[FFmpeg-cvslog] avfilter/af_dynaudnorm: add support for overlapping frames
Paul B Mahol
git at videolan.org
Mon Feb 28 22:59:19 EET 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Feb 28 09:19:42 2022 +0100| [456d48c752fb960508b373afa3f56d389c22a8e4] | committer: Paul B Mahol
avfilter/af_dynaudnorm: add support for overlapping frames
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=456d48c752fb960508b373afa3f56d389c22a8e4
---
doc/filters.texi | 7 +++++++
libavfilter/af_dynaudnorm.c | 44 +++++++++++++++++++++++++++++++++++++++-----
2 files changed, 46 insertions(+), 5 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index b3ae139613..c3623dee89 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -4392,6 +4392,13 @@ This option is mostly useful if digital noise is not wanted to be amplified.
@item channels, h
Specify which channels to filter, by default all available channels are filtered.
+
+ at item overlap, o
+Specify overlap for frames. If set to 0 (default) no frame overlapping is done.
+Using >0 and <1 values will make less conservative gain adjustments, like
+when framelen option is set to smaller value, if framelen option value is
+compensated for non-zero overlap then gain adjustments will be smoother across time
+compared to zero overlap case.
@end table
@subsection Commands
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index 77899474bf..2adbcf3e10 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/libavfilter/af_dynaudnorm.c
@@ -64,6 +64,7 @@ typedef struct DynamicAudioNormalizerContext {
int dc_correction;
int channels_coupled;
int alt_boundary_mode;
+ double overlap;
double peak_value;
double max_amplification;
@@ -76,6 +77,7 @@ typedef struct DynamicAudioNormalizerContext {
double *weights;
int channels;
+ int sample_advance;
int eof;
uint64_t channels_to_filter;
int64_t pts;
@@ -86,6 +88,8 @@ typedef struct DynamicAudioNormalizerContext {
cqueue **threshold_history;
cqueue *is_enabled;
+
+ AVFrame *window;
} DynamicAudioNormalizerContext;
#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
@@ -114,6 +118,8 @@ static const AVOption dynaudnorm_options[] = {
{ "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 },
+ { "overlap", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
+ { "o", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
{ NULL }
};
@@ -295,6 +301,8 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->weights);
ff_bufqueue_discard_all(&s->queue);
+
+ av_frame_free(&s->window);
}
static int config_input(AVFilterLink *inlink)
@@ -340,6 +348,11 @@ static int config_input(AVFilterLink *inlink)
init_gaussian_filter(s);
+ s->window = ff_get_audio_buffer(ctx->outputs[0], s->frame_len * 2);
+ if (!s->window)
+ return AVERROR(ENOMEM);
+ s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
+
return 0;
}
@@ -649,6 +662,8 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink, AVFrame **frame)
{
+ AVFrame *analyze_frame;
+
if (s->dc_correction || s->compress_factor > DBL_EPSILON) {
int ret;
@@ -683,8 +698,26 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
if (s->compress_factor > DBL_EPSILON)
perform_compression(s, *frame);
+ if (s->frame_len != s->sample_advance) {
+ const int offset = s->frame_len - s->sample_advance;
+
+ for (int c = 0; c < s->channels; c++) {
+ double *src = (double *)s->window->extended_data[c];
+
+ memmove(src, &src[s->sample_advance], offset * sizeof(double));
+ memcpy(&src[offset], (*frame)->extended_data[c], (*frame)->nb_samples * sizeof(double));
+ memset(&src[offset + (*frame)->nb_samples], 0, (s->sample_advance - (*frame)->nb_samples) * sizeof(double));
+ }
+
+ analyze_frame = s->window;
+ } else {
+ av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0,
+ s->frame_len, (*frame)->channels, (*frame)->format);
+ analyze_frame = *frame;
+ }
+
if (s->channels_coupled) {
- const local_gain gain = get_max_local_gain(s, *frame, -1);
+ const local_gain gain = get_max_local_gain(s, analyze_frame, -1);
int c;
for (c = 0; c < s->channels; c++)
@@ -693,7 +726,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
int c;
for (c = 0; c < s->channels; c++)
- update_gain_history(s, c, get_max_local_gain(s, *frame, c));
+ update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
}
return 0;
@@ -777,7 +810,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
AVFilterLink *outlink)
{
- AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
+ AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance);
int c, i;
if (!out)
@@ -830,7 +863,7 @@ static int activate(AVFilterContext *ctx)
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (!s->eof) {
- ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in);
+ ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
if (ret < 0)
return ret;
if (ret > 0) {
@@ -839,7 +872,7 @@ static int activate(AVFilterContext *ctx)
return ret;
}
- if (ff_inlink_check_available_samples(inlink, s->frame_len) > 0) {
+ if (ff_inlink_check_available_samples(inlink, s->sample_advance) > 0) {
ff_filter_set_ready(ctx, 10);
return 0;
}
@@ -888,6 +921,7 @@ static int process_command(AVFilterContext *ctx, const char *cmd, const char *ar
}
s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
+ s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
return 0;
}
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