mirror of https://github.com/mpv-player/mpv
483 lines
15 KiB
C
483 lines
15 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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* This file is part of mpv.
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* Based on Michael Niedermayer's lavcresample.
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*
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* MPlayer 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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* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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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/audioconvert.h>
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#include <libavutil/common.h>
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#include <libavutil/samplefmt.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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#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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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 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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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 mp_audio *pending;
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bool avrctx_ok;
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struct AVAudioResampleContext *avrctx;
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struct AVAudioResampleContext *avrctx_out; // for output channel reordering
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struct af_resample_opts ctx; // opts in the context
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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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};
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#if HAVE_LIBAVRESAMPLE
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static int get_delay(struct af_resample *s)
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{
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return avresample_get_delay(s->avrctx);
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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_drain_samples(struct af_resample *s)
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{
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return avresample_get_out_samples(s->avrctx, 0);
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}
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#else
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static int get_delay(struct af_resample *s)
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{
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return swr_get_delay(s->avrctx, s->ctx.in_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 (swr_drop_output(s->avrctx, 1000) > 0) {}
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}
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static int get_drain_samples(struct af_resample *s)
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{
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return 4096; // libswscale does not have this
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}
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#endif
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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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static bool needs_lavrctx_reconfigure(struct af_resample *s,
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struct mp_audio *in,
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struct mp_audio *out)
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{
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return s->ctx.in_rate_af != in->rate ||
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s->ctx.in_format != in->format ||
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!mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
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s->ctx.out_rate != out->rate ||
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s->ctx.out_format != out->format ||
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!mp_chmap_equals(&s->ctx.out_channels, &out->channels) ||
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s->ctx.filter_size != s->opts.filter_size ||
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s->ctx.phase_shift != s->opts.phase_shift ||
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s->ctx.linear != s->opts.linear ||
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s->ctx.cutoff != s->opts.cutoff;
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}
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static bool test_conversion(int src_format, int dst_format)
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{
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return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
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af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
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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)
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{
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struct af_resample *s = af->priv;
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s->avrctx_ok = false;
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enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
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enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
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if (in_samplefmt == AV_SAMPLE_FMT_NONE || out_samplefmt == AV_SAMPLE_FMT_NONE)
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return AF_ERROR;
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avresample_close(s->avrctx);
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avresample_close(s->avrctx_out);
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talloc_free(s->pending);
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s->pending = NULL;
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s->ctx.out_rate = out->rate;
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s->ctx.in_rate_af = in->rate;
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s->ctx.in_rate = rate_from_speed(in->rate, s->playback_speed);
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s->ctx.out_format = out->format;
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s->ctx.in_format = in->format;
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s->ctx.out_channels= out->channels;
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s->ctx.in_channels = in->channels;
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s->ctx.filter_size = s->opts.filter_size;
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s->ctx.phase_shift = s->opts.phase_shift;
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s->ctx.linear = s->opts.linear;
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s->ctx.cutoff = s->opts.cutoff;
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av_opt_set_int(s->avrctx, "filter_size", s->ctx.filter_size, 0);
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av_opt_set_int(s->avrctx, "phase_shift", s->ctx.phase_shift, 0);
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av_opt_set_int(s->avrctx, "linear_interp", s->ctx.linear, 0);
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av_opt_set_double(s->avrctx, "cutoff", s->ctx.cutoff, 0);
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if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
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return AF_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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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->ctx.in_rate, 0);
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av_opt_set_int(s->avrctx, "out_sample_rate", s->ctx.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_samplefmt, 0);
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struct mp_chmap in_lavc;
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mp_chmap_from_lavc(&in_lavc, in_ch_layout);
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mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
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struct mp_chmap out_lavc;
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mp_chmap_from_lavc(&out_lavc, out_ch_layout);
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mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
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// Same configuration; we just reorder.
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av_opt_set_int(s->avrctx_out, "in_channel_layout", out_ch_layout, 0);
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av_opt_set_int(s->avrctx_out, "out_channel_layout", out_ch_layout, 0);
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av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmt, 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->ctx.out_rate, 0);
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av_opt_set_int(s->avrctx_out, "out_sample_rate", s->ctx.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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avresample_set_channel_mapping(s->avrctx_out, s->reorder_out);
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if (avresample_open(s->avrctx) < 0 ||
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avresample_open(s->avrctx_out) < 0)
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{
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MP_ERR(af, "Cannot open Libavresample Context. \n");
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return AF_ERROR;
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}
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s->avrctx_ok = true;
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return AF_OK;
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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 = (struct af_resample *) af->priv;
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struct mp_audio *in = (struct mp_audio *) arg;
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struct mp_audio *out = (struct mp_audio *) af->data;
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switch (cmd) {
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case AF_CONTROL_REINIT: {
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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 (af_to_avformat(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 && needs_lavrctx_reconfigure(s, in, out))
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r = configure_lavrr(af, in, out);
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return r;
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}
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case AF_CONTROL_SET_FORMAT: {
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if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
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return AF_FALSE;
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mp_audio_set_format(af->data, *(int*)arg);
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return AF_OK;
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}
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case AF_CONTROL_SET_CHANNELS: {
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mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
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return AF_OK;
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}
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case AF_CONTROL_SET_RESAMPLE_RATE:
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out->rate = *(int *)arg;
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return AF_OK;
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case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
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s->playback_speed = *(double *)arg;
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int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
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if (new_rate != s->ctx.in_rate && s->avrctx_ok && af->fmt_out.format) {
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// Before reconfiguring, drain the audio that is still buffered
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// in the resampler.
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talloc_free(s->pending);
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s->pending = talloc_zero(NULL, struct mp_audio);
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mp_audio_copy_config(s->pending, &af->fmt_out);
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s->pending->samples = get_drain_samples(s);
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if (s->pending->samples > 0) {
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mp_audio_realloc_min(s->pending, s->pending->samples);
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int r = resample_frame(s->avrctx, s->pending, NULL);
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s->pending->samples = MPMAX(r, 0);
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}
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// Reinitialize resampler.
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configure_lavrr(af, &af->fmt_in, &af->fmt_out);
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}
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return AF_OK;
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}
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case AF_CONTROL_RESET:
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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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#undef ctx_opt_set_int
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#undef ctx_opt_set_dbl
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static void uninit(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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talloc_free(s->pending);
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}
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static bool needs_reorder(int *reorder, int num_ch)
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{
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for (int n = 0; n < num_ch; n++) {
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if (reorder[n] != n)
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return true;
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}
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return false;
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}
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static void reorder_planes(struct mp_audio *mpa, int *reorder)
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{
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struct mp_audio prev = *mpa;
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for (int n = 0; n < mpa->num_planes; n++) {
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assert(reorder[n] >= 0 && reorder[n] < mpa->num_planes);
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mpa->planes[n] = prev.planes[reorder[n]];
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}
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}
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static int filter(struct af_instance *af, struct mp_audio *in)
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{
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struct af_resample *s = af->priv;
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if (s->pending) {
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if (s->pending->samples) {
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af_add_output_frame(af, s->pending);
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} else {
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talloc_free(s->pending);
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}
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s->pending = NULL;
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}
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int samples = avresample_available(s->avrctx) +
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av_rescale_rnd(get_delay(s) + (in ? in->samples : 0),
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s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
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struct mp_audio *out = mp_audio_pool_get(af->out_pool, af->data, samples);
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if (!out)
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goto error;
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if (in)
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mp_audio_copy_attributes(out, in);
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mp_audio_realloc_min(out, out->samples);
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af->delay = get_delay(s) / (double)s->ctx.in_rate;
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if (out->samples) {
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out->samples = resample_frame(s->avrctx, out, in);
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if (out->samples < 0)
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goto error;
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}
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if (needs_reorder(s->reorder_out, out->nch)) {
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if (af_fmt_is_planar(out->format)) {
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reorder_planes(out, s->reorder_out);
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} else if (out->samples) {
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struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer, out,
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out->samples);
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if (!new)
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goto error;
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mp_audio_copy_attributes(new, out);
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int out_samples = resample_frame(s->avrctx_out, new, out);
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talloc_free(out);
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out = new;
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if (out_samples != new->samples)
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goto error;
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}
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}
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talloc_free(in);
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if (out->samples) {
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af_add_output_frame(af, out);
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} else {
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talloc_free(out);
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}
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return 0;
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error:
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talloc_free(in);
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talloc_free(out);
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return -1;
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}
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static int af_open(struct af_instance *af)
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{
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struct af_resample *s = af->priv;
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af->control = control;
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af->uninit = uninit;
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af->filter_frame = filter;
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if (s->opts.cutoff <= 0.0)
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s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
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s->avrctx = avresample_alloc_context();
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s->avrctx_out = avresample_alloc_context();
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s->reorder_buffer = mp_audio_pool_create(s);
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if (s->avrctx && s->avrctx_out) {
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return AF_OK;
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} else {
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MP_ERR(af, "Cannot initialize Libavresample Context. \n");
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uninit(af);
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return AF_ERROR;
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}
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}
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#define OPT_BASE_STRUCT struct af_resample
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const struct af_info af_info_lavrresample = {
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.info = "Sample frequency conversion using libavresample",
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.name = "lavrresample",
|
|
.open = af_open,
|
|
.test_conversion = test_conversion,
|
|
.priv_size = sizeof(struct af_resample),
|
|
.priv_defaults = &(const struct af_resample) {
|
|
.opts = {
|
|
.filter_size = 16,
|
|
.cutoff = 0.0,
|
|
.phase_shift = 10,
|
|
},
|
|
.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_KEYVALUELIST("o", avopts, 0),
|
|
{0}
|
|
},
|
|
};
|