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|
/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <libavutil/opt.h>
#include <libavutil/common.h>
#include <libavutil/samplefmt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/mathematics.h>
#include <libswresample/swresample.h>
#include "audio/aframe.h"
#include "audio/chmap_avchannel.h"
#include "audio/fmt-conversion.h"
#include "audio/format.h"
#include "common/common.h"
#include "common/av_common.h"
#include "common/msg.h"
#include "options/m_config.h"
#include "options/m_option.h"
#include "f_swresample.h"
#include "filter_internal.h"
struct priv {
struct mp_log *log;
bool is_resampling;
struct SwrContext *avrctx;
struct mp_aframe *avrctx_fmt; // output format of avrctx
struct mp_aframe *pool_fmt; // format used to allocate frames for avrctx output
struct mp_aframe *pre_out_fmt; // format before final conversion
struct SwrContext *avrctx_out; // for output channel reordering
struct mp_resample_opts *opts; // opts requested by the user
// At least libswresample keeps a pointer around for this:
int reorder_in[MP_NUM_CHANNELS];
int reorder_out[MP_NUM_CHANNELS];
struct mp_aframe_pool *reorder_buffer;
struct mp_aframe_pool *out_pool;
int in_rate_user; // user 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;
int out_format;
struct mp_chmap out_channels;
double current_pts;
struct mp_aframe *input;
double cmd_speed;
double speed;
struct mp_swresample public;
};
#define OPT_BASE_STRUCT struct mp_resample_opts
const struct m_sub_options resample_conf = {
.opts = (const m_option_t[]) {
{"audio-resample-filter-size", OPT_INT(filter_size), M_RANGE(0, 32)},
{"audio-resample-phase-shift", OPT_INT(phase_shift), M_RANGE(0, 30)},
{"audio-resample-linear", OPT_BOOL(linear)},
{"audio-resample-cutoff", OPT_DOUBLE(cutoff), M_RANGE(0, 1)},
{"audio-normalize-downmix", OPT_BOOL(normalize)},
{"audio-resample-max-output-size", OPT_DOUBLE(max_output_frame_size)},
{"audio-swresample-o", OPT_KEYVALUELIST(avopts)},
{0}
},
.size = sizeof(struct mp_resample_opts),
.defaults = &(const struct mp_resample_opts)MP_RESAMPLE_OPTS_DEF,
.change_flags = UPDATE_AUDIO,
};
static double get_delay(struct priv *p)
{
int64_t base = p->in_rate * (int64_t)p->out_rate;
return swr_get_delay(p->avrctx, base) / (double)base;
}
static int get_out_samples(struct priv *p, int in_samples)
{
return swr_get_out_samples(p->avrctx, in_samples);
}
static void close_lavrr(struct priv *p)
{
swr_free(&p->avrctx);
swr_free(&p->avrctx_out);
TA_FREEP(&p->pre_out_fmt);
TA_FREEP(&p->avrctx_fmt);
TA_FREEP(&p->pool_fmt);
}
static int rate_from_speed(int rate, double speed)
{
return lrint(rate * speed);
}
static struct mp_chmap fudge_pairs[][2] = {
{MP_CHMAP2(BL, BR), MP_CHMAP2(SL, SR)},
{MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR)},
{MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL, SR)},
{MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR)},
};
// Modify out_layout and return the new value. The intention is reducing the
// loss libswresample's rematrixing will cause by exchanging similar, but
// strictly speaking incompatible channel pairs. For example, 7.1 should be
// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave
// it to libswresample to create the remix matrix.)
static uint64_t fudge_layout_conversion(struct priv *p,
uint64_t in, uint64_t out)
{
for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) {
uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]);
uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]);
if ((in & a) == a && (in & b) == 0 &&
(out & a) == 0 && (out & b) == b)
{
out = (out & ~b) | a;
MP_VERBOSE(p, "Fudge: %s -> %s\n",
mp_chmap_to_str(&fudge_pairs[n][0]),
mp_chmap_to_str(&fudge_pairs[n][1]));
}
}
return out;
}
// mp_chmap_get_reorder() performs:
// to->speaker[n] = from->speaker[src[n]]
// but libavresample does:
// to->speaker[dst[n]] = from->speaker[n]
static void transpose_order(int *map, int num)
{
int nmap[MP_NUM_CHANNELS] = {0};
for (int n = 0; n < num; n++) {
for (int i = 0; i < num; i++) {
if (map[n] == i)
nmap[i] = n;
}
}
memcpy(map, nmap, sizeof(nmap));
}
static bool configure_lavrr(struct priv *p, bool verbose)
{
close_lavrr(p);
p->in_rate = rate_from_speed(p->in_rate_user, p->speed);
MP_VERBOSE(p, "%dHz %s %s -> %dHz %s %s\n",
p->in_rate, mp_chmap_to_str(&p->in_channels),
af_fmt_to_str(p->in_format),
p->out_rate, mp_chmap_to_str(&p->out_channels),
af_fmt_to_str(p->out_format));
p->avrctx = swr_alloc();
p->avrctx_out = swr_alloc();
if (!p->avrctx || !p->avrctx_out)
goto error;
enum AVSampleFormat in_samplefmt = af_to_avformat(p->in_format);
enum AVSampleFormat out_samplefmt = af_to_avformat(p->out_format);
enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt);
if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
out_samplefmt == AV_SAMPLE_FMT_NONE ||
out_samplefmtp == AV_SAMPLE_FMT_NONE)
{
MP_ERR(p, "unsupported conversion: %s -> %s\n",
af_fmt_to_str(p->in_format), af_fmt_to_str(p->out_format));
goto error;
}
av_opt_set_int(p->avrctx, "filter_size", p->opts->filter_size, 0);
av_opt_set_int(p->avrctx, "phase_shift", p->opts->phase_shift, 0);
av_opt_set_int(p->avrctx, "linear_interp", p->opts->linear, 0);
double cutoff = p->opts->cutoff;
if (cutoff <= 0.0)
cutoff = MPMAX(1.0 - 6.5 / (p->opts->filter_size + 8), 0.80);
av_opt_set_double(p->avrctx, "cutoff", cutoff, 0);
int normalize = p->opts->normalize;
av_opt_set_double(p->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
if (mp_set_avopts(p->log, p->avrctx, p->opts->avopts) < 0)
goto error;
struct mp_chmap map_in = p->in_channels;
struct mp_chmap map_out = p->out_channels;
// Try not to do any remixing if at least one is "unknown". Some corner
// cases also benefit from disabling all channel handling logic if the
// src/dst layouts are the same (like fl-fr-na -> fl-fr-na).
if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out) ||
mp_chmap_equals(&map_in, &map_out))
{
mp_chmap_set_unknown(&map_in, map_in.num);
mp_chmap_set_unknown(&map_out, map_out.num);
}
// unchecked: don't take any channel reordering into account
uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);
struct mp_chmap in_lavc, out_lavc;
mp_chmap_from_lavc(&in_lavc, in_ch_layout);
mp_chmap_from_lavc(&out_lavc, out_ch_layout);
if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) {
MP_VERBOSE(p, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc),
mp_chmap_to_str(&out_lavc));
}
if (in_lavc.num != map_in.num) {
// For handling NA channels, we would have to add a planarization step.
MP_FATAL(p, "Unsupported input channel layout %s.\n",
mp_chmap_to_str(&map_in));
goto error;
}
mp_chmap_get_reorder(p->reorder_in, &map_in, &in_lavc);
transpose_order(p->reorder_in, map_in.num);
if (mp_chmap_equals(&out_lavc, &map_out)) {
// No intermediate step required - output new format directly.
out_samplefmtp = out_samplefmt;
} else {
// Verify that we really just reorder and/or insert NA channels.
struct mp_chmap withna = out_lavc;
mp_chmap_fill_na(&withna, map_out.num);
if (withna.num != map_out.num)
goto error;
}
mp_chmap_get_reorder(p->reorder_out, &out_lavc, &map_out);
p->pre_out_fmt = mp_aframe_create();
mp_aframe_set_rate(p->pre_out_fmt, p->out_rate);
mp_aframe_set_chmap(p->pre_out_fmt, &p->out_channels);
mp_aframe_set_format(p->pre_out_fmt, p->out_format);
p->avrctx_fmt = mp_aframe_create();
mp_aframe_config_copy(p->avrctx_fmt, p->pre_out_fmt);
mp_aframe_set_chmap(p->avrctx_fmt, &out_lavc);
mp_aframe_set_format(p->avrctx_fmt, af_from_avformat(out_samplefmtp));
// If there are NA channels, the final output will have more channels than
// the avrctx output. Also, avrctx will output planar (out_samplefmtp was
// not overwritten). Allocate the output frame with more channels, so the
// NA channels can be trivially added.
p->pool_fmt = mp_aframe_create();
mp_aframe_config_copy(p->pool_fmt, p->avrctx_fmt);
if (map_out.num > out_lavc.num)
mp_aframe_set_chmap(p->pool_fmt, &map_out);
out_ch_layout = fudge_layout_conversion(p, in_ch_layout, out_ch_layout);
#if HAVE_AV_CHANNEL_LAYOUT
// Real conversion; output is input to avrctx_out.
AVChannelLayout in_layout, out_layout;
mp_chmap_to_av_layout(&in_layout, &in_lavc);
mp_chmap_to_av_layout(&out_layout, &out_lavc);
av_opt_set_chlayout(p->avrctx, "in_chlayout", &in_layout, 0);
av_opt_set_chlayout(p->avrctx, "out_chlayout", &out_layout, 0);
#else
av_opt_set_int(p->avrctx, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(p->avrctx, "out_channel_layout", out_ch_layout, 0);
#endif
av_opt_set_int(p->avrctx, "in_sample_rate", p->in_rate, 0);
av_opt_set_int(p->avrctx, "out_sample_rate", p->out_rate, 0);
av_opt_set_int(p->avrctx, "in_sample_fmt", in_samplefmt, 0);
av_opt_set_int(p->avrctx, "out_sample_fmt", out_samplefmtp, 0);
#if HAVE_AV_CHANNEL_LAYOUT
AVChannelLayout fake_layout;
av_channel_layout_default(&fake_layout, map_out.num);
av_opt_set_chlayout(p->avrctx_out, "in_chlayout", &fake_layout, 0);
av_opt_set_chlayout(p->avrctx_out, "out_chlayout", &fake_layout, 0);
#else
// Just needs the correct number of channels for deplanarization.
struct mp_chmap fake_chmap;
mp_chmap_set_unknown(&fake_chmap, map_out.num);
uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
if (!fake_out_ch_layout)
goto error;
av_opt_set_int(p->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0);
av_opt_set_int(p->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
#endif
av_opt_set_int(p->avrctx_out, "in_sample_fmt", out_samplefmtp, 0);
av_opt_set_int(p->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
av_opt_set_int(p->avrctx_out, "in_sample_rate", p->out_rate, 0);
av_opt_set_int(p->avrctx_out, "out_sample_rate", p->out_rate, 0);
// API has weird requirements, quoting avresample.h:
// * This function can only be called when the allocated context is not open.
// * Also, the input channel layout must have already been set.
swr_set_channel_mapping(p->avrctx, p->reorder_in);
p->is_resampling = false;
if (swr_init(p->avrctx) < 0 || swr_init(p->avrctx_out) < 0) {
MP_ERR(p, "Cannot open Libavresample context.\n");
goto error;
}
return true;
error:
close_lavrr(p);
mp_filter_internal_mark_failed(p->public.f);
MP_FATAL(p, "libswresample failed to initialize.\n");
return false;
}
static void swresample_reset(struct mp_filter *f)
{
struct priv *p = f->priv;
p->current_pts = MP_NOPTS_VALUE;
TA_FREEP(&p->input);
if (!p->avrctx)
return;
swr_close(p->avrctx);
if (swr_init(p->avrctx) < 0)
close_lavrr(p);
}
// This relies on the tricky way mpa was allocated.
static bool reorder_planes(struct mp_aframe *mpa, int *reorder,
struct mp_chmap *newmap)
{
if (!mp_aframe_set_chmap(mpa, newmap))
return false;
int num_planes = mp_aframe_get_planes(mpa);
uint8_t **planes = mp_aframe_get_data_rw(mpa);
if (num_planes && !planes)
return false;
uint8_t *old_planes[MP_NUM_CHANNELS];
assert(num_planes <= MP_NUM_CHANNELS);
for (int n = 0; n < num_planes; n++)
old_planes[n] = planes[n];
int next_na = 0;
for (int n = 0; n < num_planes; n++)
next_na += newmap->speaker[n] != MP_SPEAKER_ID_NA;
for (int n = 0; n < num_planes; n++) {
int src = reorder[n];
assert(src >= -1 && src < num_planes);
if (src >= 0) {
planes[n] = old_planes[src];
} else {
assert(next_na < num_planes);
planes[n] = old_planes[next_na++];
// The NA planes were never written by avrctx, so clear them.
af_fill_silence(planes[n],
mp_aframe_get_sstride(mpa) * mp_aframe_get_size(mpa),
mp_aframe_get_format(mpa));
}
}
return true;
}
static int resample_frame(struct SwrContext *r,
struct mp_aframe *out, struct mp_aframe *in,
int consume_in)
{
// Be aware that the channel layout and count can be different for in and
// out frames. In some situations the caller will fix up the frames before
// or after conversion. The sample rates can also be different.
AVFrame *av_i = in ? mp_aframe_get_raw_avframe(in) : NULL;
AVFrame *av_o = out ? mp_aframe_get_raw_avframe(out) : NULL;
return swr_convert(r,
av_o ? av_o->extended_data : NULL,
av_o ? av_o->nb_samples : 0,
(const uint8_t **)(av_i ? av_i->extended_data : NULL),
av_i ? MPMIN(av_i->nb_samples, consume_in) : 0);
}
static struct mp_frame filter_resample_output(struct priv *p,
struct mp_aframe *in)
{
struct mp_aframe *out = NULL;
if (!p->avrctx)
goto error;
// Limit the filtered data size for better latency when changing speed.
// Avoid buffering data within the resampler => restrict input size.
// p->in_rate already includes the speed factor.
double s = p->opts->max_output_frame_size / 1000 * p->in_rate;
int max_in = lrint(MPCLAMP(s, 128, INT_MAX));
int consume_in = in ? mp_aframe_get_size(in) : 0;
consume_in = MPMIN(consume_in, max_in);
int samples = get_out_samples(p, consume_in);
out = mp_aframe_create();
mp_aframe_config_copy(out, p->pool_fmt);
if (mp_aframe_pool_allocate(p->out_pool, out, samples) < 0)
goto error;
int out_samples = 0;
if (samples) {
out_samples = resample_frame(p->avrctx, out, in, consume_in);
if (out_samples < 0 || out_samples > samples)
goto error;
mp_aframe_set_size(out, out_samples);
}
struct mp_chmap out_chmap;
if (!mp_aframe_get_chmap(p->pool_fmt, &out_chmap))
goto error;
if (!reorder_planes(out, p->reorder_out, &out_chmap))
goto error;
if (!mp_aframe_config_equals(out, p->pre_out_fmt)) {
struct mp_aframe *new = mp_aframe_create();
mp_aframe_config_copy(new, p->pre_out_fmt);
if (mp_aframe_pool_allocate(p->reorder_buffer, new, out_samples) < 0) {
talloc_free(new);
goto error;
}
int got = 0;
if (out_samples)
got = resample_frame(p->avrctx_out, new, out, out_samples);
talloc_free(out);
out = new;
if (got != out_samples)
goto error;
}
if (in) {
mp_aframe_copy_attributes(out, in);
p->current_pts = mp_aframe_end_pts(in);
mp_aframe_skip_samples(in, consume_in);
}
if (out_samples) {
if (p->current_pts != MP_NOPTS_VALUE) {
double delay = get_delay(p) * mp_aframe_get_speed(out) +
mp_aframe_duration(out) +
(p->input ? mp_aframe_duration(p->input) : 0);
mp_aframe_set_pts(out, p->current_pts - delay);
mp_aframe_mul_speed(out, p->speed);
}
} else {
TA_FREEP(&out);
}
return out ? MAKE_FRAME(MP_FRAME_AUDIO, out) : MP_NO_FRAME;
error:
talloc_free(out);
MP_ERR(p, "Error on resampling.\n");
mp_filter_internal_mark_failed(p->public.f);
return MP_NO_FRAME;
}
static void swresample_process(struct mp_filter *f)
{
struct priv *p = f->priv;
if (!mp_pin_in_needs_data(f->ppins[1]))
return;
p->speed = p->cmd_speed * p->public.speed;
struct mp_aframe *input = NULL;
if (!p->input) {
struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
if (frame.type == MP_FRAME_AUDIO) {
input = frame.data;
} else if (!frame.type) {
return; // no new data
} else if (frame.type != MP_FRAME_EOF) {
MP_ERR(p, "Unsupported frame type.\n");
mp_frame_unref(&frame);
mp_filter_internal_mark_failed(f);
return;
}
if (!input && !p->avrctx) {
// Obviously no draining needed.
mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
return;
}
}
if (input) {
assert(!p->input);
struct mp_swresample *s = &p->public;
int in_rate = mp_aframe_get_rate(input);
int in_format = mp_aframe_get_format(input);
struct mp_chmap in_channels = {0};
mp_aframe_get_chmap(input, &in_channels);
if (!in_rate || !in_format || !in_channels.num) {
MP_ERR(p, "Frame with invalid format unsupported\n");
talloc_free(input);
mp_filter_internal_mark_failed(f);
return;
}
int out_rate = s->out_rate ? s->out_rate : in_rate;
int out_format = s->out_format ? s->out_format : in_format;
struct mp_chmap out_channels =
s->out_channels.num ? s->out_channels : in_channels;
if (p->in_rate_user != in_rate ||
p->in_format != in_format ||
!mp_chmap_equals(&p->in_channels, &in_channels) ||
p->out_rate != out_rate ||
p->out_format != out_format ||
!mp_chmap_equals(&p->out_channels, &out_channels) ||
!p->avrctx)
{
if (p->avrctx) {
// drain remaining audio
struct mp_frame out = filter_resample_output(p, NULL);
if (out.type) {
mp_pin_in_write(f->ppins[1], out);
// continue filtering next time.
mp_pin_out_unread(f->ppins[0],
MAKE_FRAME(MP_FRAME_AUDIO, input));
input = NULL;
}
}
MP_VERBOSE(p, "format change, reinitializing resampler\n");
p->in_rate_user = in_rate;
p->in_format = in_format;
p->in_channels = in_channels;
p->out_rate = out_rate;
p->out_format = out_format;
p->out_channels = out_channels;
if (!configure_lavrr(p, true)) {
talloc_free(input);
return;
}
if (!input) {
// continue filtering next time
mp_filter_internal_mark_progress(f);
return;
}
}
p->input = input;
}
int new_rate = rate_from_speed(p->in_rate_user, p->speed);
bool exact_rate = new_rate == p->in_rate;
bool use_comp = fabs(new_rate / (double)p->in_rate - 1) <= 0.01;
// If we've never used compensation, avoid setting it - even if it's in
// theory a NOP, libswresample will enable resampling. _If_ we're
// resampling, we might have to disable previously enabled compensation.
if (exact_rate && !p->is_resampling)
use_comp = false;
if (p->avrctx && use_comp) {
AVRational r =
av_d2q(p->speed * p->in_rate_user / p->in_rate, INT_MAX / 2);
// Essentially, swr_set_compensation() does 2 things:
// - adjust output sample rate by sample_delta/compensation_distance
// - reset the adjustment after compensation_distance output samples
// Increase the compensation_distance to avoid undesired reset
// semantics - we want to keep the ratio for the whole frame we're
// feeding it, until the next filter() call.
int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
r = (AVRational){ r.num * mult, r.den * mult };
if (r.den == r.num)
r = (AVRational){0}; // fully disable
if (swr_set_compensation(p->avrctx, r.den - r.num, r.den) >= 0) {
exact_rate = true;
p->is_resampling = true; // libswresample can auto-enable it
}
}
if (!exact_rate) {
// Before reconfiguring, drain the audio that is still buffered
// in the resampler.
struct mp_frame out = filter_resample_output(p, NULL);
bool need_drain = !!out.type;
if (need_drain)
mp_pin_in_write(f->ppins[1], out);
// Reinitialize resampler.
configure_lavrr(p, false);
// If we've written output, we must continue filtering next time.
if (need_drain)
return;
}
struct mp_frame out = filter_resample_output(p, p->input);
if (out.type) {
mp_pin_in_write(f->ppins[1], out);
if (!p->input)
mp_pin_out_repeat_eof(f->ppins[0]);
} else if (p->input) {
mp_filter_internal_mark_progress(f); // try to consume more input
} else {
mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
}
if (p->input && !mp_aframe_get_size(p->input))
TA_FREEP(&p->input);
}
double mp_swresample_get_delay(struct mp_swresample *s)
{
struct priv *p = s->f->priv;
return get_delay(p);
}
static bool swresample_command(struct mp_filter *f, struct mp_filter_command *cmd)
{
struct priv *p = f->priv;
if (cmd->type == MP_FILTER_COMMAND_SET_SPEED_RESAMPLE) {
p->cmd_speed = cmd->speed;
return true;
}
return false;
}
static void swresample_destroy(struct mp_filter *f)
{
struct priv *p = f->priv;
close_lavrr(p);
TA_FREEP(&p->input);
}
static const struct mp_filter_info swresample_filter = {
.name = "swresample",
.priv_size = sizeof(struct priv),
.process = swresample_process,
.command = swresample_command,
.reset = swresample_reset,
.destroy = swresample_destroy,
};
struct mp_swresample *mp_swresample_create(struct mp_filter *parent,
struct mp_resample_opts *opts)
{
struct mp_filter *f = mp_filter_create(parent, &swresample_filter);
if (!f)
return NULL;
mp_filter_add_pin(f, MP_PIN_IN, "in");
mp_filter_add_pin(f, MP_PIN_OUT, "out");
struct priv *p = f->priv;
p->public.f = f;
p->public.speed = 1.0;
p->cmd_speed = 1.0;
p->log = f->log;
if (opts) {
p->opts = talloc_dup(p, opts);
p->opts->avopts = mp_dup_str_array(p, p->opts->avopts);
} else {
p->opts = mp_get_config_group(p, f->global, &resample_conf);
}
p->reorder_buffer = mp_aframe_pool_create(p);
p->out_pool = mp_aframe_pool_create(p);
return &p->public;
}
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