The change in the commit will add some samples to the end of the audio
stream. The intention is to add a "zero_delay" option eventually to not
have the delay in the begining the output from alimiter due to
lookahead.
Signed-off-by: Wang Cao <wang...@google.com>
---
doc/filters.texi | 5 ++++
libavfilter/af_alimiter.c | 57 ++++++++++++++++++++++++++++++++-------
2 files changed, 53 insertions(+), 9 deletions(-)
If the intention is clear to you, do you prefer us add the "zero_delay"
option to the same patch or it needs to go in a separate patch? Thanks!
diff --git a/doc/filters.texi b/doc/filters.texi
index d70ac3e237..bb8f7c1a0b 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1978,6 +1978,11 @@ in release time while 1 produces higher release times.
@item level
Auto level output signal. Default is enabled.
This normalizes audio back to 0dB if enabled.
+
+@item flush_buffer
+Flushes the internal buffer so that all the input audio samples to the limiter
+will appear to the output. Currently due to lookahead buffer, the total number
+of output samples will be larger than the input.
@end table
Depending on picked setting it is recommended to upsample input 2x or 4x times
diff --git a/libavfilter/af_alimiter.c b/libavfilter/af_alimiter.c
index 133f98f165..ba0a1361ac 100644
--- a/libavfilter/af_alimiter.c
+++ b/libavfilter/af_alimiter.c
@@ -55,6 +55,9 @@ typedef struct AudioLimiterContext {
int *nextpos;
double *nextdelta;
+ int flush_buffer;
+ int total_samples_to_flush;
+
double delta;
int nextiter;
int nextlen;
@@ -65,14 +68,15 @@ typedef struct AudioLimiterContext {
#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM |
AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption alimiter_options[] = {
- { "level_in", "set input level", OFFSET(level_in),
AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
- { "level_out", "set output level", OFFSET(level_out),
AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
- { "limit", "set limit", OFFSET(limit),
AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, AF },
- { "attack", "set attack", OFFSET(attack),
AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, AF },
- { "release", "set release", OFFSET(release),
AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, AF },
- { "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL,
{.i64=0}, 0, 1, AF },
- { "asc_level", "set asc level", OFFSET(asc_coeff),
AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
- { "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL,
{.i64=1}, 0, 1, AF },
+ { "level_in", "set input level",
OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
+ { "level_out", "set output level",
OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
+ { "limit", "set limit",
OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, AF },
+ { "attack", "set attack",
OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, AF },
+ { "release", "set release",
OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, AF },
+ { "asc", "enable asc",
OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
+ { "asc_level", "set asc level",
OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
+ { "level", "auto level",
OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
+ { "flush_buffer","flush the samples in the lookahead buffer",
OFFSET(flush_buffer), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
{ NULL }
};
@@ -275,6 +279,39 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
return ff_filter_frame(outlink, out);
}
+
+static int request_frame(AVFilterLink* outlink) {
+ AVFilterContext* ctx = outlink->src;
+ AudioLimiterContext* s = (AudioLimiterContext*)ctx->priv;
+ int ret;
+ AVFilterLink *inlink;
+ AVFrame *silence_frame;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (!s->flush_buffer) {
+ return 0;
+ }
+
+ if (ret != AVERROR_EOF || s->total_samples_to_flush) {
+ // Not necessarily an error, just not EOF.s
+ return ret;
+ }
+
+ // We reach here when input filters have finished producing data (i.e. EOF),
+ // but because of the attac param, s->buffer still has meaningful
+ // audio content that needs flushing. The amount of remaining audio to flush
+ // is the same as the amount of lookahead that was trimmed from the
beginning.
+ inlink = ctx->inputs[0];
+ // Pushes silence frame to flush valid audio in the s->buffer
+ silence_frame = ff_get_audio_buffer(inlink, s->total_samples_to_flush);
+ ret = filter_frame(inlink, silence_frame);
+ if (ret < 0) {
+ return ret;
+ }
+ return AVERROR_EOF;
+}
+
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
@@ -292,7 +329,8 @@ static int config_input(AVFilterLink *inlink)
return AVERROR(ENOMEM);
memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
- s->buffer_size = inlink->sample_rate * s->attack *
inlink->ch_layout.nb_channels;
+ s->total_samples_to_flush = inlink->sample_rate * s->attack;
+ s->buffer_size = s->total_samples_to_flush * inlink->ch_layout.nb_channels;
s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
if (s->buffer_size <= 0) {
@@ -325,6 +363,7 @@ static const AVFilterPad alimiter_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
},
};
--
2.35.1.1021.g381101b075-goog
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