avfilter/af_dynaudnorm: allow to filter subset of channels

Paul B Mahol committed 4 years ago

Commit a9124a71 parent b9f91a72 files · +24 −4Browse files

Unified · Split

2 files

doc/filters.texi +3 −0
libavfilter/af_dynaudnorm.c +21 −4

@@ -4389,6 +4389,9 @@ If input frame volume is above this value frame will be normalized. · 4388 unmodified lines
4389	Otherwise frame may not be normalized at all. The default value is set	4389	Otherwise frame may not be normalized at all. The default value is set
4390	to 0, which means all input frames will be normalized.	4390	to 0, which means all input frames will be normalized.
4391	This option is mostly useful if digital noise is not wanted to be amplified.	4391	This option is mostly useful if digital noise is not wanted to be amplified.
		4392
		4393	@item channels, h
		4394	Specify which channels to filter, by default all available channels are filtered.
4392	@end table	4395	@end table
4393		4396
4394	@subsection Commands	4397	@subsection Commands

libavfilter/af_dynaudnorm.c+21−4

 #include <float.h>
 #include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #define MIN_FILTER_SIZE 3
     int channels;
     int eof;
+    uint64_t channels_to_filter;
     int64_t pts;
     cqueue **gain_history_original;
     { "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_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
+    { "h",           "set channels to filter",           OFFSET(channels_to_filter),AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
     { NULL }
 };
     return aggressiveness * new + (1.0 - aggressiveness) * old;
 }
+static inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch)
+{
+    return !(av_channel_layout_extract_channel(frame->channel_layout, ch) & s->channels_to_filter);
+}
 static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
 {
     const double diff = 1.0 / frame->nb_samples;
     int c, i;
     for (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;
         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++) {
+        for (i = 0; i < frame->nb_samples && !bypass; i++) {
             dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
         }
     }
         for (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 (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 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(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++) {
+            for (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]);
             }
     int c, i;
     for (c = 0; c < s->channels; 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], &current_amplification_factor);
-        for (i = 0; i < frame->nb_samples && enabled; i++) {
+        for (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);

avfilter/af_dynaudnorm: allow to filter subset of channels

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