Re: [FFmpeg-devel] [PATCH] avfilter/alimiter:add latency compensation
On Wed, May 11, 2022 at 12:45 AM Wang Cao
wrote:
> On Thu, May 5, 2022 at 2:14 PM Wang Cao wrote:
>
> > Also added 2 FATE tests to verify delay is compenated correctly
>
Applied with minor fixes and tests removed for now.
> >
> > Signed-off-by: Wang Cao
> > ---
> > doc/filters.texi| 5 +++
> > libavfilter/af_alimiter.c | 90 +
> > tests/fate/filter-audio.mak | 24 --
> > 3 files changed, 116 insertions(+), 3 deletions(-)
> >
> > diff --git a/doc/filters.texi b/doc/filters.texi
> > index a161754233..75a43edd88 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 latency
> > +Compensate the delay introduced by using the lookahead buffer set with
> > attack
> > +parameter. Also flush the valid audio data in the lookahead buffer when
> > the
> > +stream hits EOF
> > @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..01265758d7 100644
> > --- a/libavfilter/af_alimiter.c
> > +++ b/libavfilter/af_alimiter.c
> > @@ -26,6 +26,7 @@
> >
> > #include "libavutil/channel_layout.h"
> > #include "libavutil/common.h"
> > +#include "libavutil/fifo.h"
> > #include "libavutil/opt.h"
> >
> > #include "audio.h"
> > @@ -33,6 +34,11 @@
> > #include "formats.h"
> > #include "internal.h"
> >
> > +typedef struct MetaItem {
> > +int64_t pts;
> > +int nb_samples;
> > +} MetaItem;
> > +
> > typedef struct AudioLimiterContext {
> > const AVClass *class;
> >
> > @@ -55,6 +61,14 @@ typedef struct AudioLimiterContext {
> > int *nextpos;
> > double *nextdelta;
> >
> > +int in_trim;
> > +int out_pad;
> > +int64_t next_in_pts;
> > +int64_t next_out_pts;
> > +int latency;
> > +
> > +AVFifo *fifo;
> > +
> > double delta;
> > int nextiter;
> > int nextlen;
> > @@ -73,6 +87,7 @@ static const AVOption alimiter_options[] = {
> > { "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 },
> > +{ "latency", "compensate delay", OFFSET(latency),
> > AV_OPT_TYPE_BOOL, {.i64=0}, 0,1, AF },
> > { NULL }
> > };
> >
> > @@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink,
> AVFrame
> > *in)
> > AVFrame *out;
> > double *buf;
> > int n, c, i;
> > +int new_out_samples;
> > +int64_t out_duration;
> > +int64_t in_duration;
> > +int64_t in_pts;
> > +MetaItem meta;
> >
> > if (av_frame_is_writable(in)) {
> > out = in;
> > @@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink,
> > AVFrame *in)
> > dst += channels;
> > }
> >
> > +in_duration = av_rescale_q(in->nb_samples, inlink->time_base,
> > av_make_q(1, in->sample_rate));
> > +in_pts = in->pts;
> > +meta = (MetaItem){ in->pts, in->nb_samples };
> > +av_fifo_write(s->fifo, &meta, 1);
> > if (in != out)
> > av_frame_free(&in);
> >
> > +new_out_samples = out->nb_samples;
> > +if (s->in_trim > 0) {
> > +int trim = FFMIN(new_out_samples, s->in_trim);
> > +new_out_samples -= trim;
> > +s->in_trim -= trim;
> > +}
> > +
> > +if (new_out_samples <= 0) {
> > +av_frame_free(&out);
> > +return 0;
> > +} else if (new_out_samples < out->nb_samples) {
> > +int offset = out->nb_samples - new_out_samples;
> > +memmove(out->extended_data[0], out->extended_data[0] +
> > sizeof(double) * offset * out->ch_layout.nb_channels,
> > +sizeof(double) * new_out_samples *
> > out->ch_layout.nb_channels);
> > +out->nb_samples = new_out_samples;
> > +s->in_trim = 0;
> > +}
> > +
> > +av_fifo_read(s->fifo, &meta, 1);
> > +
> > +out_duration = av_rescale_q(out->nb_samples, inlink->time_base,
> > av_make_q(1, out->sample_rate));
> > +in_duration = av_rescale_q(meta.nb_samples, inlink->time_base,
> > av_make_q(1, out->sample_rate));
> > +in_pts = meta.pts;
> > +
> > +if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts
> > &&
> > +s->next_in_pts != AV_NOPTS_VALUE && in_pts ==
> s->next_in_pts) {
> > +out->pts = s->next_out_pts;
> > +} else {
> > +out->pts = in_pts;
> > +}
> > +s->next_in_pts = in_pts + in_duration;
> > +s->next_out_pts = out->pts + out_duration;
> > +
> > return ff_filter_frame(outlink, out);
> >
Re: [FFmpeg-devel] [PATCH] avfilter/alimiter:add latency compensation
On Thu, May 5, 2022 at 2:14 PM Wang Cao wrote:
> Also added 2 FATE tests to verify delay is compenated correctly
>
> Signed-off-by: Wang Cao
> ---
> doc/filters.texi| 5 +++
> libavfilter/af_alimiter.c | 90 +
> tests/fate/filter-audio.mak | 24 --
> 3 files changed, 116 insertions(+), 3 deletions(-)
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index a161754233..75a43edd88 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 latency
> +Compensate the delay introduced by using the lookahead buffer set with
> attack
> +parameter. Also flush the valid audio data in the lookahead buffer when
> the
> +stream hits EOF
> @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..01265758d7 100644
> --- a/libavfilter/af_alimiter.c
> +++ b/libavfilter/af_alimiter.c
> @@ -26,6 +26,7 @@
>
> #include "libavutil/channel_layout.h"
> #include "libavutil/common.h"
> +#include "libavutil/fifo.h"
> #include "libavutil/opt.h"
>
> #include "audio.h"
> @@ -33,6 +34,11 @@
> #include "formats.h"
> #include "internal.h"
>
> +typedef struct MetaItem {
> +int64_t pts;
> +int nb_samples;
> +} MetaItem;
> +
> typedef struct AudioLimiterContext {
> const AVClass *class;
>
> @@ -55,6 +61,14 @@ typedef struct AudioLimiterContext {
> int *nextpos;
> double *nextdelta;
>
> +int in_trim;
> +int out_pad;
> +int64_t next_in_pts;
> +int64_t next_out_pts;
> +int latency;
> +
> +AVFifo *fifo;
> +
> double delta;
> int nextiter;
> int nextlen;
> @@ -73,6 +87,7 @@ static const AVOption alimiter_options[] = {
> { "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 },
> +{ "latency", "compensate delay", OFFSET(latency),
> AV_OPT_TYPE_BOOL, {.i64=0}, 0,1, AF },
> { NULL }
> };
>
> @@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
> *in)
> AVFrame *out;
> double *buf;
> int n, c, i;
> +int new_out_samples;
> +int64_t out_duration;
> +int64_t in_duration;
> +int64_t in_pts;
> +MetaItem meta;
>
> if (av_frame_is_writable(in)) {
> out = in;
> @@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink,
> AVFrame *in)
> dst += channels;
> }
>
> +in_duration = av_rescale_q(in->nb_samples, inlink->time_base,
> av_make_q(1, in->sample_rate));
> +in_pts = in->pts;
> +meta = (MetaItem){ in->pts, in->nb_samples };
> +av_fifo_write(s->fifo, &meta, 1);
> if (in != out)
> av_frame_free(&in);
>
> +new_out_samples = out->nb_samples;
> +if (s->in_trim > 0) {
> +int trim = FFMIN(new_out_samples, s->in_trim);
> +new_out_samples -= trim;
> +s->in_trim -= trim;
> +}
> +
> +if (new_out_samples <= 0) {
> +av_frame_free(&out);
> +return 0;
> +} else if (new_out_samples < out->nb_samples) {
> +int offset = out->nb_samples - new_out_samples;
> +memmove(out->extended_data[0], out->extended_data[0] +
> sizeof(double) * offset * out->ch_layout.nb_channels,
> +sizeof(double) * new_out_samples *
> out->ch_layout.nb_channels);
> +out->nb_samples = new_out_samples;
> +s->in_trim = 0;
> +}
> +
> +av_fifo_read(s->fifo, &meta, 1);
> +
> +out_duration = av_rescale_q(out->nb_samples, inlink->time_base,
> av_make_q(1, out->sample_rate));
> +in_duration = av_rescale_q(meta.nb_samples, inlink->time_base,
> av_make_q(1, out->sample_rate));
> +in_pts = meta.pts;
> +
> +if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts
> &&
> +s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) {
> +out->pts = s->next_out_pts;
> +} else {
> +out->pts = in_pts;
> +}
> +s->next_in_pts = in_pts + in_duration;
> +s->next_out_pts = out->pts + out_duration;
> +
> return ff_filter_frame(outlink, out);
> }
>
> +static int request_frame(AVFilterLink* outlink)
> +{
> +AVFilterContext *ctx = outlink->src;
> +AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
> +int ret;
> +
> +ret = ff_request_frame(ctx->inputs[0]);
> +
> +if (ret == AVERROR_EOF && s->out_pad > 0) {
> +AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024,
> s->out_pad));
> +if (!frame)
> +
[FFmpeg-devel] [PATCH] avfilter/alimiter:add latency compensation
Also added 2 FATE tests to verify delay is compenated correctly
Signed-off-by: Wang Cao
---
doc/filters.texi| 5 +++
libavfilter/af_alimiter.c | 90 +
tests/fate/filter-audio.mak | 24 --
3 files changed, 116 insertions(+), 3 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index a161754233..75a43edd88 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 latency
+Compensate the delay introduced by using the lookahead buffer set with attack
+parameter. Also flush the valid audio data in the lookahead buffer when the
+stream hits EOF
@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..01265758d7 100644
--- a/libavfilter/af_alimiter.c
+++ b/libavfilter/af_alimiter.c
@@ -26,6 +26,7 @@
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
+#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "audio.h"
@@ -33,6 +34,11 @@
#include "formats.h"
#include "internal.h"
+typedef struct MetaItem {
+int64_t pts;
+int nb_samples;
+} MetaItem;
+
typedef struct AudioLimiterContext {
const AVClass *class;
@@ -55,6 +61,14 @@ typedef struct AudioLimiterContext {
int *nextpos;
double *nextdelta;
+int in_trim;
+int out_pad;
+int64_t next_in_pts;
+int64_t next_out_pts;
+int latency;
+
+AVFifo *fifo;
+
double delta;
int nextiter;
int nextlen;
@@ -73,6 +87,7 @@ static const AVOption alimiter_options[] = {
{ "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 },
+{ "latency", "compensate delay", OFFSET(latency), AV_OPT_TYPE_BOOL,
{.i64=0}, 0,1, AF },
{ NULL }
};
@@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFrame *out;
double *buf;
int n, c, i;
+int new_out_samples;
+int64_t out_duration;
+int64_t in_duration;
+int64_t in_pts;
+MetaItem meta;
if (av_frame_is_writable(in)) {
out = in;
@@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
dst += channels;
}
+in_duration = av_rescale_q(in->nb_samples, inlink->time_base,
av_make_q(1, in->sample_rate));
+in_pts = in->pts;
+meta = (MetaItem){ in->pts, in->nb_samples };
+av_fifo_write(s->fifo, &meta, 1);
if (in != out)
av_frame_free(&in);
+new_out_samples = out->nb_samples;
+if (s->in_trim > 0) {
+int trim = FFMIN(new_out_samples, s->in_trim);
+new_out_samples -= trim;
+s->in_trim -= trim;
+}
+
+if (new_out_samples <= 0) {
+av_frame_free(&out);
+return 0;
+} else if (new_out_samples < out->nb_samples) {
+int offset = out->nb_samples - new_out_samples;
+memmove(out->extended_data[0], out->extended_data[0] + sizeof(double)
* offset * out->ch_layout.nb_channels,
+sizeof(double) * new_out_samples * out->ch_layout.nb_channels);
+out->nb_samples = new_out_samples;
+s->in_trim = 0;
+}
+
+av_fifo_read(s->fifo, &meta, 1);
+
+out_duration = av_rescale_q(out->nb_samples, inlink->time_base,
av_make_q(1, out->sample_rate));
+in_duration = av_rescale_q(meta.nb_samples, inlink->time_base,
av_make_q(1, out->sample_rate));
+in_pts = meta.pts;
+
+if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts &&
+s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) {
+out->pts = s->next_out_pts;
+} else {
+out->pts = in_pts;
+}
+s->next_in_pts = in_pts + in_duration;
+s->next_out_pts = out->pts + out_duration;
+
return ff_filter_frame(outlink, out);
}
+static int request_frame(AVFilterLink* outlink)
+{
+AVFilterContext *ctx = outlink->src;
+AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
+int ret;
+
+ret = ff_request_frame(ctx->inputs[0]);
+
+if (ret == AVERROR_EOF && s->out_pad > 0) {
+AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, s->out_pad));
+if (!frame)
+return AVERROR(ENOMEM);
+
+s->out_pad -= frame->nb_samples;
+frame->pts = s->next_in_pts;
+return filter_frame(ctx->inputs[0], frame);
+}
+return ret;
+}
+
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
@@ -294,6 +371,16 @@ static int config
