[FFmpeg-devel] [PATCH] avfilter/alimiter:add latency compensation
Wang Cao
wangcao at google.com
Fri May 6 00:14:16 EEST 2022
Also added 2 FATE tests to verify delay is compenated correctly
Signed-off-by: Wang Cao <wangcao at google.com>
---
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.
+
+ at 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_input(AVFilterLink *inlink)
memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
s->buffer_size = inlink->sample_rate * s->attack * inlink->ch_layout.nb_channels;
s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
+ if (s->latency) {
+ s->in_trim = s->out_pad = s->buffer_size / inlink->ch_layout.nb_channels - 1;
+ }
+ s->next_out_pts = AV_NOPTS_VALUE;
+ s->next_in_pts = AV_NOPTS_VALUE;
+
+ s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW);
+ if (!s->fifo) {
+ return AVERROR(ENOMEM);
+ }
if (s->buffer_size <= 0) {
av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
@@ -310,6 +397,8 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->buffer);
av_freep(&s->nextdelta);
av_freep(&s->nextpos);
+
+ av_fifo_freep2(&s->fifo);
}
static const AVFilterPad alimiter_inputs[] = {
@@ -325,6 +414,7 @@ static const AVFilterPad alimiter_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
},
};
diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
index eff32b9f81..e33ffdf37f 100644
--- a/tests/fate/filter-audio.mak
+++ b/tests/fate/filter-audio.mak
@@ -63,11 +63,29 @@ fate-filter-agate: tests/data/asynth-44100-2.wav
fate-filter-agate: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-agate: CMD = framecrc -i $(SRC) -af aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample
-FATE_AFILTER-$(call FILTERDEMDECENCMUX, AFADE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-alimiter
-fate-filter-alimiter: tests/data/asynth-44100-2.wav
-fate-filter-alimiter: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
+tests/data/filter-alimiter-passthrough: TAG = GEN
+tests/data/filter-alimiter-passthrough: ffmpeg$(PROGSSUF)$(EXESUF) | tests/data
+ $(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
+ -i $(TARGET_PATH)/tests/data/asynth-44100-2.wav -af aresample -f crc $(TARGET_PATH)/$@ -y 2>/dev/null
+
+FATE_ALIMITER += fate-filter-alimiter-passthrough-default-attack
+fate-filter-alimiter-passthrough-default-attack: tests/data/filter-alimiter-passthrough
+fate-filter-alimiter-passthrough-default-attack: REF = $(TARGET_PATH)/tests/data/filter-alimiter-passthrough
+fate-filter-alimiter-passthrough-default-attack: CMD = crc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1,aresample
+
+FATE_ALIMITER += fate-filter-alimiter-passthrough-large-attack
+fate-filter-alimiter-passthrough-large-attack: tests/data/filter-alimiter-passthrough
+fate-filter-alimiter-passthrough-large-attack: REF = $(TARGET_PATH)/tests/data/filter-alimiter-passthrough
+fate-filter-alimiter-passthrough-large-attack: CMD = crc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1:attack=80,aresample
+
+FATE_ALIMITER += fate-filter-alimiter
fate-filter-alimiter: CMD = framecrc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=2:limit=0.2,aresample
+$(FATE_ALIMITER): tests/data/asynth-44100-2.wav
+$(FATE_ALIMITER): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
+
+FATE_AFILTER-$(call FILTERDEMDECENCMUX, ATRIM, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_ALIMITER)
+
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AMERGE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-amerge
fate-filter-amerge: tests/data/asynth-44100-1.wav
fate-filter-amerge: SRC = $(TARGET_PATH)/tests/data/asynth-44100-1.wav
--
2.36.0.512.ge40c2bad7a-goog
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