[FFmpeg-cvslog] avfilter/af_surround: implement smooth option/support
Paul B Mahol
git at videolan.org
Fri Nov 18 22:49:23 EET 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Nov 18 20:20:32 2022 +0100| [52291d2ac8835869116f86f03ac79efdcd99031e] | committer: Paul B Mahol
avfilter/af_surround: implement smooth option/support
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=52291d2ac8835869116f86f03ac79efdcd99031e
---
doc/filters.texi | 6 ++++++
libavfilter/af_surround.c | 22 ++++++++++++++++++----
2 files changed, 24 insertions(+), 4 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index e4da63cf48..f6385a597c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -6670,6 +6670,12 @@ In @var{add} mode, LFE channel is created from input audio and added to output.
In @var{sub} mode, LFE channel is created from input audio and added to output but
also all non-LFE output channels are subtracted with output LFE channel.
+ at item smooth
+Set temporal smoothness strength, used to gradually change factors when transforming
+stereo sound in time. Allowed range is from @var{0.0} to @var{1.0}.
+Useful to improve output quality with @var{focus} option values greater than @var{0.0}.
+Default is @var{0.0}. Only values inside this range and without edges are effective.
+
@item angle
Set angle of stereo surround transform, Allowed range is from @var{0} to @var{360}.
Default is @var{90}.
diff --git a/libavfilter/af_surround.c b/libavfilter/af_surround.c
index c6360365fd..80f6c00dd3 100644
--- a/libavfilter/af_surround.c
+++ b/libavfilter/af_surround.c
@@ -69,6 +69,7 @@ typedef struct AudioSurroundContext {
float f_i[SC_NB];
float f_o[SC_NB];
int lfe_mode;
+ float smooth;
float angle;
float focus;
int win_size;
@@ -97,6 +98,7 @@ typedef struct AudioSurroundContext {
int nb_out_channels;
AVFrame *factors;
+ AVFrame *sfactors;
AVFrame *input_in;
AVFrame *input;
AVFrame *output;
@@ -297,13 +299,14 @@ static int config_output(AVFilterLink *outlink)
set_output_levels(ctx);
s->factors = ff_get_audio_buffer(outlink, s->win_size + 2);
+ s->sfactors = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_ph = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_mag = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_out = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output = ff_get_audio_buffer(outlink, s->win_size + 2);
s->overlap_buffer = ff_get_audio_buffer(outlink, s->win_size * 2);
if (!s->overlap_buffer || !s->output || !s->output_out || !s->output_mag ||
- !s->output_ph || !s->factors)
+ !s->output_ph || !s->factors || !s->sfactors)
return AVERROR(ENOMEM);
s->rdft_size = s->win_size / 2 + 1;
@@ -404,6 +407,7 @@ static int stereo_upmix(AVFilterContext *ctx, int ch)
{
AudioSurroundContext *s = ctx->priv;
float *dst = (float *)s->output->extended_data[ch];
+ float *sfactor = (float *)s->sfactors->extended_data[ch];
float *factor = (float *)s->factors->extended_data[ch];
float *omag = (float *)s->output_mag->extended_data[ch];
float *oph = (float *)s->output_ph->extended_data[ch];
@@ -419,6 +423,7 @@ static int stereo_upmix(AVFilterContext *ctx, int ch)
const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
const float f_x = s->f_x[sc_map[chan]];
const float f_y = s->f_y[sc_map[chan]];
+ const float smooth = s->smooth;
switch (chan) {
case AV_CHAN_FRONT_CENTER:
@@ -501,6 +506,13 @@ static int stereo_upmix(AVFilterContext *ctx, int ch)
break;
}
+ if (smooth > 0.f) {
+ for (int n = 0; n < rdft_size; n++)
+ sfactor[n] = smooth * factor[n] + (1.f - smooth) * sfactor[n];
+
+ factor = sfactor;
+ }
+
for (int n = 0; n < rdft_size; n++)
omag[n] *= factor[n];
@@ -1076,9 +1088,9 @@ static void allchannels_spread(AVFilterContext *ctx)
static av_cold int init(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
- float overlap;
int64_t in_channel_layout, out_channel_layout;
- int i, ret;
+ float overlap;
+ int ret;
if ((ret = av_channel_layout_from_string(&s->out_ch_layout, s->out_channel_layout_str)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing output channel layout '%s'.\n",
@@ -1203,7 +1215,7 @@ fail:
if (s->overlap == 1)
s->overlap = overlap;
- for (i = 0; i < s->win_size; i++)
+ for (int i = 0; i < s->win_size; i++)
s->window_func_lut[i] = sqrtf(s->window_func_lut[i] / s->win_size);
s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap));
@@ -1354,6 +1366,7 @@ static av_cold void uninit(AVFilterContext *ctx)
AudioSurroundContext *s = ctx->priv;
av_frame_free(&s->factors);
+ av_frame_free(&s->sfactors);
av_frame_free(&s->window);
av_frame_free(&s->input_in);
av_frame_free(&s->input);
@@ -1417,6 +1430,7 @@ static const AVOption surround_options[] = {
{ "lfe_mode", "set LFE channel mode", OFFSET(lfe_mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, TFLAGS, "lfe_mode" },
{ "add", "just add LFE channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 1, TFLAGS, "lfe_mode" },
{ "sub", "substract LFE channel with others", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 1, TFLAGS, "lfe_mode" },
+ { "smooth", "set temporal smoothness strength", OFFSET(smooth), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, TFLAGS },
{ "angle", "set soundfield transform angle", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl=90}, 0, 360, TFLAGS },
{ "focus", "set soundfield transform focus", OFFSET(focus), AV_OPT_TYPE_FLOAT, {.dbl=0}, -1, 1, TFLAGS },
{ "fc_in", "set front center channel input level", OFFSET(f_i[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
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