frame(outlink, out); |
| } |
| |
| return ret; |
| } |
| |
| 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; |
| |
| FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); |
| |
| if (!s->eof) { |
| ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &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->frame_len) > 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); |
| |
| return 0; |
| } |
| |
| static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_input, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| }, |
| { 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, |
| .activate = activate, |
| .inputs = avfilter_af_dynaudnorm_inputs, |
| .outputs = avfilter_af_dynaudnorm_outputs, |
| .priv_class = &dynaudnorm_class, |
| .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, |
| .process_command = process_command, |
| }; |
