diff options
Diffstat (limited to 'audio/filter/af_lavrresample.c')
-rw-r--r-- | audio/filter/af_lavrresample.c | 617 |
1 files changed, 0 insertions, 617 deletions
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c deleted file mode 100644 index 47c374b227..0000000000 --- a/audio/filter/af_lavrresample.c +++ /dev/null @@ -1,617 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2013 Stefano Pigozzi <stefano.pigozzi@gmail.com> - * - * Based on Michael Niedermayer's lavcresample. - * - * This file is part of mpv. - * - * mpv is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * mpv is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with mpv. If not, see <http://www.gnu.org/licenses/>. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <inttypes.h> -#include <math.h> -#include <assert.h> - -#include <libavutil/opt.h> -#include <libavutil/common.h> -#include <libavutil/samplefmt.h> -#include <libavutil/channel_layout.h> -#include <libavutil/mathematics.h> - -#include "common/common.h" -#include "config.h" - -#define HAVE_LIBSWRESAMPLE HAVE_IS_FFMPEG -#define HAVE_LIBAVRESAMPLE HAVE_IS_LIBAV - -#if HAVE_LIBAVRESAMPLE -#include <libavresample/avresample.h> -#elif HAVE_LIBSWRESAMPLE -#include <libswresample/swresample.h> -#define AVAudioResampleContext SwrContext -#define avresample_alloc_context swr_alloc -#define avresample_open swr_init -#define avresample_close(x) do { } while(0) -#define avresample_free swr_free -#define avresample_available(x) 0 -#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \ - swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples) -#define avresample_set_channel_mapping swr_set_channel_mapping -#define avresample_set_compensation swr_set_compensation -#else -#error "config.h broken or no resampler found" -#endif - -#include "common/av_common.h" -#include "common/msg.h" -#include "options/m_option.h" -#include "audio/filter/af.h" -#include "audio/fmt-conversion.h" -#include "osdep/endian.h" - -struct af_resample_opts { - int filter_size; - int phase_shift; - int linear; - double cutoff; - int normalize; -}; - -struct af_resample { - int allow_detach; - char **avopts; - double playback_speed; - struct AVAudioResampleContext *avrctx; - struct mp_audio avrctx_fmt; // output format of avrctx - struct mp_audio pool_fmt; // format used to allocate frames for avrctx output - struct mp_audio pre_out_fmt; // format before final conversion (S24) - struct AVAudioResampleContext *avrctx_out; // for output channel reordering - struct af_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_audio_pool *reorder_buffer; - - int in_rate_af; // filter input sample rate - int in_rate; // actual rate (used by lavr), adjusted for playback speed - int in_format; - struct mp_chmap in_channels; - int out_rate; - int out_format; - struct mp_chmap out_channels; -}; - -#if HAVE_LIBAVRESAMPLE -static double get_delay(struct af_resample *s) -{ - return avresample_get_delay(s->avrctx) / (double)s->in_rate + - avresample_available(s->avrctx) / (double)s->out_rate; -} -static int get_out_samples(struct af_resample *s, int in_samples) -{ - return avresample_get_out_samples(s->avrctx, in_samples); -} -#else -static double get_delay(struct af_resample *s) -{ - int64_t base = s->in_rate * (int64_t)s->out_rate; - return swr_get_delay(s->avrctx, base) / (double)base; -} -static int get_out_samples(struct af_resample *s, int in_samples) -{ - return swr_get_out_samples(s->avrctx, in_samples); -} -#endif - -static void close_lavrr(struct af_instance *af) -{ - struct af_resample *s = af->priv; - - if (s->avrctx) - avresample_close(s->avrctx); - avresample_free(&s->avrctx); - if (s->avrctx_out) - avresample_close(s->avrctx_out); - avresample_free(&s->avrctx_out); -} - -static int resample_frame(struct AVAudioResampleContext *r, - struct mp_audio *out, struct mp_audio *in) -{ - return avresample_convert(r, - out ? (uint8_t **)out->planes : NULL, - out ? mp_audio_get_allocated_size(out) : 0, - out ? out->samples : 0, - in ? (uint8_t **)in->planes : NULL, - in ? mp_audio_get_allocated_size(in) : 0, - in ? in->samples : 0); -} - -static double af_resample_default_cutoff(int filter_size) -{ - return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80); -} - -static int rate_from_speed(int rate, double speed) -{ - return lrint(rate * speed); -} - -// Return the format libavresample should convert to, given the final output -// format mp_format. In some cases (S24) we perform an extra conversion step, -// and signal here what exactly libavresample should output. It will be the -// input to the final conversion to mp_format. -static int check_output_conversion(int mp_format) -{ - if (mp_format == AF_FORMAT_S24) - return AV_SAMPLE_FMT_S32; - return af_to_avformat(mp_format); -} - -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 af_instance *af, - 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(af, "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 int configure_lavrr(struct af_instance *af, struct mp_audio *in, - struct mp_audio *out, bool verbose) -{ - struct af_resample *s = af->priv; - - close_lavrr(af); - - s->avrctx = avresample_alloc_context(); - s->avrctx_out = avresample_alloc_context(); - if (!s->avrctx || !s->avrctx_out) - goto error; - - enum AVSampleFormat in_samplefmt = af_to_avformat(in->format); - enum AVSampleFormat out_samplefmt = check_output_conversion(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) - goto error; - - s->out_rate = out->rate; - s->in_rate_af = in->rate; - s->in_rate = rate_from_speed(in->rate, s->playback_speed); - s->out_format = out->format; - s->in_format = in->format; - s->out_channels= out->channels; - s->in_channels = in->channels; - - av_opt_set_int(s->avrctx, "filter_size", s->opts.filter_size, 0); - av_opt_set_int(s->avrctx, "phase_shift", s->opts.phase_shift, 0); - av_opt_set_int(s->avrctx, "linear_interp", s->opts.linear, 0); - - av_opt_set_double(s->avrctx, "cutoff", s->opts.cutoff, 0); - - int normalize = s->opts.normalize; - if (normalize < 0) - normalize = af->opts->audio_normalize; -#if HAVE_LIBSWRESAMPLE - av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0); -#else - av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0); -#endif - - if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0) - goto error; - - struct mp_chmap map_in = in->channels; - struct mp_chmap map_out = out->channels; - - // Try not to do any remixing if at least one is "unknown". - if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&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(af, "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(af, "Unsupported channel remapping.\n"); - goto error; - } - - mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc); - transpose_order(s->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(s->reorder_out, &out_lavc, &map_out); - - s->avrctx_fmt = *out; - mp_audio_set_channels(&s->avrctx_fmt, &out_lavc); - mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp)); - - s->pre_out_fmt = *out; - mp_audio_set_format(&s->pre_out_fmt, af_from_avformat(out_samplefmt)); - - // 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. - s->pool_fmt = s->avrctx_fmt; - if (map_out.num > out_lavc.num) - mp_audio_set_channels(&s->pool_fmt, &map_out); - - out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout); - - // Real conversion; output is input to avrctx_out. - av_opt_set_int(s->avrctx, "in_channel_layout", in_ch_layout, 0); - av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0); - av_opt_set_int(s->avrctx, "in_sample_rate", s->in_rate, 0); - av_opt_set_int(s->avrctx, "out_sample_rate", s->out_rate, 0); - av_opt_set_int(s->avrctx, "in_sample_fmt", in_samplefmt, 0); - av_opt_set_int(s->avrctx, "out_sample_fmt", out_samplefmtp, 0); - - // 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(s->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0); - av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0); - - av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmtp, 0); - av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0); - av_opt_set_int(s->avrctx_out, "in_sample_rate", s->out_rate, 0); - av_opt_set_int(s->avrctx_out, "out_sample_rate", s->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. - avresample_set_channel_mapping(s->avrctx, s->reorder_in); - - if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) { - MP_ERR(af, "Cannot open Libavresample Context. \n"); - goto error; - } - return AF_OK; - -error: - close_lavrr(af); - return AF_ERROR; -} - - -static int control(struct af_instance *af, int cmd, void *arg) -{ - struct af_resample *s = af->priv; - - switch (cmd) { - case AF_CONTROL_REINIT: { - struct mp_audio *in = arg; - struct mp_audio *out = af->data; - struct mp_audio orig_in = *in; - - if (((out->rate == in->rate) || (out->rate == 0)) && - (out->format == in->format) && - (mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) && - s->allow_detach && s->playback_speed == 1.0) - return AF_DETACH; - - if (out->rate == 0) - out->rate = in->rate; - - if (mp_chmap_is_empty(&out->channels)) - mp_audio_set_channels(out, &in->channels); - - if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) - mp_audio_set_format(in, AF_FORMAT_FLOAT); - if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE) - mp_audio_set_format(out, in->format); - - int r = ((in->format == orig_in.format) && - mp_chmap_equals(&in->channels, &orig_in.channels)) - ? AF_OK : AF_FALSE; - - if (r == AF_OK) - r = configure_lavrr(af, in, out, true); - return r; - } - case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: { - s->playback_speed = *(double *)arg; - return AF_OK; - } - case AF_CONTROL_RESET: - if (s->avrctx) { -#if HAVE_LIBSWRESAMPLE - swr_close(s->avrctx); - if (swr_init(s->avrctx) < 0) { - close_lavrr(af); - return AF_ERROR; - } -#else - while (avresample_read(s->avrctx, NULL, 1000) > 0) {} -#endif - } - return AF_OK; - } - return AF_UNKNOWN; -} - -static void uninit(struct af_instance *af) -{ - close_lavrr(af); -} - -// The LSB is always ignored. -#if BYTE_ORDER == BIG_ENDIAN -#define SHIFT24(x) ((3-(x))*8) -#else -#define SHIFT24(x) (((x)+1)*8) -#endif - -static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa) -{ - if (mpa->format == AF_FORMAT_S32 && af->data->format == AF_FORMAT_S24) { - size_t len = mp_audio_psize(mpa) / mpa->bps; - for (int s = 0; s < len; s++) { - uint32_t val = *((uint32_t *)mpa->planes[0] + s); - uint8_t *ptr = (uint8_t *)mpa->planes[0] + s * 3; - 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; - struct mp_audio *out = NULL; - - if (!s->avrctx) - goto error; - - int samples = get_out_samples(s, in ? in->samples : 0); - - struct mp_audio out_format = s->pool_fmt; - out = mp_audio_pool_get(af->out_pool, &out_format, samples); - if (!out) - goto error; - if (in) - mp_audio_copy_attributes(out, in); - - 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} - }, -}; |