avfilter/af_dynaudnorm: add support for commands
This commit is contained in:
Paul B Mahol
parent
1e3f4b5f19
commit
27ec72db06
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@ -3448,6 +3448,10 @@ to 0, which means all input frames will be normalized.
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This option is mostly useful if digital noise is not wanted to be amplified.
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@end table
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@subsection Commands
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This filter supports the all above options as @ref{commands}.
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@section earwax
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Make audio easier to listen to on headphones.
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@ -29,7 +29,10 @@
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#define FF_BUFQUEUE_SIZE 302
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#define MIN_FILTER_SIZE 3
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#define MAX_FILTER_SIZE 301
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#define FF_BUFQUEUE_SIZE (MAX_FILTER_SIZE + 1)
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#include "libavfilter/bufferqueue.h"
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#include "audio.h"
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@ -45,8 +48,8 @@ typedef struct local_gain {
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typedef struct cqueue {
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double *elements;
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int size;
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int max_size;
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int nb_elements;
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int first;
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} cqueue;
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typedef struct DynamicAudioNormalizerContext {
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@ -69,7 +72,6 @@ typedef struct DynamicAudioNormalizerContext {
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double *prev_amplification_factor;
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double *dc_correction_value;
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double *compress_threshold;
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double *fade_factors[2];
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double *weights;
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int channels;
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@ -85,7 +87,7 @@ typedef struct DynamicAudioNormalizerContext {
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} DynamicAudioNormalizerContext;
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#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption dynaudnorm_options[] = {
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{ "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
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@ -161,30 +163,22 @@ static inline int frame_size(int sample_rate, int frame_len_msec)
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return frame_size + (frame_size % 2);
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}
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static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
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{
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const double step_size = 1.0 / frame_len;
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int pos;
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for (pos = 0; pos < frame_len; pos++) {
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fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
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fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
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}
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}
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static cqueue *cqueue_create(int size)
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static cqueue *cqueue_create(int size, int max_size)
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{
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cqueue *q;
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if (max_size < size)
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return NULL;
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q = av_malloc(sizeof(cqueue));
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if (!q)
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return NULL;
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q->max_size = max_size;
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q->size = size;
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q->nb_elements = 0;
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q->first = 0;
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q->elements = av_malloc_array(size, sizeof(double));
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q->elements = av_malloc_array(max_size, sizeof(double));
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if (!q->elements) {
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av_free(q);
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return NULL;
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@ -207,17 +201,14 @@ static int cqueue_size(cqueue *q)
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static int cqueue_empty(cqueue *q)
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{
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return !q->nb_elements;
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return q->nb_elements <= 0;
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}
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static int cqueue_enqueue(cqueue *q, double element)
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{
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int i;
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av_assert2(q->nb_elements < q->max_size);
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av_assert2(q->nb_elements != q->size);
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i = (q->first + q->nb_elements) % q->size;
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q->elements[i] = element;
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q->elements[q->nb_elements] = element;
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q->nb_elements++;
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return 0;
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@ -226,15 +217,15 @@ static int cqueue_enqueue(cqueue *q, double element)
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static double cqueue_peek(cqueue *q, int index)
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{
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av_assert2(index < q->nb_elements);
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return q->elements[(q->first + index) % q->size];
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return q->elements[index];
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}
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static int cqueue_dequeue(cqueue *q, double *element)
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{
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av_assert2(!cqueue_empty(q));
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*element = q->elements[q->first];
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q->first = (q->first + 1) % q->size;
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*element = q->elements[0];
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memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
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q->nb_elements--;
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return 0;
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@ -244,12 +235,34 @@ static int cqueue_pop(cqueue *q)
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{
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av_assert2(!cqueue_empty(q));
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q->first = (q->first + 1) % q->size;
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memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
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q->nb_elements--;
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return 0;
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}
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static void cqueue_resize(cqueue *q, int new_size)
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{
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av_assert2(q->max_size >= new_size);
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av_assert2(MIN_FILTER_SIZE <= new_size);
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if (new_size > q->nb_elements) {
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const int side = (new_size - q->nb_elements) / 2;
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memmove(q->elements + side, q->elements, sizeof(double) * q->nb_elements);
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for (int i = 0; i < side; i++)
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q->elements[i] = q->elements[side];
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q->nb_elements = new_size - 1 - side;
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} else {
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int count = (q->size - new_size + 1) / 2;
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while (count-- > 0)
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cqueue_pop(q);
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}
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q->size = new_size;
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}
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static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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{
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double total_weight = 0.0;
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@ -285,8 +298,6 @@ static av_cold void uninit(AVFilterContext *ctx)
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av_freep(&s->prev_amplification_factor);
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av_freep(&s->dc_correction_value);
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av_freep(&s->compress_threshold);
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av_freep(&s->fade_factors[0]);
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av_freep(&s->fade_factors[1]);
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for (c = 0; c < s->channels; c++) {
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if (s->gain_history_original)
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@ -324,9 +335,6 @@ static int config_input(AVFilterLink *inlink)
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s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
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av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
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s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
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s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
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s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
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s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
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s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
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@ -334,10 +342,10 @@ static int config_input(AVFilterLink *inlink)
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s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
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s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
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s->threshold_history = av_calloc(inlink->channels, sizeof(*s->threshold_history));
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s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
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s->is_enabled = cqueue_create(s->filter_size);
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s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights));
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s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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if (!s->prev_amplification_factor || !s->dc_correction_value ||
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!s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
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!s->compress_threshold ||
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!s->gain_history_original || !s->gain_history_minimum ||
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!s->gain_history_smoothed || !s->threshold_history ||
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!s->is_enabled || !s->weights)
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@ -346,26 +354,27 @@ static int config_input(AVFilterLink *inlink)
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for (c = 0; c < inlink->channels; c++) {
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s->prev_amplification_factor[c] = 1.0;
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s->gain_history_original[c] = cqueue_create(s->filter_size);
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s->gain_history_minimum[c] = cqueue_create(s->filter_size);
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s->gain_history_smoothed[c] = cqueue_create(s->filter_size);
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s->threshold_history[c] = cqueue_create(s->filter_size);
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s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->gain_history_minimum[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->gain_history_smoothed[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->threshold_history[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
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!s->gain_history_smoothed[c] || !s->threshold_history[c])
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return AVERROR(ENOMEM);
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}
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precalculate_fade_factors(s->fade_factors, s->frame_len);
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init_gaussian_filter(s);
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return 0;
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}
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static inline double fade(double prev, double next, int pos,
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double *fade_factors[2])
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static inline double fade(double prev, double next, int pos, int length)
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{
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return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
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const double step_size = 1.0 / length;
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const double f0 = 1.0 - (step_size * (pos + 1.0));
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const double f1 = 1.0 - f0;
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return f0 * prev + f1 * next;
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}
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static inline double pow_2(const double value)
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@ -473,8 +482,7 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu
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static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
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local_gain gain)
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{
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if (cqueue_empty(s->gain_history_original[channel]) ||
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cqueue_empty(s->gain_history_minimum[channel])) {
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if (cqueue_empty(s->gain_history_original[channel])) {
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const int pre_fill_size = s->filter_size / 2;
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const double initial_value = s->alt_boundary_mode ? gain.max_gain : s->peak_value;
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@ -487,11 +495,9 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
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}
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cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
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cqueue_enqueue(s->threshold_history[channel], gain.threshold);
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while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
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double minimum;
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av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
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if (cqueue_empty(s->gain_history_minimum[channel])) {
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const int pre_fill_size = s->filter_size / 2;
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@ -509,12 +515,14 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
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cqueue_enqueue(s->gain_history_minimum[channel], minimum);
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cqueue_enqueue(s->threshold_history[channel], gain.threshold);
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cqueue_pop(s->gain_history_original[channel]);
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}
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while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
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double smoothed;
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av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
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smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
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smoothed = FFMIN(smoothed, cqueue_peek(s->gain_history_minimum[channel], s->filter_size / 2));
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@ -549,7 +557,7 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra
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s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
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for (i = 0; i < frame->nb_samples; i++) {
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dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
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dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
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}
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}
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}
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@ -622,7 +630,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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for (c = 0; c < s->channels; c++) {
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double *const dst_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++) {
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
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dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
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}
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}
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@ -641,7 +649,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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dst_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++) {
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
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dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
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}
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}
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@ -685,12 +693,9 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame, int
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for (i = 0; i < frame->nb_samples && enabled; i++) {
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const double amplification_factor = fade(s->prev_amplification_factor[c],
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current_amplification_factor, i,
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s->fade_factors);
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frame->nb_samples);
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dst_ptr[i] *= amplification_factor;
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if (fabs(dst_ptr[i]) > s->peak_value)
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dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
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}
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s->prev_amplification_factor[c] = current_amplification_factor;
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@ -704,9 +709,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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AVFilterLink *outlink = ctx->outputs[0];
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int ret = 1;
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if (!cqueue_empty(s->gain_history_smoothed[0])) {
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double is_enabled;
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while (((s->queue.available >= s->filter_size) ||
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(s->eof && s->queue.available)) &&
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!cqueue_empty(s->gain_history_smoothed[0])) {
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AVFrame *out = ff_bufqueue_get(&s->queue);
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double is_enabled;
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cqueue_dequeue(s->is_enabled, &is_enabled);
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@ -715,13 +722,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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}
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av_frame_make_writable(in);
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if (!s->eof)
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cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
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analyze_frame(s, in);
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if (!s->eof)
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if (!s->eof) {
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ff_bufqueue_add(ctx, &s->queue, in);
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else
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cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
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} else {
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av_frame_free(&in);
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}
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return ret;
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}
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@ -814,6 +821,34 @@ static int activate(AVFilterContext *ctx)
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return FFERROR_NOT_READY;
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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char *res, int res_len, int flags)
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{
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DynamicAudioNormalizerContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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int prev_filter_size = s->filter_size;
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int ret;
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ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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if (ret < 0)
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return ret;
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s->filter_size |= 1;
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if (prev_filter_size != s->filter_size) {
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init_gaussian_filter(s);
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for (int c = 0; c < s->channels; c++) {
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cqueue_resize(s->gain_history_original[c], s->filter_size);
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cqueue_resize(s->gain_history_minimum[c], s->filter_size);
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cqueue_resize(s->threshold_history[c], s->filter_size);
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}
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}
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s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
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return 0;
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}
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static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
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{
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.name = "default",
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@ -843,4 +878,5 @@ AVFilter ff_af_dynaudnorm = {
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.outputs = avfilter_af_dynaudnorm_outputs,
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.priv_class = &dynaudnorm_class,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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.process_command = process_command,
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};
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