diff --git a/audio/filter/af.c b/audio/filter/af.c
index 846ebea3c3..2889e87bb7 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -54,7 +54,6 @@ extern const struct af_info af_info_center;
 extern const struct af_info af_info_sinesuppress;
 extern const struct af_info af_info_karaoke;
 extern const struct af_info af_info_scaletempo;
-extern const struct af_info af_info_forcespeed;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
 extern const struct af_info af_info_convert24;
@@ -87,7 +86,6 @@ static const struct af_info *const filter_list[] = {
     &af_info_center,
     &af_info_sinesuppress,
     &af_info_karaoke,
-    &af_info_forcespeed,
     &af_info_scaletempo,
 #if HAVE_LIBBS2B
     &af_info_bs2b,
diff --git a/audio/filter/af_forcespeed.c b/audio/filter/af_forcespeed.c
deleted file mode 100644
index 3f2ca831c9..0000000000
--- a/audio/filter/af_forcespeed.c
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include "af.h"
-
-struct priv {
-    double speed;
-};
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct priv *priv = af->priv;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        mp_audio_copy_config(out, in);
-        out->rate = in->rate * priv->speed;
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
-        priv->speed = *(double *)arg;
-        return AF_OK;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    if (data)
-        mp_audio_copy_config(data, af->data);
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    struct priv *priv = af->priv;
-    af->control = control;
-    af->filter_frame = filter;
-    priv->speed = 1.0;
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct priv
-
-const struct af_info af_info_forcespeed = {
-    .info = "Force audio speed",
-    .name = "forcespeed",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-};
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index dc2f0628b7..48f69b72e8 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -24,6 +24,7 @@
 #include <stdlib.h>
 #include <string.h>
 #include <inttypes.h>
+#include <math.h>
 #include <assert.h>
 
 #include <libavutil/opt.h>
@@ -32,7 +33,7 @@
 #include <libavutil/samplefmt.h>
 #include <libavutil/mathematics.h>
 
-#include "talloc.h"
+#include "common/common.h"
 #include "config.h"
 
 #if HAVE_LIBAVRESAMPLE
@@ -64,7 +65,8 @@ struct af_resample_opts {
     int linear;
     double cutoff;
 
-    int in_rate;
+    int in_rate_af; // filter input sample rate
+    int in_rate;    // actual rate (used by lavr), adjusted for playback speed
     int in_format;
     struct mp_chmap in_channels;
     int out_rate;
@@ -75,6 +77,9 @@ struct af_resample_opts {
 struct af_resample {
     int allow_detach;
     char **avopts;
+    double playback_speed;
+    struct mp_audio *pending;
+    bool avrctx_ok;
     struct AVAudioResampleContext *avrctx;
     struct AVAudioResampleContext *avrctx_out; // for output channel reordering
     struct af_resample_opts ctx;   // opts in the context
@@ -94,6 +99,10 @@ static void drop_all_output(struct af_resample *s)
 {
     while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
 }
+static int get_drain_samples(struct af_resample *s)
+{
+    return avresample_get_out_samples(s->avrctx, 0);
+}
 #else
 static int get_delay(struct af_resample *s)
 {
@@ -103,18 +112,39 @@ static void drop_all_output(struct af_resample *s)
 {
     while (swr_drop_output(s->avrctx, 1000) > 0) {}
 }
+static int get_drain_samples(struct af_resample *s)
+{
+    return 4096; // libswscale does not have this
+}
 #endif
 
+static int resample_frame(struct AVAudioResampleContext *r,
+                          struct mp_audio *out, struct mp_audio *in)
+{
+    return avresample_convert(r,
+        out ? (uint8_t **)out->planes : NULL,
+        out ? mp_audio_get_allocated_size(out) : 0,
+        out ? out->samples : 0,
+        in ? (uint8_t **)in->planes : NULL,
+        in ? mp_audio_get_allocated_size(in) : 0,
+        in ? in->samples : 0);
+}
+
 static double af_resample_default_cutoff(int filter_size)
 {
     return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
 }
 
+static int rate_from_speed(int rate, double speed)
+{
+    return lrint(rate * speed);
+}
+
 static bool needs_lavrctx_reconfigure(struct af_resample *s,
                                       struct mp_audio *in,
                                       struct mp_audio *out)
 {
-    return s->ctx.in_rate     != in->rate ||
+    return s->ctx.in_rate_af  != in->rate ||
            s->ctx.in_format   != in->format ||
            !mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
            s->ctx.out_rate    != out->rate ||
@@ -138,6 +168,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
 {
     struct af_resample *s = af->priv;
 
+    s->avrctx_ok = false;
+
     enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
     enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
 
@@ -147,8 +179,12 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
     avresample_close(s->avrctx);
     avresample_close(s->avrctx_out);
 
+    talloc_free(s->pending);
+    s->pending = NULL;
+
     s->ctx.out_rate    = out->rate;
-    s->ctx.in_rate     = in->rate;
+    s->ctx.in_rate_af  = in->rate;
+    s->ctx.in_rate     = rate_from_speed(in->rate, s->playback_speed);
     s->ctx.out_format  = out->format;
     s->ctx.in_format   = in->format;
     s->ctx.out_channels= out->channels;
@@ -217,6 +253,7 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
         MP_ERR(af, "Cannot open Libavresample Context. \n");
         return AF_ERROR;
     }
+    s->avrctx_ok = true;
     return AF_OK;
 }
 
@@ -234,7 +271,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (((out->rate    == in->rate) || (out->rate == 0)) &&
             (out->format   == in->format) &&
             (mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
-            s->allow_detach)
+            s->allow_detach && s->playback_speed == 1.0)
             return AF_DETACH;
 
         if (out->rate == 0)
@@ -270,6 +307,26 @@ static int control(struct af_instance *af, int cmd, void *arg)
     case AF_CONTROL_SET_RESAMPLE_RATE:
         out->rate = *(int *)arg;
         return AF_OK;
+    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
+        s->playback_speed = *(double *)arg;
+        int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
+        if (new_rate != s->ctx.in_rate && s->avrctx_ok && af->fmt_out.format) {
+            // Before reconfiguring, drain the audio that is still buffered
+            // in the resampler.
+            talloc_free(s->pending);
+            s->pending = talloc_zero(NULL, struct mp_audio);
+            mp_audio_copy_config(s->pending, &af->fmt_out);
+            s->pending->samples = get_drain_samples(s);
+            if (s->pending->samples > 0) {
+                mp_audio_realloc_min(s->pending, s->pending->samples);
+                int r = resample_frame(s->avrctx, s->pending, NULL);
+                s->pending->samples = MPMAX(r, 0);
+            }
+            // Reinitialize resampler.
+            configure_lavrr(af, &af->fmt_in, &af->fmt_out);
+        }
+        return AF_OK;
+    }
     case AF_CONTROL_RESET:
         drop_all_output(s);
         return AF_OK;
@@ -289,6 +346,7 @@ static void uninit(struct af_instance *af)
     if (s->avrctx_out)
         avresample_close(s->avrctx_out);
     avresample_free(&s->avrctx_out);
+    talloc_free(s->pending);
 }
 
 static bool needs_reorder(int *reorder, int num_ch)
@@ -309,22 +367,19 @@ static void reorder_planes(struct mp_audio *mpa, int *reorder)
     }
 }
 
-static int resample_frame(struct AVAudioResampleContext *r,
-                          struct mp_audio *out, struct mp_audio *in)
-{
-    return avresample_convert(r,
-        out ? (uint8_t **)out->planes : NULL,
-        out ? mp_audio_get_allocated_size(out) : 0,
-        out ? out->samples : 0,
-        in ? (uint8_t **)in->planes : NULL,
-        in ? mp_audio_get_allocated_size(in) : 0,
-        in ? in->samples : 0);
-}
-
 static int filter(struct af_instance *af, struct mp_audio *in)
 {
     struct af_resample *s = af->priv;
 
+    if (s->pending) {
+        if (s->pending->samples) {
+            af_add_output_frame(af, s->pending);
+        } else {
+            talloc_free(s->pending);
+        }
+        s->pending = NULL;
+    }
+
     int samples = avresample_available(s->avrctx) +
         av_rescale_rnd(get_delay(s) + (in ? in->samples : 0),
                        s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
@@ -412,6 +467,7 @@ const struct af_info af_info_lavrresample = {
             .cutoff      = 0.0,
             .phase_shift = 10,
         },
+        .playback_speed = 1.0,
         .allow_detach = 1,
     },
     .options = (const struct m_option[]) {
diff --git a/old-makefile b/old-makefile
index 08ab079d06..b9fbbaaa50 100644
--- a/old-makefile
+++ b/old-makefile
@@ -134,7 +134,6 @@ SOURCES = audio/audio.c \
           audio/filter/af_equalizer.c \
           audio/filter/af_export.c \
           audio/filter/af_extrastereo.c \
-          audio/filter/af_forcespeed.c \
           audio/filter/af_format.c \
           audio/filter/af_hrtf.c \
           audio/filter/af_karaoke.c \
diff --git a/player/audio.c b/player/audio.c
index c4329f07c6..79be29a359 100644
--- a/player/audio.c
+++ b/player/audio.c
@@ -66,7 +66,6 @@ static int recreate_audio_filters(struct MPContext *mpctx)
 
     struct MPOpts *opts = mpctx->opts;
     struct af_stream *afs = mpctx->d_audio->afilter;
-    bool need_reinit = false;
 
     double speed = opts->playback_speed;
 
@@ -84,7 +83,7 @@ static int recreate_audio_filters(struct MPContext *mpctx)
             if (!af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed))
             {
                 char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
-                             ? "scaletempo" : "forcespeed";
+                             ? "scaletempo" : "lavrresample";
                 if (try_filter(mpctx, filter, "playback-speed", NULL) < 0)
                     return -1;
                 // Try again.
@@ -94,11 +93,6 @@ static int recreate_audio_filters(struct MPContext *mpctx)
                 method = AF_CONTROL_SET_PLAYBACK_SPEED;
             }
         }
-        // AF_CONTROL_SET_PLAYBACK_SPEED does not require reinitialization,
-        // while AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE requires updating
-        // the samplerate on the resampler, and possibly inserting the
-        // resampler itself.
-        need_reinit |= (method == AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE);
     } else {
         if (af_remove_by_label(afs, "playback-speed") < 0)
             return -1;
@@ -107,9 +101,7 @@ static int recreate_audio_filters(struct MPContext *mpctx)
         af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &speed);
     }
 
-    need_reinit |= afs->initialized < 1;
-
-    if (need_reinit && af_init(afs) < 0) {
+    if (afs->initialized < 1 && af_init(afs) < 0) {
         MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
         return -1;
     }
diff --git a/wscript_build.py b/wscript_build.py
index c57cb9ed76..7c4c77d709 100644
--- a/wscript_build.py
+++ b/wscript_build.py
@@ -109,7 +109,6 @@ def build(ctx):
         ( "audio/filter/af_equalizer.c" ),
         ( "audio/filter/af_export.c" ),
         ( "audio/filter/af_extrastereo.c" ),
-        ( "audio/filter/af_forcespeed.c" ),
         ( "audio/filter/af_format.c" ),
         ( "audio/filter/af_hrtf.c" ),
         ( "audio/filter/af_karaoke.c" ),