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-rw-r--r--audio/filter/af.c2
-rw-r--r--audio/filter/af_forcespeed.c71
-rw-r--r--audio/filter/af_lavrresample.c90
-rw-r--r--old-makefile1
-rw-r--r--player/audio.c12
-rw-r--r--wscript_build.py1
6 files changed, 75 insertions, 102 deletions
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" ),
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-25 08:17:06 +0000