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_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
