mirror of https://github.com/mpv-player/mpv
618 lines
21 KiB
C
618 lines
21 KiB
C
/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2013 Stefano Pigozzi <stefano.pigozzi@gmail.com>
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*
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* Based on Michael Niedermayer's lavcresample.
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <inttypes.h>
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#include <math.h>
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#include <assert.h>
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#include <libavutil/opt.h>
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#include <libavutil/common.h>
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#include <libavutil/samplefmt.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/mathematics.h>
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#include "common/common.h"
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#include "config.h"
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#if HAVE_LIBAVRESAMPLE
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#include <libavresample/avresample.h>
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#elif HAVE_LIBSWRESAMPLE
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#include <libswresample/swresample.h>
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#define AVAudioResampleContext SwrContext
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#define avresample_alloc_context swr_alloc
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#define avresample_open swr_init
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#define avresample_close(x) do { } while(0)
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#define avresample_free swr_free
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#define avresample_available(x) 0
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#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
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swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
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#define avresample_set_channel_mapping swr_set_channel_mapping
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#define avresample_set_compensation swr_set_compensation
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#else
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#error "config.h broken or no resampler found"
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#endif
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#include "common/av_common.h"
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#include "common/msg.h"
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#include "options/m_option.h"
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#include "audio/filter/af.h"
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#include "audio/fmt-conversion.h"
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#include "osdep/endian.h"
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struct af_resample_opts {
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int filter_size;
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int phase_shift;
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int linear;
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double cutoff;
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int normalize;
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};
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struct af_resample {
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int allow_detach;
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char **avopts;
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double playback_speed;
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struct AVAudioResampleContext *avrctx;
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struct mp_audio avrctx_fmt; // output format of avrctx
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struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
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struct mp_audio pre_out_fmt; // format before final conversion (S24)
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struct AVAudioResampleContext *avrctx_out; // for output channel reordering
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struct af_resample_opts opts; // opts requested by the user
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// At least libswresample keeps a pointer around for this:
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int reorder_in[MP_NUM_CHANNELS];
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int reorder_out[MP_NUM_CHANNELS];
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struct mp_audio_pool *reorder_buffer;
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int in_rate_af; // filter input sample rate
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int in_rate; // actual rate (used by lavr), adjusted for playback speed
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int in_format;
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struct mp_chmap in_channels;
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int out_rate;
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int out_format;
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struct mp_chmap out_channels;
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};
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#if HAVE_LIBAVRESAMPLE
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static double get_delay(struct af_resample *s)
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{
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return avresample_get_delay(s->avrctx) / (double)s->in_rate +
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avresample_available(s->avrctx) / (double)s->out_rate;
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}
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static void drop_all_output(struct af_resample *s)
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{
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while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
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}
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static int get_out_samples(struct af_resample *s, int in_samples)
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{
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return avresample_get_out_samples(s->avrctx, in_samples);
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}
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#else
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static double get_delay(struct af_resample *s)
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{
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int64_t base = s->in_rate * (int64_t)s->out_rate;
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return swr_get_delay(s->avrctx, base) / (double)base;
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}
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static void drop_all_output(struct af_resample *s)
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{
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while (swr_drop_output(s->avrctx, 1000) > 0) {}
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}
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static int get_out_samples(struct af_resample *s, int in_samples)
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{
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#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
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return swr_get_out_samples(s->avrctx, in_samples);
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#else
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return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP)
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+ swr_get_delay(s->avrctx, s->out_rate);
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#endif
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}
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#endif
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static void close_lavrr(struct af_instance *af)
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{
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struct af_resample *s = af->priv;
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if (s->avrctx)
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avresample_close(s->avrctx);
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avresample_free(&s->avrctx);
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if (s->avrctx_out)
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avresample_close(s->avrctx_out);
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avresample_free(&s->avrctx_out);
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}
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static int resample_frame(struct AVAudioResampleContext *r,
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struct mp_audio *out, struct mp_audio *in)
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{
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return avresample_convert(r,
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out ? (uint8_t **)out->planes : NULL,
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out ? mp_audio_get_allocated_size(out) : 0,
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out ? out->samples : 0,
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in ? (uint8_t **)in->planes : NULL,
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in ? mp_audio_get_allocated_size(in) : 0,
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in ? in->samples : 0);
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}
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static double af_resample_default_cutoff(int filter_size)
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{
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return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
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}
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static int rate_from_speed(int rate, double speed)
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{
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return lrint(rate * speed);
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}
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// Return the format libavresample should convert to, given the final output
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// format mp_format. In some cases (S24) we perform an extra conversion step,
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// and signal here what exactly libavresample should output. It will be the
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// input to the final conversion to mp_format.
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static int check_output_conversion(int mp_format)
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{
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if (mp_format == AF_FORMAT_S24)
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return AV_SAMPLE_FMT_S32;
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return af_to_avformat(mp_format);
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}
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static struct mp_chmap fudge_pairs[][2] = {
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{MP_CHMAP2(BL, BR), MP_CHMAP2(SL, SR)},
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{MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR)},
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{MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL, SR)},
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{MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR)},
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};
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// Modify out_layout and return the new value. The intention is reducing the
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// loss libswresample's rematrixing will cause by exchanging similar, but
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// strictly speaking incompatible channel pairs. For example, 7.1 should be
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// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave
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// it to libswresample to create the remix matrix.)
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static uint64_t fudge_layout_conversion(struct af_instance *af,
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uint64_t in, uint64_t out)
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{
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for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) {
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uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]);
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uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]);
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if ((in & a) == a && (in & b) == 0 &&
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(out & a) == 0 && (out & b) == b)
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{
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out = (out & ~b) | a;
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MP_VERBOSE(af, "Fudge: %s -> %s\n",
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mp_chmap_to_str(&fudge_pairs[n][0]),
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mp_chmap_to_str(&fudge_pairs[n][1]));
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}
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}
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return out;
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}
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// mp_chmap_get_reorder() performs:
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// to->speaker[n] = from->speaker[src[n]]
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// but libavresample does:
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// to->speaker[dst[n]] = from->speaker[n]
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static void transpose_order(int *map, int num)
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{
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int nmap[MP_NUM_CHANNELS] = {0};
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for (int n = 0; n < num; n++) {
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for (int i = 0; i < num; i++) {
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if (map[n] == i)
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nmap[i] = n;
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}
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}
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memcpy(map, nmap, sizeof(nmap));
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}
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static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
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struct mp_audio *out, bool verbose)
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{
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struct af_resample *s = af->priv;
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close_lavrr(af);
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s->avrctx = avresample_alloc_context();
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s->avrctx_out = avresample_alloc_context();
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if (!s->avrctx || !s->avrctx_out)
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goto error;
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enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
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enum AVSampleFormat out_samplefmt = check_output_conversion(out->format);
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enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt);
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if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
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out_samplefmt == AV_SAMPLE_FMT_NONE ||
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out_samplefmtp == AV_SAMPLE_FMT_NONE)
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goto error;
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s->out_rate = out->rate;
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s->in_rate_af = in->rate;
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s->in_rate = rate_from_speed(in->rate, s->playback_speed);
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s->out_format = out->format;
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s->in_format = in->format;
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s->out_channels= out->channels;
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s->in_channels = in->channels;
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av_opt_set_int(s->avrctx, "filter_size", s->opts.filter_size, 0);
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av_opt_set_int(s->avrctx, "phase_shift", s->opts.phase_shift, 0);
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av_opt_set_int(s->avrctx, "linear_interp", s->opts.linear, 0);
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av_opt_set_double(s->avrctx, "cutoff", s->opts.cutoff, 0);
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int normalize = s->opts.normalize;
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if (normalize < 0)
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normalize = af->opts->audio_normalize;
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#if HAVE_LIBSWRESAMPLE
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av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
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#else
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av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0);
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#endif
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if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
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goto error;
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struct mp_chmap map_in = in->channels;
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struct mp_chmap map_out = out->channels;
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// Try not to do any remixing if at least one is "unknown".
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if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out)) {
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mp_chmap_set_unknown(&map_in, map_in.num);
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mp_chmap_set_unknown(&map_out, map_out.num);
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}
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// unchecked: don't take any channel reordering into account
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uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
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uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);
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struct mp_chmap in_lavc, out_lavc;
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mp_chmap_from_lavc(&in_lavc, in_ch_layout);
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mp_chmap_from_lavc(&out_lavc, out_ch_layout);
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if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) {
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MP_VERBOSE(af, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc),
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mp_chmap_to_str(&out_lavc));
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}
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if (in_lavc.num != map_in.num) {
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// For handling NA channels, we would have to add a planarization step.
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MP_FATAL(af, "Unsupported channel remapping.\n");
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goto error;
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}
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mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
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transpose_order(s->reorder_in, map_in.num);
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if (mp_chmap_equals(&out_lavc, &map_out)) {
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// No intermediate step required - output new format directly.
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out_samplefmtp = out_samplefmt;
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} else {
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// Verify that we really just reorder and/or insert NA channels.
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struct mp_chmap withna = out_lavc;
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mp_chmap_fill_na(&withna, map_out.num);
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if (withna.num != map_out.num)
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goto error;
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}
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mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
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s->avrctx_fmt = *out;
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mp_audio_set_channels(&s->avrctx_fmt, &out_lavc);
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mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp));
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s->pre_out_fmt = *out;
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mp_audio_set_format(&s->pre_out_fmt, af_from_avformat(out_samplefmt));
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// If there are NA channels, the final output will have more channels than
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// the avrctx output. Also, avrctx will output planar (out_samplefmtp was
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// not overwritten). Allocate the output frame with more channels, so the
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// NA channels can be trivially added.
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s->pool_fmt = s->avrctx_fmt;
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if (map_out.num > out_lavc.num)
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mp_audio_set_channels(&s->pool_fmt, &map_out);
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out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout);
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// Real conversion; output is input to avrctx_out.
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av_opt_set_int(s->avrctx, "in_channel_layout", in_ch_layout, 0);
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av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
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av_opt_set_int(s->avrctx, "in_sample_rate", s->in_rate, 0);
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av_opt_set_int(s->avrctx, "out_sample_rate", s->out_rate, 0);
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av_opt_set_int(s->avrctx, "in_sample_fmt", in_samplefmt, 0);
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av_opt_set_int(s->avrctx, "out_sample_fmt", out_samplefmtp, 0);
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// Just needs the correct number of channels for deplanarization.
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struct mp_chmap fake_chmap;
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mp_chmap_set_unknown(&fake_chmap, map_out.num);
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uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
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if (!fake_out_ch_layout)
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goto error;
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av_opt_set_int(s->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0);
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av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
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av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmtp, 0);
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av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
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av_opt_set_int(s->avrctx_out, "in_sample_rate", s->out_rate, 0);
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av_opt_set_int(s->avrctx_out, "out_sample_rate", s->out_rate, 0);
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// API has weird requirements, quoting avresample.h:
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// * This function can only be called when the allocated context is not open.
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// * Also, the input channel layout must have already been set.
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avresample_set_channel_mapping(s->avrctx, s->reorder_in);
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if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) {
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MP_ERR(af, "Cannot open Libavresample Context. \n");
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goto error;
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}
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return AF_OK;
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error:
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close_lavrr(af);
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return AF_ERROR;
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}
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static int control(struct af_instance *af, int cmd, void *arg)
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{
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struct af_resample *s = af->priv;
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switch (cmd) {
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case AF_CONTROL_REINIT: {
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struct mp_audio *in = arg;
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struct mp_audio *out = af->data;
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struct mp_audio orig_in = *in;
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if (((out->rate == in->rate) || (out->rate == 0)) &&
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(out->format == in->format) &&
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(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
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s->allow_detach && s->playback_speed == 1.0)
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return AF_DETACH;
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if (out->rate == 0)
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out->rate = in->rate;
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if (mp_chmap_is_empty(&out->channels))
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mp_audio_set_channels(out, &in->channels);
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if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
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mp_audio_set_format(in, AF_FORMAT_FLOAT);
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if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
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mp_audio_set_format(out, in->format);
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int r = ((in->format == orig_in.format) &&
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mp_chmap_equals(&in->channels, &orig_in.channels))
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? AF_OK : AF_FALSE;
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if (r == AF_OK)
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r = configure_lavrr(af, in, out, true);
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return r;
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}
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case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
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s->playback_speed = *(double *)arg;
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return AF_OK;
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}
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case AF_CONTROL_RESET:
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if (s->avrctx)
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drop_all_output(s);
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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static void uninit(struct af_instance *af)
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{
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close_lavrr(af);
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}
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// The LSB is always ignored.
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#if BYTE_ORDER == BIG_ENDIAN
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#define SHIFT24(x) ((3-(x))*8)
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#else
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#define SHIFT24(x) (((x)+1)*8)
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#endif
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static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa)
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{
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if (mpa->format == AF_FORMAT_S32 && af->data->format == AF_FORMAT_S24) {
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size_t len = mp_audio_psize(mpa) / mpa->bps;
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for (int s = 0; s < len; s++) {
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uint32_t val = *((uint32_t *)mpa->planes[0] + s);
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uint8_t *ptr = (uint8_t *)mpa->planes[0] + s * 3;
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ptr[0] = val >> SHIFT24(0);
|
|
ptr[1] = val >> SHIFT24(1);
|
|
ptr[2] = val >> SHIFT24(2);
|
|
}
|
|
mp_audio_set_format(mpa, AF_FORMAT_S24);
|
|
}
|
|
|
|
for (int p = 0; p < mpa->num_planes; p++) {
|
|
void *ptr = mpa->planes[p];
|
|
int total = mpa->samples * mpa->spf;
|
|
if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) {
|
|
for (int s = 0; s < total; s++)
|
|
((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f);
|
|
} else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) {
|
|
for (int s = 0; s < total; s++)
|
|
((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0);
|
|
}
|
|
}
|
|
}
|
|
|
|
// This relies on the tricky way mpa was allocated.
|
|
static void reorder_planes(struct mp_audio *mpa, int *reorder,
|
|
struct mp_chmap *newmap)
|
|
{
|
|
struct mp_audio prev = *mpa;
|
|
mp_audio_set_channels(mpa, newmap);
|
|
|
|
// The trailing planes were never written by avrctx, they're the NA channels.
|
|
int next_na = prev.num_planes;
|
|
|
|
for (int n = 0; n < mpa->num_planes; n++) {
|
|
int src = reorder[n];
|
|
assert(src >= -1 && src < prev.num_planes);
|
|
if (src >= 0) {
|
|
mpa->planes[n] = prev.planes[src];
|
|
} else {
|
|
assert(next_na < mpa->num_planes);
|
|
mpa->planes[n] = prev.planes[next_na++];
|
|
af_fill_silence(mpa->planes[n], mpa->sstride * mpa->samples,
|
|
mpa->format);
|
|
}
|
|
}
|
|
}
|
|
|
|
static int filter_resample(struct af_instance *af, struct mp_audio *in)
|
|
{
|
|
struct af_resample *s = af->priv;
|
|
|
|
int samples = get_out_samples(s, in ? in->samples : 0);
|
|
|
|
struct mp_audio out_format = s->pool_fmt;
|
|
struct mp_audio *out = mp_audio_pool_get(af->out_pool, &out_format, samples);
|
|
if (!out)
|
|
goto error;
|
|
if (in)
|
|
mp_audio_copy_attributes(out, in);
|
|
|
|
if (!s->avrctx)
|
|
goto error;
|
|
|
|
if (out->samples) {
|
|
out->samples = resample_frame(s->avrctx, out, in);
|
|
if (out->samples < 0)
|
|
goto error;
|
|
}
|
|
|
|
struct mp_audio real_out = *out;
|
|
mp_audio_copy_config(out, &s->avrctx_fmt);
|
|
|
|
if (out->samples && !mp_audio_config_equals(out, &s->pre_out_fmt)) {
|
|
assert(af_fmt_is_planar(out->format) && out->format == real_out.format);
|
|
reorder_planes(out, s->reorder_out, &s->pool_fmt.channels);
|
|
if (!mp_audio_config_equals(out, &s->pre_out_fmt)) {
|
|
struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer,
|
|
&s->pre_out_fmt,
|
|
out->samples);
|
|
if (!new)
|
|
goto error;
|
|
mp_audio_copy_attributes(new, out);
|
|
int out_samples = resample_frame(s->avrctx_out, new, out);
|
|
talloc_free(out);
|
|
out = new;
|
|
if (out_samples != new->samples)
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
extra_output_conversion(af, out);
|
|
|
|
talloc_free(in);
|
|
if (out->samples) {
|
|
af_add_output_frame(af, out);
|
|
} else {
|
|
talloc_free(out);
|
|
}
|
|
|
|
af->delay = get_delay(s);
|
|
|
|
return 0;
|
|
error:
|
|
talloc_free(in);
|
|
talloc_free(out);
|
|
return -1;
|
|
}
|
|
|
|
static int filter(struct af_instance *af, struct mp_audio *in)
|
|
{
|
|
struct af_resample *s = af->priv;
|
|
|
|
int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
|
|
bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
|
|
|
|
if (s->avrctx) {
|
|
AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
|
|
INT_MAX / 2);
|
|
// Essentially, swr/avresample_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 (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) < 0)
|
|
need_reinit = true;
|
|
}
|
|
|
|
if (need_reinit && new_rate != s->in_rate) {
|
|
// Before reconfiguring, drain the audio that is still buffered
|
|
// in the resampler.
|
|
filter_resample(af, NULL);
|
|
// Reinitialize resampler.
|
|
configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
|
|
}
|
|
|
|
return filter_resample(af, in);
|
|
}
|
|
|
|
static int af_open(struct af_instance *af)
|
|
{
|
|
struct af_resample *s = af->priv;
|
|
|
|
af->control = control;
|
|
af->uninit = uninit;
|
|
af->filter_frame = filter;
|
|
|
|
if (s->opts.cutoff <= 0.0)
|
|
s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
|
|
|
|
s->reorder_buffer = mp_audio_pool_create(s);
|
|
|
|
return AF_OK;
|
|
}
|
|
|
|
#define OPT_BASE_STRUCT struct af_resample
|
|
|
|
const struct af_info af_info_lavrresample = {
|
|
.info = "Sample frequency conversion using libavresample",
|
|
.name = "lavrresample",
|
|
.open = af_open,
|
|
.priv_size = sizeof(struct af_resample),
|
|
.priv_defaults = &(const struct af_resample) {
|
|
.opts = {
|
|
.filter_size = 16,
|
|
.cutoff = 0.0,
|
|
.phase_shift = 10,
|
|
.normalize = -1,
|
|
},
|
|
.playback_speed = 1.0,
|
|
.allow_detach = 1,
|
|
},
|
|
.options = (const struct m_option[]) {
|
|
OPT_INTRANGE("filter-size", opts.filter_size, 0, 0, 32),
|
|
OPT_INTRANGE("phase-shift", opts.phase_shift, 0, 0, 30),
|
|
OPT_FLAG("linear", opts.linear, 0),
|
|
OPT_DOUBLE("cutoff", opts.cutoff, M_OPT_RANGE, .min = 0, .max = 1),
|
|
OPT_FLAG("detach", allow_detach, 0),
|
|
OPT_CHOICE("normalize", opts.normalize, 0,
|
|
({"no", 0}, {"yes", 1}, {"auto", -1})),
|
|
OPT_KEYVALUELIST("o", avopts, 0),
|
|
{0}
|
|
},
|
|
};
|