X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavfilter%2Faf_dynaudnorm.c;h=365453d60d9479d6b77e897a05786be09556d268;hb=1b64954e42b4dc685a82d5576527a2166d818773;hp=bc573d3482334d3fc7577696cc0f19dcea957e66;hpb=7bb09e57e0657dfd0b63a81130329775afaf4e7a;p=ffmpeg

diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index bc573d34823..365453d60d9 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/libavfilter/af_dynaudnorm.c
@@ -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"
@@ -37,11 +40,16 @@
 #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;
-    int first;
 } cqueue;
 
 typedef struct DynamicAudioNormalizerContext {
@@ -60,26 +68,26 @@ typedef struct DynamicAudioNormalizerContext {
     double max_amplification;
     double target_rms;
     double compress_factor;
+    double threshold;
     double *prev_amplification_factor;
     double *dc_correction_value;
     double *compress_threshold;
-    double *fade_factors[2];
     double *weights;
 
     int channels;
-    int delay;
     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
+#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 },
@@ -100,6 +108,8 @@ static const AVOption dynaudnorm_options[] = {
     { "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 }
 };
 
@@ -153,30 +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;
@@ -199,17 +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;
@@ -218,15 +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;
@@ -236,12 +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;
@@ -277,8 +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)
@@ -287,11 +306,14 @@ static av_cold void uninit(AVFilterContext *ctx)
             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;
@@ -309,51 +331,50 @@ static int config_input(AVFilterLink *inlink)
 
     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->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));
     s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
     s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
     s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
-    s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
-    s->is_enabled = cqueue_create(s->filter_size);
+    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->fade_factors[0] || !s->fade_factors[1] ||
+        !s->compress_threshold ||
         !s->gain_history_original || !s->gain_history_minimum ||
-        !s->gain_history_smoothed || !s->is_enabled || !s->weights)
+        !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);
-        s->gain_history_minimum[c]  = cqueue_create(s->filter_size);
-        s->gain_history_smoothed[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->gain_history_smoothed[c] || !s->threshold_history[c])
             return AVERROR(ENOMEM);
     }
 
-    precalculate_fade_factors(s->fade_factors, s->frame_len);
     init_gaussian_filter(s);
 
-    s->channels = inlink->channels;
-    s->delay = s->filter_size;
-
     return 0;
 }
 
-static 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)
@@ -416,12 +437,18 @@ static double compute_frame_rms(AVFrame *frame, int channel)
     return FFMAX(sqrt(rms_value), DBL_EPSILON);
 }
 
-static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
-                                 int channel)
+static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
+                                     int channel)
 {
-    const double maximum_gain = s->peak_value / find_peak_magnitude(frame, 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;
-    return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
+    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)
@@ -436,38 +463,41 @@ static double minimum_filter(cqueue *q)
     return min;
 }
 
-static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
+static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq)
 {
-    double result = 0.0;
+    double result = 0.0, tsum = 0.0;
     int i;
 
     for (i = 0; i < cqueue_size(q); i++) {
-        result += cqueue_peek(q, i) * s->weights[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,
-                                double current_gain_factor)
+                                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 ? current_gain_factor : 1.0;
+        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], current_gain_factor);
+    cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
 
     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;
@@ -485,17 +515,22 @@ 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]);
+        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]);
     }
 }
 
@@ -523,7 +558,7 @@ 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);
         }
     }
 }
@@ -596,7 +631,7 @@ 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]);
             }
         }
@@ -615,7 +650,7 @@ 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]);
             }
         }
@@ -633,11 +668,11 @@ static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
     }
 
     if (s->channels_coupled) {
-        const double current_gain_factor = get_max_local_gain(s, frame, -1);
+        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, current_gain_factor);
+            update_gain_history(s, c, gain);
     } else {
         int c;
 
@@ -659,12 +694,9 @@ 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;
@@ -675,12 +707,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     DynamicAudioNormalizerContext *s = ctx->priv;
-    AVFilterLink *outlink = inlink->dst->outputs[0];
+    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);
 
@@ -689,9 +723,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     }
 
     av_frame_make_writable(in);
-    cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
     analyze_frame(s, in);
-    ff_bufqueue_add(ctx, &s->queue, 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;
 }
@@ -717,7 +755,6 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
         }
     }
 
-    s->delay--;
     return filter_frame(inlink, out);
 }
 
@@ -734,7 +771,6 @@ static int flush(AVFilterLink *outlink)
 
         s->pts = out->pts;
         ret = ff_filter_frame(outlink, out);
-        s->delay = s->queue.available;
     }
 
     return ret;
@@ -772,10 +808,10 @@ static int activate(AVFilterContext *ctx)
             s->eof = 1;
     }
 
-    if (s->eof && s->delay > 0)
+    if (s->eof && s->queue.available)
         return flush(outlink);
 
-    if (s->eof && s->delay <= 0) {
+    if (s->eof && !s->queue.available) {
         ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
         return 0;
     }
@@ -786,6 +822,34 @@ 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",
@@ -815,4 +879,5 @@ AVFilter ff_af_dynaudnorm = {
     .outputs       = avfilter_af_dynaudnorm_outputs,
     .priv_class    = &dynaudnorm_class,
     .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+    .process_command = process_command,
 };