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ffmpeg.wasm-core
Commit 27ec72db authored by Paul B Mahol's avatar Paul B Mahol
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avfilter/af_dynaudnorm: add support for commands

parent 1e3f4b5f
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Showing with 102 additions and 62 deletions
doc/filters.texi
View file @ 27ec72db
......@@ -3448,6 +3448,10 @@ to 0, which means all input frames will be normalized.
This option is mostly useful if digital noise is not wanted to be amplified.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@section earwax
Make audio easier to listen to on headphones.
......
libavfilter/af_dynaudnorm.c
View file @ 27ec72db
......@@ -29,7 +29,10 @@
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#define FF_BUFQUEUE_SIZE 302
#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"
......@@ -45,8 +48,8 @@ typedef struct local_gain {
typedef struct cqueue {
double *elements;
int size;
int max_size;
int nb_elements;
int first;
} cqueue;
typedef struct DynamicAudioNormalizerContext {
......@@ -69,7 +72,6 @@ typedef struct DynamicAudioNormalizerContext {
double *prev_amplification_factor;
double *dc_correction_value;
double *compress_threshold;
double *fade_factors[2];
double *weights;
int channels;
......@@ -85,7 +87,7 @@ typedef struct DynamicAudioNormalizerContext {
} DynamicAudioNormalizerContext;
#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#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 },
......@@ -161,30 +163,22 @@ static inline int frame_size(int sample_rate, int frame_len_msec)
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)
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->first = 0;
q->elements = av_malloc_array(size, sizeof(double));
q->elements = av_malloc_array(max_size, sizeof(double));
if (!q->elements) {
av_free(q);
return NULL;
......@@ -207,17 +201,14 @@ static int cqueue_size(cqueue *q)
static int cqueue_empty(cqueue *q)
{
return !q->nb_elements;
return q->nb_elements <= 0;
}
static int cqueue_enqueue(cqueue *q, double element)
{
int i;
av_assert2(q->nb_elements != q->size);
av_assert2(q->nb_elements < q->max_size);
i = (q->first + q->nb_elements) % q->size;
q->elements[i] = element;
q->elements[q->nb_elements] = element;
q->nb_elements++;
return 0;
......@@ -226,15 +217,15 @@ static int cqueue_enqueue(cqueue *q, double element)
static double cqueue_peek(cqueue *q, int index)
{
av_assert2(index < q->nb_elements);
return q->elements[(q->first + index) % q->size];
return q->elements[index];
}
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;
*element = q->elements[0];
memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
q->nb_elements--;
return 0;
......@@ -244,12 +235,34 @@ static int cqueue_pop(cqueue *q)
{
av_assert2(!cqueue_empty(q));
q->first = (q->first + 1) % q->size;
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;
......@@ -285,8 +298,6 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->prev_amplification_factor);
av_freep(&s->dc_correction_value);
av_freep(&s->compress_threshold);
av_freep(&s->fade_factors[0]);
av_freep(&s->fade_factors[1]);
for (c = 0; c < s->channels; c++) {
if (s->gain_history_original)
......@@ -324,9 +335,6 @@ static int config_input(AVFilterLink *inlink)
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->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
......@@ -334,10 +342,10 @@ static int config_input(AVFilterLink *inlink)
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(s->filter_size, sizeof(*s->weights));
s->is_enabled = cqueue_create(s->filter_size);
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->fade_factors[0] || !s->fade_factors[1] ||
!s->compress_threshold ||
!s->gain_history_original || !s->gain_history_minimum ||
!s->gain_history_smoothed || !s->threshold_history ||
!s->is_enabled || !s->weights)
......@@ -346,26 +354,27 @@ static int config_input(AVFilterLink *inlink)
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);
s->threshold_history[c] = cqueue_create(s->filter_size);
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);
}
precalculate_fade_factors(s->fade_factors, s->frame_len);
init_gaussian_filter(s);
return 0;
}
static inline double fade(double prev, double next, int pos,
double *fade_factors[2])
static inline double fade(double prev, double next, int pos, int length)
{
return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
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)
......@@ -473,8 +482,7 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu
static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
local_gain gain)
{
if (cqueue_empty(s->gain_history_original[channel]) ||
cqueue_empty(s->gain_history_minimum[channel])) {
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;
......@@ -487,11 +495,9 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
}
cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
cqueue_enqueue(s->threshold_history[channel], gain.threshold);
while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
double minimum;
av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
if (cqueue_empty(s->gain_history_minimum[channel])) {
const int pre_fill_size = s->filter_size / 2;
......@@ -509,12 +515,14 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int 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;
av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
smoothed = FFMIN(smoothed, cqueue_peek(s->gain_history_minimum[channel], s->filter_size / 2));
......@@ -549,7 +557,7 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra
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);
dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
}
}
}
......@@ -622,7 +630,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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);
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]);
}
}
......@@ -641,7 +649,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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);
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]);
}
}
......@@ -685,12 +693,9 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame, int
for (i = 0; i < frame->nb_samples && enabled; i++) {
const double amplification_factor = fade(s->prev_amplification_factor[c],
current_amplification_factor, i,
s->fade_factors);
frame->nb_samples);
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;
......@@ -704,9 +709,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterLink *outlink = ctx->outputs[0];
int ret = 1;
if (!cqueue_empty(s->gain_history_smoothed[0])) {
double is_enabled;
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);
......@@ -715,13 +722,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
}
av_frame_make_writable(in);
if (!s->eof)
cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
analyze_frame(s, in);
if (!s->eof)
if (!s->eof) {
ff_bufqueue_add(ctx, &s->queue, in);
else
cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
} else {
av_frame_free(&in);
}
return ret;
}
......@@ -814,6 +821,34 @@ static int activate(AVFilterContext *ctx)
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",
......@@ -843,4 +878,5 @@ AVFilter ff_af_dynaudnorm = {
.outputs = avfilter_af_dynaudnorm_outputs,
.priv_class = &dynaudnorm_class,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.process_command = process_command,
};
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